asterisk/channels/chan_misdn.c

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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2004 - 2006, Christian Richter
*
* Christian Richter <crich@beronet.com>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*
*/
/*!
* \file
*
* \brief the chan_misdn channel driver for Asterisk
*
* \author Christian Richter <crich@beronet.com>
*
* MISDN http://www.misdn.org/
*
* \ingroup channel_drivers
*/
/*! \li \ref chan_misdn.c uses the configuration file \ref misdn.conf
* \addtogroup configuration_file
*/
/*! \page misdn.conf misdn.conf
* \verbinclude misdn.conf.sample
*/
/*!
* \note
* To use the CCBS/CCNR supplementary service feature and other
* supplementary services using FACILITY messages requires a
* modified version of mISDN.
*
* \note
* The latest modified mISDN v1.1.x based version is available at:
* http://svn.digium.com/svn/thirdparty/mISDN/trunk
* http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
*
* \note
* Taged versions of the modified mISDN code are available under:
* http://svn.digium.com/svn/thirdparty/mISDN/tags
* http://svn.digium.com/svn/thirdparty/mISDNuser/tags
*/
/* Define to enable cli commands to generate canned CCBS messages. */
// #define CCBS_TEST_MESSAGES 1
/*
* XXX The mISDN channel driver needs its native bridge code
* converted to the new bridge technology scheme. The
* chan_dahdi native bridge code can be used as an example. It
* is unlikely that this will ever get done. Support for this
* channel driver is dwindling because the supported version of
* mISDN does not support newer kernels.
*
* Without native bridge support, the following config file
* parameters have no effect: bridging.
*
* The existing native bridge code is marked with the
* mISDN_NATIVE_BRIDGING conditional.
*/
/*** MODULEINFO
<depend>isdnnet</depend>
<depend>misdn</depend>
<depend>suppserv</depend>
<support_level>extended</support_level>
***/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include <pthread.h>
#include <sys/socket.h>
#include <sys/time.h>
#include <arpa/inet.h>
#include <fcntl.h>
#include <sys/ioctl.h>
#include <signal.h>
#include <sys/file.h>
#include <semaphore.h>
#include <ctype.h>
#include <time.h>
#include "asterisk/channel.h"
#include "asterisk/config.h"
#include "asterisk/module.h"
#include "asterisk/pbx.h"
#include "asterisk/io.h"
#include "asterisk/frame.h"
#include "asterisk/translate.h"
#include "asterisk/cli.h"
#include "asterisk/musiconhold.h"
#include "asterisk/dsp.h"
#include "asterisk/file.h"
#include "asterisk/callerid.h"
#include "asterisk/indications.h"
#include "asterisk/app.h"
#include "asterisk/features.h"
#include "asterisk/term.h"
#include "asterisk/sched.h"
#include "asterisk/stringfields.h"
#include "asterisk/abstract_jb.h"
#include "asterisk/causes.h"
#include "asterisk/format.h"
#include "asterisk/format_cap.h"
#include "asterisk/features_config.h"
#include "asterisk/bridge.h"
#include "asterisk/pickup.h"
#include "chan_misdn_config.h"
#include "isdn_lib.h"
static char global_tracefile[BUFFERSIZE + 1];
static int g_config_initialized = 0;
struct misdn_jb{
int size;
int upper_threshold;
char *samples, *ok;
int wp,rp;
int state_empty;
int state_full;
int state_buffer;
int bytes_wrote;
ast_mutex_t mutexjb;
};
/*! \brief allocates the jb-structure and initialize the elements */
struct misdn_jb *misdn_jb_init(int size, int upper_threshold);
/*! \brief frees the data and destroys the given jitterbuffer struct */
void misdn_jb_destroy(struct misdn_jb *jb);
/*! \brief fills the jitterbuffer with len data returns < 0 if there was an
error (buffer overrun). */
int misdn_jb_fill(struct misdn_jb *jb, const char *data, int len);
/*! \brief gets len bytes out of the jitterbuffer if available, else only the
available data is returned and the return value indicates the number
of data. */
int misdn_jb_empty(struct misdn_jb *jb, char *data, int len);
static char *complete_ch(struct ast_cli_args *a);
static char *complete_debug_port(struct ast_cli_args *a);
static char *complete_show_config(struct ast_cli_args *a);
/* BEGIN: chan_misdn.h */
#if defined(AST_MISDN_ENHANCEMENTS)
/*
* This timeout duration is to clean up any call completion records that
* are forgotten about by the switch.
*/
#define MISDN_CC_RECORD_AGE_MAX (6UL * 60 * 60) /* seconds */
#define MISDN_CC_REQUEST_WAIT_MAX 5 /* seconds */
/*!
* \brief Caller that initialized call completion services
*
* \details
* This data is the payload for a datastore that is put on the channel that
* initializes call completion services. This datastore is set to be inherited
* by the outbound mISDN channel. When one of these channels hangs up, the
* channel pointer will be set to NULL. That way, we can ensure that we do not
* touch this channel after it gets destroyed.
*/
struct misdn_cc_caller {
/*! \brief The channel that initialized call completion services */
struct ast_channel *chan;
};
struct misdn_cc_notify {
/*! \brief Dialplan: Notify extension priority */
int priority;
/*! \brief Dialplan: Notify extension context */
char context[AST_MAX_CONTEXT];
/*! \brief Dialplan: Notify extension number (User-A) */
char exten[AST_MAX_EXTENSION];
};
/*! \brief mISDN call completion record */
struct misdn_cc_record {
/*! \brief Call completion record linked list */
AST_LIST_ENTRY(misdn_cc_record) list;
/*! \brief Time the record was created. */
time_t time_created;
/*! \brief MISDN_CC_RECORD_ID value */
long record_id;
/*!
* \brief Logical Layer 1 port associated with this
* call completion record
*/
int port;
/*! \brief TRUE if point-to-point mode (CCBS-T/CCNR-T mode) */
int ptp;
/*! \brief Mode specific parameters */
union {
/*! \brief point-to-point specific parameters. */
struct {
/*!
* \brief Call-completion signaling link.
* NULL if signaling link not established.
*/
struct misdn_bchannel *bc;
/*!
* \brief TRUE if we requested the request retention option
* to be enabled.
*/
int requested_retention;
/*!
* \brief TRUE if the request retention option is enabled.
*/
int retention_enabled;
} ptp;
/*! \brief point-to-multi-point specific parameters. */
struct {
/*! \brief CallLinkageID (valid when port determined) */
int linkage_id;
/*! \breif CCBSReference (valid when activated is TRUE) */
int reference_id;
/*! \brief globalRecall(0), specificRecall(1) */
int recall_mode;
} ptmp;
} mode;
/*! \brief TRUE if call completion activated */
int activated;
/*! \brief Outstanding message ID (valid when outstanding_message) */
int invoke_id;
/*! \brief TRUE if waiting for a response from a message (invoke_id is valid) */
int outstanding_message;
/*! \brief TRUE if activation has been requested */
int activation_requested;
/*!
* \brief TRUE if User-A is free
* \note PTMP - Used to answer CCBSStatusRequest.
* PTP - Determines how to respond to CCBS_T_RemoteUserFree.
*/
int party_a_free;
/*! \brief Error code received from last outstanding message. */
enum FacErrorCode error_code;
/*! \brief Reject code received from last outstanding message. */
enum FacRejectCode reject_code;
/*!
* \brief Saved struct misdn_bchannel call information when
* attempted to call User-B
*/
struct {
/*! \brief User-A caller id information */
struct misdn_party_id caller;
/*! \brief User-B number information */
struct misdn_party_dialing dialed;
/*! \brief The BC, HLC (optional) and LLC (optional) contents from the SETUP message. */
struct Q931_Bc_Hlc_Llc setup_bc_hlc_llc;
/*! \brief SETUP message bearer capability field code value */
int capability;
/*! \brief TRUE if call made in digital HDLC mode */
int hdlc;
} redial;
/*! \brief Dialplan location to indicate User-B free and User-A is free */
struct misdn_cc_notify remote_user_free;
/*! \brief Dialplan location to indicate User-B free and User-A is busy */
struct misdn_cc_notify b_free;
};
/*! \brief mISDN call completion record database */
static AST_LIST_HEAD_STATIC(misdn_cc_records_db, misdn_cc_record);
/*! \brief Next call completion record ID to use */
static __u16 misdn_cc_record_id;
/*! \brief Next invoke ID to use */
static __s16 misdn_invoke_id;
static const char misdn_no_response_from_network[] = "No response from network";
static const char misdn_cc_record_not_found[] = "Call completion record not found";
/* mISDN channel variable names */
#define MISDN_CC_RECORD_ID "MISDN_CC_RECORD_ID"
#define MISDN_CC_STATUS "MISDN_CC_STATUS"
#define MISDN_ERROR_MSG "MISDN_ERROR_MSG"
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
static ast_mutex_t release_lock;
enum misdn_chan_state {
MISDN_NOTHING = 0, /*!< at beginning */
MISDN_WAITING4DIGS, /*!< when waiting for info */
MISDN_EXTCANTMATCH, /*!< when asterisk couldn't match our ext */
MISDN_INCOMING_SETUP, /*!< for incoming setup */
MISDN_DIALING, /*!< when pbx_start */
MISDN_PROGRESS, /*!< we have progress */
MISDN_PROCEEDING, /*!< we have progress */
MISDN_CALLING, /*!< when misdn_call is called */
MISDN_CALLING_ACKNOWLEDGE, /*!< when we get SETUP_ACK */
MISDN_ALERTING, /*!< when Alerting */
MISDN_BUSY, /*!< when BUSY */
MISDN_CONNECTED, /*!< when connected */
MISDN_DISCONNECTED, /*!< when connected */
MISDN_CLEANING, /*!< when hangup from * but we were connected before */
};
/*! Asterisk created the channel (outgoing call) */
#define ORG_AST 1
/*! mISDN created the channel (incoming call) */
#define ORG_MISDN 2
enum misdn_hold_state {
MISDN_HOLD_IDLE, /*!< HOLD not active */
MISDN_HOLD_ACTIVE, /*!< Call is held */
MISDN_HOLD_TRANSFER, /*!< Held call is being transferred */
MISDN_HOLD_DISCONNECT, /*!< Held call is being disconnected */
};
struct hold_info {
/*!
* \brief Call HOLD state.
*/
enum misdn_hold_state state;
/*!
* \brief Logical port the channel call record is HELD on
* because the B channel is no longer associated.
*/
int port;
/*!
* \brief Original B channel number the HELD call was using.
* \note Used only for debug display messages.
*/
int channel;
};
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
#define chan_list_ref(obj, debug) (ao2_t_ref((obj), +1, (debug)), (obj))
#define chan_list_unref(obj, debug) (ao2_t_ref((obj), -1, (debug)), NULL)
/*!
* \brief Channel call record structure
*/
struct chan_list {
/*!
* \brief The "allowed_bearers" string read in from /etc/asterisk/misdn.conf
*/
char allowed_bearers[BUFFERSIZE + 1];
/*!
* \brief State of the channel
*/
enum misdn_chan_state state;
/*!
* \brief TRUE if a hangup needs to be queued
* \note This is a debug flag only used to catch calls to hangup_chan() that are already hungup.
*/
int need_queue_hangup;
/*!
* \brief TRUE if a channel can be hung up by calling asterisk directly when done.
*/
int need_hangup;
/*!
* \brief TRUE if we could send an AST_CONTROL_BUSY if needed.
*/
int need_busy;
/*!
* \brief Who originally created this channel. ORG_AST or ORG_MISDN
*/
int originator;
/*!
* \brief TRUE of we are not to respond immediately to a SETUP message. Check the dialplan first.
* \note The "noautorespond_on_setup" boolean read in from /etc/asterisk/misdn.conf
*/
int noautorespond_on_setup;
int norxtone; /*!< Boolean assigned values but the value is not used. */
/*!
* \brief TRUE if we are not to generate tones (Playtones)
*/
int notxtone;
/*!
* \brief TRUE if echo canceller is enabled. Value is toggled.
*/
int toggle_ec;
/*!
* \brief TRUE if you want to send Tone Indications to an incoming
* ISDN channel on a TE Port.
* \note The "incoming_early_audio" boolean read in from /etc/asterisk/misdn.conf
*/
int incoming_early_audio;
/*!
* \brief TRUE if DTMF digits are to be passed inband only.
* \note It is settable by the misdn_set_opt() application.
*/
int ignore_dtmf;
/*!
* \brief Pipe file descriptor handles array.
* Read from pipe[0], write to pipe[1]
*/
int pipe[2];
/*!
* \brief Read buffer for inbound audio from pipe[0]
*/
char ast_rd_buf[4096];
/*!
* \brief Inbound audio frame returned by misdn_read().
*/
struct ast_frame frame;
/*!
* \brief Fax detection option. (0:no 1:yes 2:yes+nojump)
* \note The "faxdetect" option string read in from /etc/asterisk/misdn.conf
* \note It is settable by the misdn_set_opt() application.
*/
int faxdetect;
/*!
* \brief Number of seconds to detect a Fax machine when detection enabled.
* \note 0 disables the timeout.
* \note The "faxdetect_timeout" value read in from /etc/asterisk/misdn.conf
*/
int faxdetect_timeout;
/*!
* \brief Starting time of fax detection with timeout when nonzero.
*/
struct timeval faxdetect_tv;
/*!
* \brief TRUE if a fax has been detected.
*/
int faxhandled;
/*!
* \brief TRUE if we will use the Asterisk DSP to detect DTMF/Fax
* \note The "astdtmf" boolean read in from /etc/asterisk/misdn.conf
*/
int ast_dsp;
/*!
* \brief Jitterbuffer length
* \note The "jitterbuffer" value read in from /etc/asterisk/misdn.conf
*/
int jb_len;
/*!
* \brief Jitterbuffer upper threshold
* \note The "jitterbuffer_upper_threshold" value read in from /etc/asterisk/misdn.conf
*/
int jb_upper_threshold;
/*!
* \brief Allocated jitterbuffer controller
* \note misdn_jb_init() creates the jitterbuffer.
* \note Must use misdn_jb_destroy() to clean up.
*/
struct misdn_jb *jb;
/*!
* \brief Allocated DSP controller
* \note ast_dsp_new() creates the DSP controller.
* \note Must use ast_dsp_free() to clean up.
*/
struct ast_dsp *dsp;
/*!
* \brief Associated Asterisk channel structure.
*/
struct ast_channel * ast;
/*!
* \brief Associated B channel structure.
*/
struct misdn_bchannel *bc;
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \brief Peer channel for which call completion was initialized.
*/
struct misdn_cc_caller *peer;
/*! \brief Associated call completion record ID (-1 if not associated) */
long record_id;
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
/*!
* \brief HELD channel call information
*/
struct hold_info hold;
/*!
* \brief From associated B channel: Layer 3 process ID
* \note Used to find the HELD channel call record when retrieving a call.
*/
unsigned int l3id;
/*!
* \brief From associated B channel: B Channel mISDN driver layer ID from mISDN_get_layerid()
* \note Used only for debug display messages.
*/
int addr;
/*!
* \brief Incoming call dialplan context identifier.
* \note The "context" string read in from /etc/asterisk/misdn.conf
*/
char context[AST_MAX_CONTEXT];
/*!
* \brief The configured music-on-hold class to use for this call.
* \note The "musicclass" string read in from /etc/asterisk/misdn.conf
*/
char mohinterpret[MAX_MUSICCLASS];
/*!
* \brief Number of outgoing audio frames dropped since last debug gripe message.
*/
int dropped_frame_cnt;
/*!
* \brief TRUE if we must do the ringback tones.
* \note The "far_alerting" boolean read in from /etc/asterisk/misdn.conf
*/
int far_alerting;
/*!
* \brief TRUE if NT should disconnect an overlap dialing call when a timeout occurs.
* \note The "nttimeout" boolean read in from /etc/asterisk/misdn.conf
*/
int nttimeout;
/*!
* \brief Tone zone sound used for dialtone generation.
* \note Used as a boolean. Non-NULL to prod generation if enabled.
*/
Merge a large set of updates to the Asterisk indications API. This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17 20:41:24 +00:00
struct ast_tone_zone_sound *ts;
/*!
* \brief Enables overlap dialing for the set amount of seconds. (0 = Disabled)
* \note The "overlapdial" value read in from /etc/asterisk/misdn.conf
*/
int overlap_dial;
/*!
* \brief Overlap dialing timeout Task ID. -1 if not running.
*/
int overlap_dial_task;
/*!
* \brief overlap_tv access lock.
*/
ast_mutex_t overlap_tv_lock;
/*!
* \brief Overlap timer start time. Timer restarted for every digit received.
*/
struct timeval overlap_tv;
/*!
* \brief Next channel call record in the list.
*/
struct chan_list *next;
};
int MAXTICS = 8;
void export_ch(struct ast_channel *chan, struct misdn_bchannel *bc, struct chan_list *ch);
void import_ch(struct ast_channel *chan, struct misdn_bchannel *bc, struct chan_list *ch);
static struct ast_frame *process_ast_dsp(struct chan_list *tmp, struct ast_frame *frame);
struct robin_list {
char *group;
int port;
int channel;
struct robin_list *next;
struct robin_list *prev;
};
static struct robin_list *robin = NULL;
static void free_robin_list(void)
{
struct robin_list *r;
struct robin_list *next;
for (r = robin, robin = NULL; r; r = next) {
next = r->next;
ast_free(r->group);
ast_free(r);
}
}
static struct robin_list *get_robin_position(char *group)
{
struct robin_list *new;
struct robin_list *iter = robin;
for (; iter; iter = iter->next) {
if (!strcasecmp(iter->group, group)) {
return iter;
}
}
new = ast_calloc(1, sizeof(*new));
if (!new) {
return NULL;
}
new->group = ast_strdup(group);
if (!new->group) {
ast_free(new);
return NULL;
}
new->channel = 1;
if (robin) {
new->next = robin;
robin->prev = new;
}
robin = new;
return robin;
}
/*! \brief the main schedule context for stuff like l1 watcher, overlap dial, ... */
static struct ast_sched_context *misdn_tasks = NULL;
static pthread_t misdn_tasks_thread;
static int *misdn_ports;
static void chan_misdn_log(int level, int port, char *tmpl, ...)
__attribute__((format(printf, 3, 4)));
static struct ast_channel *misdn_new(struct chan_list *cl, int state, char *exten, char *callerid, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, int port, int c);
static void send_digit_to_chan(struct chan_list *cl, char digit);
static int pbx_start_chan(struct chan_list *ch);
#define MISDN_ASTERISK_TECH_PVT(ast) ast_channel_tech_pvt(ast)
#define MISDN_ASTERISK_TECH_PVT_SET(ast, value) ast_channel_tech_pvt_set(ast, value)
#include "asterisk/strings.h"
/* #define MISDN_DEBUG 1 */
static const char misdn_type[] = "mISDN";
static int tracing = 0;
/*! \brief Only alaw and mulaw is allowed for now */
static struct ast_format prefformat; /* AST_FORMAT_SLINEAR ; AST_FORMAT_ULAW | */
static int *misdn_debug;
static int *misdn_debug_only;
static int max_ports;
static int *misdn_in_calls;
static int *misdn_out_calls;
/*!
* \brief Global channel call record list head.
*/
static struct chan_list *cl_te=NULL;
static ast_mutex_t cl_te_lock;
static enum event_response_e
cb_events(enum event_e event, struct misdn_bchannel *bc, void *user_data);
static int send_cause2ast(struct ast_channel *ast, struct misdn_bchannel *bc, struct chan_list *ch);
static void cl_queue_chan(struct chan_list *chan);
static int dialtone_indicate(struct chan_list *cl);
static void hanguptone_indicate(struct chan_list *cl);
static int stop_indicate(struct chan_list *cl);
static int start_bc_tones(struct chan_list *cl);
static int stop_bc_tones(struct chan_list *cl);
static void release_chan_early(struct chan_list *ch);
static void release_chan(struct chan_list *ch, struct misdn_bchannel *bc);
#if defined(AST_MISDN_ENHANCEMENTS)
static const char misdn_command_name[] = "misdn_command";
static int misdn_command_exec(struct ast_channel *chan, const char *data);
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
static int misdn_check_l2l1(struct ast_channel *chan, const char *data);
static int misdn_set_opt_exec(struct ast_channel *chan, const char *data);
static int misdn_facility_exec(struct ast_channel *chan, const char *data);
int chan_misdn_jb_empty(struct misdn_bchannel *bc, char *buf, int len);
void debug_numtype(int port, int numtype, char *type);
int add_out_calls(int port);
int add_in_calls(int port);
#ifdef MISDN_1_2
static int update_pipeline_config(struct misdn_bchannel *bc);
#else
static int update_ec_config(struct misdn_bchannel *bc);
#endif
/*************** Helpers *****************/
static int misdn_chan_is_valid(struct chan_list *ch)
{
struct chan_list *list;
ast_mutex_lock(&cl_te_lock);
for (list = cl_te; list; list = list->next) {
if (list == ch) {
ast_mutex_unlock(&cl_te_lock);
return 1;
}
}
ast_mutex_unlock(&cl_te_lock);
return 0;
}
#if defined(mISDN_NATIVE_BRIDGING)
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
/*! Returns a reference to the found chan_list. */
static struct chan_list *get_chan_by_ast(struct ast_channel *ast)
{
struct chan_list *tmp;
ast_mutex_lock(&cl_te_lock);
for (tmp = cl_te; tmp; tmp = tmp->next) {
if (tmp->ast == ast) {
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
chan_list_ref(tmp, "Found chan_list by ast");
ast_mutex_unlock(&cl_te_lock);
return tmp;
}
}
ast_mutex_unlock(&cl_te_lock);
return NULL;
}
#endif /* defined(mISDN_NATIVE_BRIDGING) */
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
/*! Returns a reference to the found chan_list. */
static struct chan_list *get_chan_by_ast_name(const char *name)
{
struct chan_list *tmp;
ast_mutex_lock(&cl_te_lock);
for (tmp = cl_te; tmp; tmp = tmp->next) {
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
if (tmp->ast && strcmp(ast_channel_name(tmp->ast), name) == 0) {
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
chan_list_ref(tmp, "Found chan_list by ast name");
ast_mutex_unlock(&cl_te_lock);
return tmp;
}
}
ast_mutex_unlock(&cl_te_lock);
return NULL;
}
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Destroy the misdn_cc_ds_info datastore payload
*
* \param[in] data the datastore payload, a reference to an misdn_cc_caller
*
* \details
* Since the payload is a reference to an astobj2 object, we just decrement its
* reference count. Before doing so, we NULL out the channel pointer inside of
* the misdn_cc_caller instance. This function will be called in one of two
* cases. In both cases, we no longer need the channel pointer:
*
* - The original channel that initialized call completion services, the same
* channel that is stored here, has been destroyed early. This could happen
* if it transferred the mISDN channel, for example.
*
* - The mISDN channel that had this datastore inherited on to it is now being
* destroyed. If this is the case, then the call completion events have
* already occurred and the appropriate channel variables have already been
* set on the original channel that requested call completion services.
*
* \return Nothing
*/
static void misdn_cc_ds_destroy(void *data)
{
struct misdn_cc_caller *cc_caller = data;
ao2_lock(cc_caller);
cc_caller->chan = NULL;
ao2_unlock(cc_caller);
ao2_ref(cc_caller, -1);
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Duplicate the misdn_cc_ds_info datastore payload
*
* \param[in] data the datastore payload, a reference to an misdn_cc_caller
*
* \details
* All we need to do is bump the reference count and return the same instance.
*
* \return A reference to an instance of a misdn_cc_caller
*/
static void *misdn_cc_ds_duplicate(void *data)
{
struct misdn_cc_caller *cc_caller = data;
ao2_ref(cc_caller, +1);
return cc_caller;
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
static const struct ast_datastore_info misdn_cc_ds_info = {
.type = "misdn_cc",
.destroy = misdn_cc_ds_destroy,
.duplicate = misdn_cc_ds_duplicate,
};
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Set a channel var on the peer channel for call completion services
*
* \param[in] peer The peer that initialized call completion services
* \param[in] var The variable name to set
* \param[in] value The variable value to set
*
* This function may be called from outside of the channel thread. It handles
* the fact that the peer channel may be hung up and destroyed at any time.
*
* \return nothing
*/
static void misdn_cc_set_peer_var(struct misdn_cc_caller *peer, const char *var,
const char *value)
{
ao2_lock(peer);
/*! \todo XXX This nastiness can go away once ast_channel is ref counted! */
while (peer->chan && ast_channel_trylock(peer->chan)) {
ao2_unlock(peer);
sched_yield();
ao2_lock(peer);
}
if (peer->chan) {
pbx_builtin_setvar_helper(peer->chan, var, value);
ast_channel_unlock(peer->chan);
}
ao2_unlock(peer);
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Get a reference to the CC caller if it exists
*/
static struct misdn_cc_caller *misdn_cc_caller_get(struct ast_channel *chan)
{
struct ast_datastore *datastore;
struct misdn_cc_caller *cc_caller;
ast_channel_lock(chan);
if (!(datastore = ast_channel_datastore_find(chan, &misdn_cc_ds_info, NULL))) {
ast_channel_unlock(chan);
return NULL;
}
ao2_ref(datastore->data, +1);
cc_caller = datastore->data;
ast_channel_unlock(chan);
return cc_caller;
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Find the call completion record given the record id.
*
* \param record_id
*
* \retval pointer to found call completion record
* \retval NULL if not found
*
* \note Assumes the misdn_cc_records_db lock is already obtained.
*/
static struct misdn_cc_record *misdn_cc_find_by_id(long record_id)
{
struct misdn_cc_record *current;
AST_LIST_TRAVERSE(&misdn_cc_records_db, current, list) {
if (current->record_id == record_id) {
/* Found the record */
break;
}
}
return current;
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Find the call completion record given the port and call linkage id.
*
* \param port Logical port number
* \param linkage_id Call linkage ID number from switch.
*
* \retval pointer to found call completion record
* \retval NULL if not found
*
* \note Assumes the misdn_cc_records_db lock is already obtained.
*/
static struct misdn_cc_record *misdn_cc_find_by_linkage(int port, int linkage_id)
{
struct misdn_cc_record *current;
AST_LIST_TRAVERSE(&misdn_cc_records_db, current, list) {
if (current->port == port
&& !current->ptp
&& current->mode.ptmp.linkage_id == linkage_id) {
/* Found the record */
break;
}
}
return current;
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Find the call completion record given the port and outstanding invocation id.
*
* \param port Logical port number
* \param invoke_id Outstanding message invocation ID number.
*
* \retval pointer to found call completion record
* \retval NULL if not found
*
* \note Assumes the misdn_cc_records_db lock is already obtained.
*/
static struct misdn_cc_record *misdn_cc_find_by_invoke(int port, int invoke_id)
{
struct misdn_cc_record *current;
AST_LIST_TRAVERSE(&misdn_cc_records_db, current, list) {
if (current->outstanding_message
&& current->invoke_id == invoke_id
&& current->port == port) {
/* Found the record */
break;
}
}
return current;
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Find the call completion record given the port and CCBS reference id.
*
* \param port Logical port number
* \param reference_id CCBS reference ID number from switch.
*
* \retval pointer to found call completion record
* \retval NULL if not found
*
* \note Assumes the misdn_cc_records_db lock is already obtained.
*/
static struct misdn_cc_record *misdn_cc_find_by_reference(int port, int reference_id)
{
struct misdn_cc_record *current;
AST_LIST_TRAVERSE(&misdn_cc_records_db, current, list) {
if (current->activated
&& current->port == port
&& !current->ptp
&& current->mode.ptmp.reference_id == reference_id) {
/* Found the record */
break;
}
}
return current;
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Find the call completion record given the B channel pointer
*
* \param bc B channel control structure pointer.
*
* \retval pointer to found call completion record
* \retval NULL if not found
*
* \note Assumes the misdn_cc_records_db lock is already obtained.
*/
static struct misdn_cc_record *misdn_cc_find_by_bc(const struct misdn_bchannel *bc)
{
struct misdn_cc_record *current;
if (bc) {
AST_LIST_TRAVERSE(&misdn_cc_records_db, current, list) {
if (current->ptp
&& current->mode.ptp.bc == bc) {
/* Found the record */
break;
}
}
} else {
current = NULL;
}
return current;
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Delete the given call completion record
*
* \param doomed Call completion record to destroy
*
* \return Nothing
*
* \note Assumes the misdn_cc_records_db lock is already obtained.
*/
static void misdn_cc_delete(struct misdn_cc_record *doomed)
{
struct misdn_cc_record *current;
AST_LIST_TRAVERSE_SAFE_BEGIN(&misdn_cc_records_db, current, list) {
if (current == doomed) {
AST_LIST_REMOVE_CURRENT(list);
ast_free(current);
return;
}
}
AST_LIST_TRAVERSE_SAFE_END;
/* The doomed node is not in the call completion database */
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Delete all old call completion records
*
* \return Nothing
*
* \note Assumes the misdn_cc_records_db lock is already obtained.
*/
static void misdn_cc_remove_old(void)
{
struct misdn_cc_record *current;
time_t now;
now = time(NULL);
AST_LIST_TRAVERSE_SAFE_BEGIN(&misdn_cc_records_db, current, list) {
if (MISDN_CC_RECORD_AGE_MAX < now - current->time_created) {
if (current->ptp && current->mode.ptp.bc) {
/* Close the old call-completion signaling link */
current->mode.ptp.bc->fac_out.Function = Fac_None;
current->mode.ptp.bc->out_cause = AST_CAUSE_NORMAL_CLEARING;
misdn_lib_send_event(current->mode.ptp.bc, EVENT_RELEASE_COMPLETE);
}
/* Remove the old call completion record */
AST_LIST_REMOVE_CURRENT(list);
ast_free(current);
}
}
AST_LIST_TRAVERSE_SAFE_END;
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Allocate the next record id.
*
* \retval New record id on success.
* \retval -1 on error.
*
* \note Assumes the misdn_cc_records_db lock is already obtained.
*/
static long misdn_cc_record_id_new(void)
{
long record_id;
long first_id;
record_id = ++misdn_cc_record_id;
first_id = record_id;
while (misdn_cc_find_by_id(record_id)) {
record_id = ++misdn_cc_record_id;
if (record_id == first_id) {
/*
* We have a resource leak.
* We should never need to allocate 64k records.
*/
chan_misdn_log(0, 0, " --> ERROR Too many call completion records!\n");
record_id = -1;
break;
}
}
return record_id;
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Create a new call completion record
*
* \retval pointer to new call completion record
* \retval NULL if failed
*
* \note Assumes the misdn_cc_records_db lock is already obtained.
*/
static struct misdn_cc_record *misdn_cc_new(void)
{
struct misdn_cc_record *cc_record;
long record_id;
misdn_cc_remove_old();
cc_record = ast_calloc(1, sizeof(*cc_record));
if (cc_record) {
record_id = misdn_cc_record_id_new();
if (record_id < 0) {
ast_free(cc_record);
return NULL;
}
/* Initialize the new record */
cc_record->record_id = record_id;
cc_record->port = -1;/* Invalid port so it will never be found this way */
cc_record->invoke_id = ++misdn_invoke_id;
cc_record->party_a_free = 1;/* Default User-A as free */
cc_record->error_code = FacError_None;
cc_record->reject_code = FacReject_None;
cc_record->time_created = time(NULL);
/* Insert the new record into the database */
AST_LIST_INSERT_HEAD(&misdn_cc_records_db, cc_record, list);
}
return cc_record;
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Destroy the call completion record database
*
* \return Nothing
*/
static void misdn_cc_destroy(void)
{
struct misdn_cc_record *current;
while ((current = AST_LIST_REMOVE_HEAD(&misdn_cc_records_db, list))) {
/* Do a misdn_cc_delete(current) inline */
ast_free(current);
}
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Initialize the call completion record database
*
* \return Nothing
*/
static void misdn_cc_init(void)
{
misdn_cc_record_id = 0;
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Check the status of an outstanding invocation request.
*
* \param data Points to an integer containing the call completion record id.
*
* \retval 0 if got a response.
* \retval -1 if no response yet.
*/
static int misdn_cc_response_check(void *data)
{
int not_responded;
struct misdn_cc_record *cc_record;
AST_LIST_LOCK(&misdn_cc_records_db);
cc_record = misdn_cc_find_by_id(*(long *) data);
if (cc_record) {
if (cc_record->outstanding_message) {
not_responded = -1;
} else {
not_responded = 0;
}
} else {
/* No record so there is no response to check. */
not_responded = 0;
}
AST_LIST_UNLOCK(&misdn_cc_records_db);
return not_responded;
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Wait for a response from the switch for an outstanding
* invocation request.
*
* \param chan Asterisk channel to operate upon.
* \param wait_seconds Number of seconds to wait
* \param record_id Call completion record ID.
*
* \return Nothing
*/
static void misdn_cc_response_wait(struct ast_channel *chan, int wait_seconds, long record_id)
{
unsigned count;
for (count = 2 * MISDN_CC_REQUEST_WAIT_MAX; count--;) {
/* Sleep in 500 ms increments */
if (ast_safe_sleep_conditional(chan, 500, misdn_cc_response_check, &record_id) != 0) {
/* We got hung up or our response came in. */
break;
}
}
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Convert the mISDN reject code to a string
*
* \param code mISDN reject code.
*
* \return The mISDN reject code as a string
*/
static const char *misdn_to_str_reject_code(enum FacRejectCode code)
{
static const struct {
enum FacRejectCode code;
char *name;
} arr[] = {
/* *INDENT-OFF* */
{ FacReject_None, "No reject occurred" },
{ FacReject_Unknown, "Unknown reject code" },
{ FacReject_Gen_UnrecognizedComponent, "General: Unrecognized Component" },
{ FacReject_Gen_MistypedComponent, "General: Mistyped Component" },
{ FacReject_Gen_BadlyStructuredComponent, "General: Badly Structured Component" },
{ FacReject_Inv_DuplicateInvocation, "Invoke: Duplicate Invocation" },
{ FacReject_Inv_UnrecognizedOperation, "Invoke: Unrecognized Operation" },
{ FacReject_Inv_MistypedArgument, "Invoke: Mistyped Argument" },
{ FacReject_Inv_ResourceLimitation, "Invoke: Resource Limitation" },
{ FacReject_Inv_InitiatorReleasing, "Invoke: Initiator Releasing" },
{ FacReject_Inv_UnrecognizedLinkedID, "Invoke: Unrecognized Linked ID" },
{ FacReject_Inv_LinkedResponseUnexpected, "Invoke: Linked Response Unexpected" },
{ FacReject_Inv_UnexpectedChildOperation, "Invoke: Unexpected Child Operation" },
{ FacReject_Res_UnrecognizedInvocation, "Result: Unrecognized Invocation" },
{ FacReject_Res_ResultResponseUnexpected, "Result: Result Response Unexpected" },
{ FacReject_Res_MistypedResult, "Result: Mistyped Result" },
{ FacReject_Err_UnrecognizedInvocation, "Error: Unrecognized Invocation" },
{ FacReject_Err_ErrorResponseUnexpected, "Error: Error Response Unexpected" },
{ FacReject_Err_UnrecognizedError, "Error: Unrecognized Error" },
{ FacReject_Err_UnexpectedError, "Error: Unexpected Error" },
{ FacReject_Err_MistypedParameter, "Error: Mistyped Parameter" },
/* *INDENT-ON* */
};
unsigned index;
for (index = 0; index < ARRAY_LEN(arr); ++index) {
if (arr[index].code == code) {
return arr[index].name;
}
}
return "unknown";
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Convert the mISDN error code to a string
*
* \param code mISDN error code.
*
* \return The mISDN error code as a string
*/
static const char *misdn_to_str_error_code(enum FacErrorCode code)
{
static const struct {
enum FacErrorCode code;
char *name;
} arr[] = {
/* *INDENT-OFF* */
{ FacError_None, "No error occurred" },
{ FacError_Unknown, "Unknown OID error code" },
{ FacError_Gen_NotSubscribed, "General: Not Subscribed" },
{ FacError_Gen_NotAvailable, "General: Not Available" },
{ FacError_Gen_NotImplemented, "General: Not Implemented" },
{ FacError_Gen_InvalidServedUserNr, "General: Invalid Served User Number" },
{ FacError_Gen_InvalidCallState, "General: Invalid Call State" },
{ FacError_Gen_BasicServiceNotProvided, "General: Basic Service Not Provided" },
{ FacError_Gen_NotIncomingCall, "General: Not Incoming Call" },
{ FacError_Gen_SupplementaryServiceInteractionNotAllowed,"General: Supplementary Service Interaction Not Allowed" },
{ FacError_Gen_ResourceUnavailable, "General: Resource Unavailable" },
{ FacError_Div_InvalidDivertedToNr, "Diversion: Invalid Diverted To Number" },
{ FacError_Div_SpecialServiceNr, "Diversion: Special Service Number" },
{ FacError_Div_DiversionToServedUserNr, "Diversion: Diversion To Served User Number" },
{ FacError_Div_IncomingCallAccepted, "Diversion: Incoming Call Accepted" },
{ FacError_Div_NumberOfDiversionsExceeded, "Diversion: Number Of Diversions Exceeded" },
{ FacError_Div_NotActivated, "Diversion: Not Activated" },
{ FacError_Div_RequestAlreadyAccepted, "Diversion: Request Already Accepted" },
{ FacError_AOC_NoChargingInfoAvailable, "AOC: No Charging Info Available" },
{ FacError_CCBS_InvalidCallLinkageID, "CCBS: Invalid Call Linkage ID" },
{ FacError_CCBS_InvalidCCBSReference, "CCBS: Invalid CCBS Reference" },
{ FacError_CCBS_LongTermDenial, "CCBS: Long Term Denial" },
{ FacError_CCBS_ShortTermDenial, "CCBS: Short Term Denial" },
{ FacError_CCBS_IsAlreadyActivated, "CCBS: Is Already Activated" },
{ FacError_CCBS_AlreadyAccepted, "CCBS: Already Accepted" },
{ FacError_CCBS_OutgoingCCBSQueueFull, "CCBS: Outgoing CCBS Queue Full" },
{ FacError_CCBS_CallFailureReasonNotBusy, "CCBS: Call Failure Reason Not Busy" },
{ FacError_CCBS_NotReadyForCall, "CCBS: Not Ready For Call" },
{ FacError_CCBS_T_LongTermDenial, "CCBS-T: Long Term Denial" },
{ FacError_CCBS_T_ShortTermDenial, "CCBS-T: Short Term Denial" },
{ FacError_ECT_LinkIdNotAssignedByNetwork, "ECT: Link ID Not Assigned By Network" },
/* *INDENT-ON* */
};
unsigned index;
for (index = 0; index < ARRAY_LEN(arr); ++index) {
if (arr[index].code == code) {
return arr[index].name;
}
}
return "unknown";
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Convert mISDN redirecting reason to diversion reason.
*
* \param reason mISDN redirecting reason code.
*
* \return Supported diversion reason code.
*/
static unsigned misdn_to_diversion_reason(enum mISDN_REDIRECTING_REASON reason)
{
unsigned diversion_reason;
switch (reason) {
case mISDN_REDIRECTING_REASON_CALL_FWD:
diversion_reason = 1;/* cfu */
break;
case mISDN_REDIRECTING_REASON_CALL_FWD_BUSY:
diversion_reason = 2;/* cfb */
break;
case mISDN_REDIRECTING_REASON_NO_REPLY:
diversion_reason = 3;/* cfnr */
break;
default:
diversion_reason = 0;/* unknown */
break;
}
return diversion_reason;
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Convert diversion reason to mISDN redirecting reason
*
* \param diversion_reason Diversion reason to convert
*
* \return Supported redirecting reason code.
*/
static enum mISDN_REDIRECTING_REASON diversion_reason_to_misdn(unsigned diversion_reason)
{
enum mISDN_REDIRECTING_REASON reason;
switch (diversion_reason) {
case 1:/* cfu */
reason = mISDN_REDIRECTING_REASON_CALL_FWD;
break;
case 2:/* cfb */
reason = mISDN_REDIRECTING_REASON_CALL_FWD_BUSY;
break;
case 3:/* cfnr */
reason = mISDN_REDIRECTING_REASON_NO_REPLY;
break;
default:
reason = mISDN_REDIRECTING_REASON_UNKNOWN;
break;
}
return reason;
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Convert the mISDN presentation to PresentedNumberUnscreened type
*
* \param presentation mISDN presentation to convert
* \param number_present TRUE if the number is present
*
* \return PresentedNumberUnscreened type
*/
static unsigned misdn_to_PresentedNumberUnscreened_type(int presentation, int number_present)
{
unsigned type;
switch (presentation) {
case 0:/* allowed */
if (number_present) {
type = 0;/* presentationAllowedNumber */
} else {
type = 2;/* numberNotAvailableDueToInterworking */
}
break;
case 1:/* restricted */
if (number_present) {
type = 3;/* presentationRestrictedNumber */
} else {
type = 1;/* presentationRestricted */
}
break;
default:
type = 2;/* numberNotAvailableDueToInterworking */
break;
}
return type;
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Convert the PresentedNumberUnscreened type to mISDN presentation
*
* \param type PresentedNumberUnscreened type
*
* \return mISDN presentation
*/
static int PresentedNumberUnscreened_to_misdn_pres(unsigned type)
{
int presentation;
switch (type) {
default:
case 0:/* presentationAllowedNumber */
presentation = 0;/* allowed */
break;
case 1:/* presentationRestricted */
case 3:/* presentationRestrictedNumber */
presentation = 1;/* restricted */
break;
case 2:/* numberNotAvailableDueToInterworking */
presentation = 2;/* unavailable */
break;
}
return presentation;
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Convert the mISDN numbering plan to PartyNumber numbering plan
*
* \param number_plan mISDN numbering plan
*
* \return PartyNumber numbering plan
*/
static unsigned misdn_to_PartyNumber_plan(enum mISDN_NUMBER_PLAN number_plan)
{
unsigned party_plan;
switch (number_plan) {
default:
case NUMPLAN_UNKNOWN:
party_plan = 0;/* unknown */
break;
case NUMPLAN_ISDN:
party_plan = 1;/* public */
break;
case NUMPLAN_DATA:
party_plan = 3;/* data */
break;
case NUMPLAN_TELEX:
party_plan = 4;/* telex */
break;
case NUMPLAN_NATIONAL:
party_plan = 8;/* nationalStandard */
break;
case NUMPLAN_PRIVATE:
party_plan = 5;/* private */
break;
}
return party_plan;
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Convert PartyNumber numbering plan to mISDN numbering plan
*
* \param party_plan PartyNumber numbering plan
*
* \return mISDN numbering plan
*/
static enum mISDN_NUMBER_PLAN PartyNumber_to_misdn_plan(unsigned party_plan)
{
enum mISDN_NUMBER_PLAN number_plan;
switch (party_plan) {
default:
case 0:/* unknown */
number_plan = NUMPLAN_UNKNOWN;
break;
case 1:/* public */
number_plan = NUMPLAN_ISDN;
break;
case 3:/* data */
number_plan = NUMPLAN_DATA;
break;
case 4:/* telex */
number_plan = NUMPLAN_TELEX;
break;
case 8:/* nationalStandard */
number_plan = NUMPLAN_NATIONAL;
break;
case 5:/* private */
number_plan = NUMPLAN_PRIVATE;
break;
}
return number_plan;
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Convert mISDN type-of-number to PartyNumber public type-of-number
*
* \param ton mISDN type-of-number
*
* \return PartyNumber public type-of-number
*/
static unsigned misdn_to_PartyNumber_ton_public(enum mISDN_NUMBER_TYPE ton)
{
unsigned party_ton;
switch (ton) {
default:
case NUMTYPE_UNKNOWN:
party_ton = 0;/* unknown */
break;
case NUMTYPE_INTERNATIONAL:
party_ton = 1;/* internationalNumber */
break;
case NUMTYPE_NATIONAL:
party_ton = 2;/* nationalNumber */
break;
case NUMTYPE_NETWORK_SPECIFIC:
party_ton = 3;/* networkSpecificNumber */
break;
case NUMTYPE_SUBSCRIBER:
party_ton = 4;/* subscriberNumber */
break;
case NUMTYPE_ABBREVIATED:
party_ton = 6;/* abbreviatedNumber */
break;
}
return party_ton;
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Convert the PartyNumber public type-of-number to mISDN type-of-number
*
* \param party_ton PartyNumber public type-of-number
*
* \return mISDN type-of-number
*/
static enum mISDN_NUMBER_TYPE PartyNumber_to_misdn_ton_public(unsigned party_ton)
{
enum mISDN_NUMBER_TYPE ton;
switch (party_ton) {
default:
case 0:/* unknown */
ton = NUMTYPE_UNKNOWN;
break;
case 1:/* internationalNumber */
ton = NUMTYPE_INTERNATIONAL;
break;
case 2:/* nationalNumber */
ton = NUMTYPE_NATIONAL;
break;
case 3:/* networkSpecificNumber */
ton = NUMTYPE_NETWORK_SPECIFIC;
break;
case 4:/* subscriberNumber */
ton = NUMTYPE_SUBSCRIBER;
break;
case 6:/* abbreviatedNumber */
ton = NUMTYPE_ABBREVIATED;
break;
}
return ton;
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Convert mISDN type-of-number to PartyNumber private type-of-number
*
* \param ton mISDN type-of-number
*
* \return PartyNumber private type-of-number
*/
static unsigned misdn_to_PartyNumber_ton_private(enum mISDN_NUMBER_TYPE ton)
{
unsigned party_ton;
switch (ton) {
default:
case NUMTYPE_UNKNOWN:
party_ton = 0;/* unknown */
break;
case NUMTYPE_INTERNATIONAL:
party_ton = 1;/* level2RegionalNumber */
break;
case NUMTYPE_NATIONAL:
party_ton = 2;/* level1RegionalNumber */
break;
case NUMTYPE_NETWORK_SPECIFIC:
party_ton = 3;/* pTNSpecificNumber */
break;
case NUMTYPE_SUBSCRIBER:
party_ton = 4;/* localNumber */
break;
case NUMTYPE_ABBREVIATED:
party_ton = 6;/* abbreviatedNumber */
break;
}
return party_ton;
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Convert the PartyNumber private type-of-number to mISDN type-of-number
*
* \param party_ton PartyNumber private type-of-number
*
* \return mISDN type-of-number
*/
static enum mISDN_NUMBER_TYPE PartyNumber_to_misdn_ton_private(unsigned party_ton)
{
enum mISDN_NUMBER_TYPE ton;
switch (party_ton) {
default:
case 0:/* unknown */
ton = NUMTYPE_UNKNOWN;
break;
case 1:/* level2RegionalNumber */
ton = NUMTYPE_INTERNATIONAL;
break;
case 2:/* level1RegionalNumber */
ton = NUMTYPE_NATIONAL;
break;
case 3:/* pTNSpecificNumber */
ton = NUMTYPE_NETWORK_SPECIFIC;
break;
case 4:/* localNumber */
ton = NUMTYPE_SUBSCRIBER;
break;
case 6:/* abbreviatedNumber */
ton = NUMTYPE_ABBREVIATED;
break;
}
return ton;
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
/*!
* \internal
* \brief Convert the mISDN type of number code to a string
*
* \param number_type mISDN type of number code.
*
* \return The mISDN type of number code as a string
*/
static const char *misdn_to_str_ton(enum mISDN_NUMBER_TYPE number_type)
{
const char *str;
switch (number_type) {
default:
case NUMTYPE_UNKNOWN:
str = "Unknown";
break;
case NUMTYPE_INTERNATIONAL:
str = "International";
break;
case NUMTYPE_NATIONAL:
str = "National";
break;
case NUMTYPE_NETWORK_SPECIFIC:
str = "Network Specific";
break;
case NUMTYPE_SUBSCRIBER:
str = "Subscriber";
break;
case NUMTYPE_ABBREVIATED:
str = "Abbreviated";
break;
}
return str;
}
/*!
* \internal
* \brief Convert the mISDN type of number code to Asterisk type of number code
*
* \param number_type mISDN type of number code.
*
* \return Asterisk type of number code
*/
static int misdn_to_ast_ton(enum mISDN_NUMBER_TYPE number_type)
{
int ast_number_type;
switch (number_type) {
default:
case NUMTYPE_UNKNOWN:
ast_number_type = NUMTYPE_UNKNOWN << 4;
break;
case NUMTYPE_INTERNATIONAL:
ast_number_type = NUMTYPE_INTERNATIONAL << 4;
break;
case NUMTYPE_NATIONAL:
ast_number_type = NUMTYPE_NATIONAL << 4;
break;
case NUMTYPE_NETWORK_SPECIFIC:
ast_number_type = NUMTYPE_NETWORK_SPECIFIC << 4;
break;
case NUMTYPE_SUBSCRIBER:
ast_number_type = NUMTYPE_SUBSCRIBER << 4;
break;
case NUMTYPE_ABBREVIATED:
ast_number_type = NUMTYPE_ABBREVIATED << 4;
break;
}
return ast_number_type;
}
/*!
* \internal
* \brief Convert the Asterisk type of number code to mISDN type of number code
*
* \param ast_number_type Asterisk type of number code.
*
* \return mISDN type of number code
*/
static enum mISDN_NUMBER_TYPE ast_to_misdn_ton(unsigned ast_number_type)
{
enum mISDN_NUMBER_TYPE number_type;
switch ((ast_number_type >> 4) & 0x07) {
default:
case NUMTYPE_UNKNOWN:
number_type = NUMTYPE_UNKNOWN;
break;
case NUMTYPE_INTERNATIONAL:
number_type = NUMTYPE_INTERNATIONAL;
break;
case NUMTYPE_NATIONAL:
number_type = NUMTYPE_NATIONAL;
break;
case NUMTYPE_NETWORK_SPECIFIC:
number_type = NUMTYPE_NETWORK_SPECIFIC;
break;
case NUMTYPE_SUBSCRIBER:
number_type = NUMTYPE_SUBSCRIBER;
break;
case NUMTYPE_ABBREVIATED:
number_type = NUMTYPE_ABBREVIATED;
break;
}
return number_type;
}
/*!
* \internal
* \brief Convert the mISDN numbering plan code to a string
*
* \param number_plan mISDN numbering plan code.
*
* \return The mISDN numbering plan code as a string
*/
static const char *misdn_to_str_plan(enum mISDN_NUMBER_PLAN number_plan)
{
const char *str;
switch (number_plan) {
default:
case NUMPLAN_UNKNOWN:
str = "Unknown";
break;
case NUMPLAN_ISDN:
str = "ISDN";
break;
case NUMPLAN_DATA:
str = "Data";
break;
case NUMPLAN_TELEX:
str = "Telex";
break;
case NUMPLAN_NATIONAL:
str = "National";
break;
case NUMPLAN_PRIVATE:
str = "Private";
break;
}
return str;
}
/*!
* \internal
* \brief Convert the mISDN numbering plan code to Asterisk numbering plan code
*
* \param number_plan mISDN numbering plan code.
*
* \return Asterisk numbering plan code
*/
static int misdn_to_ast_plan(enum mISDN_NUMBER_PLAN number_plan)
{
int ast_number_plan;
switch (number_plan) {
default:
case NUMPLAN_UNKNOWN:
ast_number_plan = NUMPLAN_UNKNOWN;
break;
case NUMPLAN_ISDN:
ast_number_plan = NUMPLAN_ISDN;
break;
case NUMPLAN_DATA:
ast_number_plan = NUMPLAN_DATA;
break;
case NUMPLAN_TELEX:
ast_number_plan = NUMPLAN_TELEX;
break;
case NUMPLAN_NATIONAL:
ast_number_plan = NUMPLAN_NATIONAL;
break;
case NUMPLAN_PRIVATE:
ast_number_plan = NUMPLAN_PRIVATE;
break;
}
return ast_number_plan;
}
/*!
* \internal
* \brief Convert the Asterisk numbering plan code to mISDN numbering plan code
*
* \param ast_number_plan Asterisk numbering plan code.
*
* \return mISDN numbering plan code
*/
static enum mISDN_NUMBER_PLAN ast_to_misdn_plan(unsigned ast_number_plan)
{
enum mISDN_NUMBER_PLAN number_plan;
switch (ast_number_plan & 0x0F) {
default:
case NUMPLAN_UNKNOWN:
number_plan = NUMPLAN_UNKNOWN;
break;
case NUMPLAN_ISDN:
number_plan = NUMPLAN_ISDN;
break;
case NUMPLAN_DATA:
number_plan = NUMPLAN_DATA;
break;
case NUMPLAN_TELEX:
number_plan = NUMPLAN_TELEX;
break;
case NUMPLAN_NATIONAL:
number_plan = NUMPLAN_NATIONAL;
break;
case NUMPLAN_PRIVATE:
number_plan = NUMPLAN_PRIVATE;
break;
}
return number_plan;
}
/*!
* \internal
* \brief Convert the mISDN presentation code to a string
*
* \param presentation mISDN number presentation restriction code.
*
* \return The mISDN presentation code as a string
*/
static const char *misdn_to_str_pres(int presentation)
{
const char *str;
switch (presentation) {
case 0:
str = "Allowed";
break;
case 1:
str = "Restricted";
break;
case 2:
str = "Unavailable";
break;
default:
str = "Unknown";
break;
}
return str;
}
/*!
* \internal
* \brief Convert the mISDN presentation code to Asterisk presentation code
*
* \param presentation mISDN number presentation restriction code.
*
* \return Asterisk presentation code
*/
static int misdn_to_ast_pres(int presentation)
{
switch (presentation) {
default:
case 0:
presentation = AST_PRES_ALLOWED;
break;
case 1:
presentation = AST_PRES_RESTRICTED;
break;
case 2:
presentation = AST_PRES_UNAVAILABLE;
break;
}
return presentation;
}
/*!
* \internal
* \brief Convert the Asterisk presentation code to mISDN presentation code
*
* \param presentation Asterisk number presentation restriction code.
*
* \return mISDN presentation code
*/
static int ast_to_misdn_pres(int presentation)
{
switch (presentation & AST_PRES_RESTRICTION) {
default:
case AST_PRES_ALLOWED:
presentation = 0;
break;
case AST_PRES_RESTRICTED:
presentation = 1;
break;
case AST_PRES_UNAVAILABLE:
presentation = 2;
break;
}
return presentation;
}
/*!
* \internal
* \brief Convert the mISDN screening code to a string
*
* \param screening mISDN number screening code.
*
* \return The mISDN screening code as a string
*/
static const char *misdn_to_str_screen(int screening)
{
const char *str;
switch (screening) {
case 0:
str = "Unscreened";
break;
case 1:
str = "Passed Screen";
break;
case 2:
str = "Failed Screen";
break;
case 3:
str = "Network Number";
break;
default:
str = "Unknown";
break;
}
return str;
}
/*!
* \internal
* \brief Convert the mISDN screening code to Asterisk screening code
*
* \param screening mISDN number screening code.
*
* \return Asterisk screening code
*/
static int misdn_to_ast_screen(int screening)
{
switch (screening) {
default:
case 0:
screening = AST_PRES_USER_NUMBER_UNSCREENED;
break;
case 1:
screening = AST_PRES_USER_NUMBER_PASSED_SCREEN;
break;
case 2:
screening = AST_PRES_USER_NUMBER_FAILED_SCREEN;
break;
case 3:
screening = AST_PRES_NETWORK_NUMBER;
break;
}
return screening;
}
/*!
* \internal
* \brief Convert the Asterisk screening code to mISDN screening code
*
* \param screening Asterisk number screening code.
*
* \return mISDN screening code
*/
static int ast_to_misdn_screen(int screening)
{
switch (screening & AST_PRES_NUMBER_TYPE) {
default:
case AST_PRES_USER_NUMBER_UNSCREENED:
screening = 0;
break;
case AST_PRES_USER_NUMBER_PASSED_SCREEN:
screening = 1;
break;
case AST_PRES_USER_NUMBER_FAILED_SCREEN:
screening = 2;
break;
case AST_PRES_NETWORK_NUMBER:
screening = 3;
break;
}
return screening;
}
/*!
* \internal
* \brief Convert Asterisk redirecting reason to mISDN redirecting reason code.
*
* \param ast Asterisk redirecting reason code.
*
* \return mISDN reason code
*/
static enum mISDN_REDIRECTING_REASON ast_to_misdn_reason(const enum AST_REDIRECTING_REASON ast)
{
unsigned index;
static const struct misdn_reasons {
enum AST_REDIRECTING_REASON ast;
enum mISDN_REDIRECTING_REASON q931;
} misdn_reason_table[] = {
/* *INDENT-OFF* */
{ AST_REDIRECTING_REASON_UNKNOWN, mISDN_REDIRECTING_REASON_UNKNOWN },
{ AST_REDIRECTING_REASON_USER_BUSY, mISDN_REDIRECTING_REASON_CALL_FWD_BUSY },
{ AST_REDIRECTING_REASON_NO_ANSWER, mISDN_REDIRECTING_REASON_NO_REPLY },
{ AST_REDIRECTING_REASON_UNAVAILABLE, mISDN_REDIRECTING_REASON_NO_REPLY },
{ AST_REDIRECTING_REASON_UNCONDITIONAL, mISDN_REDIRECTING_REASON_CALL_FWD },
{ AST_REDIRECTING_REASON_TIME_OF_DAY, mISDN_REDIRECTING_REASON_UNKNOWN },
{ AST_REDIRECTING_REASON_DO_NOT_DISTURB, mISDN_REDIRECTING_REASON_UNKNOWN },
{ AST_REDIRECTING_REASON_DEFLECTION, mISDN_REDIRECTING_REASON_DEFLECTION },
{ AST_REDIRECTING_REASON_FOLLOW_ME, mISDN_REDIRECTING_REASON_UNKNOWN },
{ AST_REDIRECTING_REASON_OUT_OF_ORDER, mISDN_REDIRECTING_REASON_OUT_OF_ORDER },
{ AST_REDIRECTING_REASON_AWAY, mISDN_REDIRECTING_REASON_UNKNOWN },
{ AST_REDIRECTING_REASON_CALL_FWD_DTE, mISDN_REDIRECTING_REASON_CALL_FWD_DTE }
/* *INDENT-ON* */
};
for (index = 0; index < ARRAY_LEN(misdn_reason_table); ++index) {
if (misdn_reason_table[index].ast == ast) {
return misdn_reason_table[index].q931;
}
}
return mISDN_REDIRECTING_REASON_UNKNOWN;
}
/*!
* \internal
* \brief Convert the mISDN redirecting reason to Asterisk redirecting reason code
*
* \param q931 mISDN redirecting reason code.
*
* \return Asterisk redirecting reason code
*/
static enum AST_REDIRECTING_REASON misdn_to_ast_reason(const enum mISDN_REDIRECTING_REASON q931)
{
enum AST_REDIRECTING_REASON ast;
switch (q931) {
default:
case mISDN_REDIRECTING_REASON_UNKNOWN:
ast = AST_REDIRECTING_REASON_UNKNOWN;
break;
case mISDN_REDIRECTING_REASON_CALL_FWD_BUSY:
ast = AST_REDIRECTING_REASON_USER_BUSY;
break;
case mISDN_REDIRECTING_REASON_NO_REPLY:
ast = AST_REDIRECTING_REASON_NO_ANSWER;
break;
case mISDN_REDIRECTING_REASON_DEFLECTION:
ast = AST_REDIRECTING_REASON_DEFLECTION;
break;
case mISDN_REDIRECTING_REASON_OUT_OF_ORDER:
ast = AST_REDIRECTING_REASON_OUT_OF_ORDER;
break;
case mISDN_REDIRECTING_REASON_CALL_FWD_DTE:
ast = AST_REDIRECTING_REASON_CALL_FWD_DTE;
break;
case mISDN_REDIRECTING_REASON_CALL_FWD:
ast = AST_REDIRECTING_REASON_UNCONDITIONAL;
break;
}
return ast;
}
struct allowed_bearers {
char *name; /*!< Bearer capability name string used in /etc/misdn.conf allowed_bearers */
char *display; /*!< Bearer capability displayable name */
int cap; /*!< SETUP message bearer capability field code value */
int deprecated; /*!< TRUE if this entry is deprecated. (Misspelled or bad name to use) */
};
/* *INDENT-OFF* */
static const struct allowed_bearers allowed_bearers_array[] = {
/* Name, Displayable Name Bearer Capability, Deprecated */
{ "speech", "Speech", INFO_CAPABILITY_SPEECH, 0 },
{ "3_1khz", "3.1KHz Audio", INFO_CAPABILITY_AUDIO_3_1K, 0 },
{ "digital_unrestricted", "Unrestricted Digital", INFO_CAPABILITY_DIGITAL_UNRESTRICTED, 0 },
{ "digital_restricted", "Restricted Digital", INFO_CAPABILITY_DIGITAL_RESTRICTED, 0 },
{ "digital_restriced", "Restricted Digital", INFO_CAPABILITY_DIGITAL_RESTRICTED, 1 }, /* Allow misspelling for backwards compatibility */
{ "video", "Video", INFO_CAPABILITY_VIDEO, 0 }
};
/* *INDENT-ON* */
static const char *bearer2str(int cap)
{
unsigned index;
for (index = 0; index < ARRAY_LEN(allowed_bearers_array); ++index) {
if (allowed_bearers_array[index].cap == cap) {
return allowed_bearers_array[index].display;
}
}
return "Unknown Bearer";
}
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Fill in facility PartyNumber information
*
* \param party PartyNumber structure to fill in.
* \param id Information to put in PartyNumber structure.
*
* \return Nothing
*/
static void misdn_PartyNumber_fill(struct FacPartyNumber *party, const struct misdn_party_id *id)
{
ast_copy_string((char *) party->Number, id->number, sizeof(party->Number));
party->LengthOfNumber = strlen((char *) party->Number);
party->Type = misdn_to_PartyNumber_plan(id->number_plan);
switch (party->Type) {
case 1:/* public */
party->TypeOfNumber = misdn_to_PartyNumber_ton_public(id->number_type);
break;
case 5:/* private */
party->TypeOfNumber = misdn_to_PartyNumber_ton_private(id->number_type);
break;
default:
party->TypeOfNumber = 0;/* Don't care */
break;
}
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Extract the information from PartyNumber
*
* \param id Where to put extracted PartyNumber information
* \param party PartyNumber information to extract
*
* \return Nothing
*/
static void misdn_PartyNumber_extract(struct misdn_party_id *id, const struct FacPartyNumber *party)
{
if (party->LengthOfNumber) {
ast_copy_string(id->number, (char *) party->Number, sizeof(id->number));
id->number_plan = PartyNumber_to_misdn_plan(party->Type);
switch (party->Type) {
case 1:/* public */
id->number_type = PartyNumber_to_misdn_ton_public(party->TypeOfNumber);
break;
case 5:/* private */
id->number_type = PartyNumber_to_misdn_ton_private(party->TypeOfNumber);
break;
default:
id->number_type = NUMTYPE_UNKNOWN;
break;
}
} else {
/* Number not present */
id->number_type = NUMTYPE_UNKNOWN;
id->number_plan = NUMPLAN_ISDN;
id->number[0] = 0;
}
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Fill in facility Address information
*
* \param Address Address structure to fill in.
* \param id Information to put in Address structure.
*
* \return Nothing
*/
static void misdn_Address_fill(struct FacAddress *Address, const struct misdn_party_id *id)
{
misdn_PartyNumber_fill(&Address->Party, id);
/* Subaddresses are not supported yet */
Address->Subaddress.Length = 0;
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Fill in facility PresentedNumberUnscreened information
*
* \param presented PresentedNumberUnscreened structure to fill in.
* \param id Information to put in PresentedNumberUnscreened structure.
*
* \return Nothing
*/
static void misdn_PresentedNumberUnscreened_fill(struct FacPresentedNumberUnscreened *presented, const struct misdn_party_id *id)
{
presented->Type = misdn_to_PresentedNumberUnscreened_type(id->presentation, id->number[0] ? 1 : 0);
misdn_PartyNumber_fill(&presented->Unscreened, id);
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Extract the information from PartyNumber
*
* \param id Where to put extracted PresentedNumberUnscreened information
* \param presented PresentedNumberUnscreened information to extract
*
* \return Nothing
*/
static void misdn_PresentedNumberUnscreened_extract(struct misdn_party_id *id, const struct FacPresentedNumberUnscreened *presented)
{
id->presentation = PresentedNumberUnscreened_to_misdn_pres(presented->Type);
id->screening = 0;/* unscreened */
switch (presented->Type) {
case 0:/* presentationAllowedNumber */
case 3:/* presentationRestrictedNumber */
misdn_PartyNumber_extract(id, &presented->Unscreened);
break;
case 1:/* presentationRestricted */
case 2:/* numberNotAvailableDueToInterworking */
default:
/* Number not present (And uninitialized so do not even look at it!) */
id->number_type = NUMTYPE_UNKNOWN;
id->number_plan = NUMPLAN_ISDN;
id->number[0] = 0;
break;
}
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
static const char Level_Spacing[] = " ";/* Work for up to 10 levels */
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
static void print_facility_PartyNumber(unsigned Level, const struct FacPartyNumber *Party, const struct misdn_bchannel *bc)
{
if (Party->LengthOfNumber) {
const char *Spacing;
Spacing = &Level_Spacing[sizeof(Level_Spacing) - 1 - Level];
chan_misdn_log(1, bc->port, " -->%s PartyNumber: Type:%d\n",
Spacing, Party->Type);
switch (Party->Type) {
case 0: /* Unknown PartyNumber */
chan_misdn_log(1, bc->port, " -->%s Unknown: %s\n",
Spacing, Party->Number);
break;
case 1: /* Public PartyNumber */
chan_misdn_log(1, bc->port, " -->%s Public TON:%d %s\n",
Spacing, Party->TypeOfNumber, Party->Number);
break;
case 2: /* NSAP encoded PartyNumber */
chan_misdn_log(1, bc->port, " -->%s NSAP: %s\n",
Spacing, Party->Number);
break;
case 3: /* Data PartyNumber (Not used) */
chan_misdn_log(1, bc->port, " -->%s Data: %s\n",
Spacing, Party->Number);
break;
case 4: /* Telex PartyNumber (Not used) */
chan_misdn_log(1, bc->port, " -->%s Telex: %s\n",
Spacing, Party->Number);
break;
case 5: /* Private PartyNumber */
chan_misdn_log(1, bc->port, " -->%s Private TON:%d %s\n",
Spacing, Party->TypeOfNumber, Party->Number);
break;
case 8: /* National Standard PartyNumber (Not used) */
chan_misdn_log(1, bc->port, " -->%s National: %s\n",
Spacing, Party->Number);
break;
default:
break;
}
}
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
static void print_facility_Subaddress(unsigned Level, const struct FacPartySubaddress *Subaddress, const struct misdn_bchannel *bc)
{
if (Subaddress->Length) {
const char *Spacing;
Spacing = &Level_Spacing[sizeof(Level_Spacing) - 1 - Level];
chan_misdn_log(1, bc->port, " -->%s Subaddress: Type:%d\n",
Spacing, Subaddress->Type);
switch (Subaddress->Type) {
case 0: /* UserSpecified */
if (Subaddress->u.UserSpecified.OddCountPresent) {
chan_misdn_log(1, bc->port, " -->%s User BCD OddCount:%d NumOctets:%d\n",
Spacing, Subaddress->u.UserSpecified.OddCount, Subaddress->Length);
} else {
chan_misdn_log(1, bc->port, " -->%s User: %s\n",
Spacing, Subaddress->u.UserSpecified.Information);
}
break;
case 1: /* NSAP */
chan_misdn_log(1, bc->port, " -->%s NSAP: %s\n",
Spacing, Subaddress->u.Nsap);
break;
default:
break;
}
}
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
static void print_facility_Address(unsigned Level, const struct FacAddress *Address, const struct misdn_bchannel *bc)
{
print_facility_PartyNumber(Level, &Address->Party, bc);
print_facility_Subaddress(Level, &Address->Subaddress, bc);
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
static void print_facility_PresentedNumberUnscreened(unsigned Level, const struct FacPresentedNumberUnscreened *Presented, const struct misdn_bchannel *bc)
{
const char *Spacing;
Spacing = &Level_Spacing[sizeof(Level_Spacing) - 1 - Level];
chan_misdn_log(1, bc->port, " -->%s Unscreened Type:%d\n", Spacing, Presented->Type);
switch (Presented->Type) {
case 0: /* presentationAllowedNumber */
chan_misdn_log(1, bc->port, " -->%s Allowed:\n", Spacing);
print_facility_PartyNumber(Level + 2, &Presented->Unscreened, bc);
break;
case 1: /* presentationRestricted */
chan_misdn_log(1, bc->port, " -->%s Restricted\n", Spacing);
break;
case 2: /* numberNotAvailableDueToInterworking */
chan_misdn_log(1, bc->port, " -->%s Not Available\n", Spacing);
break;
case 3: /* presentationRestrictedNumber */
chan_misdn_log(1, bc->port, " -->%s Restricted:\n", Spacing);
print_facility_PartyNumber(Level + 2, &Presented->Unscreened, bc);
break;
default:
break;
}
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
static void print_facility_AddressScreened(unsigned Level, const struct FacAddressScreened *Address, const struct misdn_bchannel *bc)
{
const char *Spacing;
Spacing = &Level_Spacing[sizeof(Level_Spacing) - 1 - Level];
chan_misdn_log(1, bc->port, " -->%s ScreeningIndicator:%d\n", Spacing, Address->ScreeningIndicator);
print_facility_PartyNumber(Level, &Address->Party, bc);
print_facility_Subaddress(Level, &Address->Subaddress, bc);
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
static void print_facility_PresentedAddressScreened(unsigned Level, const struct FacPresentedAddressScreened *Presented, const struct misdn_bchannel *bc)
{
const char *Spacing;
Spacing = &Level_Spacing[sizeof(Level_Spacing) - 1 - Level];
chan_misdn_log(1, bc->port, " -->%s Screened Type:%d\n", Spacing, Presented->Type);
switch (Presented->Type) {
case 0: /* presentationAllowedAddress */
chan_misdn_log(1, bc->port, " -->%s Allowed:\n", Spacing);
print_facility_AddressScreened(Level + 2, &Presented->Address, bc);
break;
case 1: /* presentationRestricted */
chan_misdn_log(1, bc->port, " -->%s Restricted\n", Spacing);
break;
case 2: /* numberNotAvailableDueToInterworking */
chan_misdn_log(1, bc->port, " -->%s Not Available\n", Spacing);
break;
case 3: /* presentationRestrictedAddress */
chan_misdn_log(1, bc->port, " -->%s Restricted:\n", Spacing);
print_facility_AddressScreened(Level + 2, &Presented->Address, bc);
break;
default:
break;
}
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
static void print_facility_Q931_Bc_Hlc_Llc(unsigned Level, const struct Q931_Bc_Hlc_Llc *Q931ie, const struct misdn_bchannel *bc)
{
const char *Spacing;
Spacing = &Level_Spacing[sizeof(Level_Spacing) - 1 - Level];
chan_misdn_log(1, bc->port, " -->%s Q931ie:\n", Spacing);
if (Q931ie->Bc.Length) {
chan_misdn_log(1, bc->port, " -->%s Bc Len:%d\n", Spacing, Q931ie->Bc.Length);
}
if (Q931ie->Hlc.Length) {
chan_misdn_log(1, bc->port, " -->%s Hlc Len:%d\n", Spacing, Q931ie->Hlc.Length);
}
if (Q931ie->Llc.Length) {
chan_misdn_log(1, bc->port, " -->%s Llc Len:%d\n", Spacing, Q931ie->Llc.Length);
}
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
static void print_facility_Q931_Bc_Hlc_Llc_Uu(unsigned Level, const struct Q931_Bc_Hlc_Llc_Uu *Q931ie, const struct misdn_bchannel *bc)
{
const char *Spacing;
Spacing = &Level_Spacing[sizeof(Level_Spacing) - 1 - Level];
chan_misdn_log(1, bc->port, " -->%s Q931ie:\n", Spacing);
if (Q931ie->Bc.Length) {
chan_misdn_log(1, bc->port, " -->%s Bc Len:%d\n", Spacing, Q931ie->Bc.Length);
}
if (Q931ie->Hlc.Length) {
chan_misdn_log(1, bc->port, " -->%s Hlc Len:%d\n", Spacing, Q931ie->Hlc.Length);
}
if (Q931ie->Llc.Length) {
chan_misdn_log(1, bc->port, " -->%s Llc Len:%d\n", Spacing, Q931ie->Llc.Length);
}
if (Q931ie->UserInfo.Length) {
chan_misdn_log(1, bc->port, " -->%s UserInfo Len:%d\n", Spacing, Q931ie->UserInfo.Length);
}
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
static void print_facility_CallInformation(unsigned Level, const struct FacCallInformation *CallInfo, const struct misdn_bchannel *bc)
{
const char *Spacing;
Spacing = &Level_Spacing[sizeof(Level_Spacing) - 1 - Level];
chan_misdn_log(1, bc->port, " -->%s CCBSReference:%d\n",
Spacing, CallInfo->CCBSReference);
chan_misdn_log(1, bc->port, " -->%s AddressOfB:\n", Spacing);
print_facility_Address(Level + 1, &CallInfo->AddressOfB, bc);
print_facility_Q931_Bc_Hlc_Llc(Level, &CallInfo->Q931ie, bc);
if (CallInfo->SubaddressOfA.Length) {
chan_misdn_log(1, bc->port, " -->%s SubaddressOfA:\n", Spacing);
print_facility_Subaddress(Level + 1, &CallInfo->SubaddressOfA, bc);
}
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
static void print_facility_ServedUserNr(unsigned Level, const struct FacPartyNumber *Party, const struct misdn_bchannel *bc)
{
const char *Spacing;
Spacing = &Level_Spacing[sizeof(Level_Spacing) - 1 - Level];
if (Party->LengthOfNumber) {
print_facility_PartyNumber(Level, Party, bc);
} else {
chan_misdn_log(1, bc->port, " -->%s All Numbers\n", Spacing);
}
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
static void print_facility_IntResult(unsigned Level, const struct FacForwardingRecord *ForwardingRecord, const struct misdn_bchannel *bc)
{
const char *Spacing;
Spacing = &Level_Spacing[sizeof(Level_Spacing) - 1 - Level];
chan_misdn_log(1, bc->port, " -->%s Procedure:%d BasicService:%d\n",
Spacing,
ForwardingRecord->Procedure,
ForwardingRecord->BasicService);
chan_misdn_log(1, bc->port, " -->%s ForwardedTo:\n", Spacing);
print_facility_Address(Level + 1, &ForwardingRecord->ForwardedTo, bc);
chan_misdn_log(1, bc->port, " -->%s ServedUserNr:\n", Spacing);
print_facility_ServedUserNr(Level + 1, &ForwardingRecord->ServedUser, bc);
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
static void print_facility(const struct FacParm *fac, const struct misdn_bchannel *bc)
{
#if defined(AST_MISDN_ENHANCEMENTS)
unsigned Index;
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
switch (fac->Function) {
#if defined(AST_MISDN_ENHANCEMENTS)
case Fac_ActivationDiversion:
chan_misdn_log(1, bc->port, " --> ActivationDiversion: InvokeID:%d\n",
fac->u.ActivationDiversion.InvokeID);
switch (fac->u.ActivationDiversion.ComponentType) {
case FacComponent_Invoke:
chan_misdn_log(1, bc->port, " --> Invoke: Procedure:%d BasicService:%d\n",
fac->u.ActivationDiversion.Component.Invoke.Procedure,
fac->u.ActivationDiversion.Component.Invoke.BasicService);
chan_misdn_log(1, bc->port, " --> ForwardedTo:\n");
print_facility_Address(3, &fac->u.ActivationDiversion.Component.Invoke.ForwardedTo, bc);
chan_misdn_log(1, bc->port, " --> ServedUserNr:\n");
print_facility_ServedUserNr(3, &fac->u.ActivationDiversion.Component.Invoke.ServedUser, bc);
break;
case FacComponent_Result:
chan_misdn_log(1, bc->port, " --> Result\n");
break;
default:
break;
}
break;
case Fac_DeactivationDiversion:
chan_misdn_log(1, bc->port, " --> DeactivationDiversion: InvokeID:%d\n",
fac->u.DeactivationDiversion.InvokeID);
switch (fac->u.DeactivationDiversion.ComponentType) {
case FacComponent_Invoke:
chan_misdn_log(1, bc->port, " --> Invoke: Procedure:%d BasicService:%d\n",
fac->u.DeactivationDiversion.Component.Invoke.Procedure,
fac->u.DeactivationDiversion.Component.Invoke.BasicService);
chan_misdn_log(1, bc->port, " --> ServedUserNr:\n");
print_facility_ServedUserNr(3, &fac->u.DeactivationDiversion.Component.Invoke.ServedUser, bc);
break;
case FacComponent_Result:
chan_misdn_log(1, bc->port, " --> Result\n");
break;
default:
break;
}
break;
case Fac_ActivationStatusNotificationDiv:
chan_misdn_log(1, bc->port, " --> ActivationStatusNotificationDiv: InvokeID:%d Procedure:%d BasicService:%d\n",
fac->u.ActivationStatusNotificationDiv.InvokeID,
fac->u.ActivationStatusNotificationDiv.Procedure,
fac->u.ActivationStatusNotificationDiv.BasicService);
chan_misdn_log(1, bc->port, " --> ForwardedTo:\n");
print_facility_Address(2, &fac->u.ActivationStatusNotificationDiv.ForwardedTo, bc);
chan_misdn_log(1, bc->port, " --> ServedUserNr:\n");
print_facility_ServedUserNr(2, &fac->u.ActivationStatusNotificationDiv.ServedUser, bc);
break;
case Fac_DeactivationStatusNotificationDiv:
chan_misdn_log(1, bc->port, " --> DeactivationStatusNotificationDiv: InvokeID:%d Procedure:%d BasicService:%d\n",
fac->u.DeactivationStatusNotificationDiv.InvokeID,
fac->u.DeactivationStatusNotificationDiv.Procedure,
fac->u.DeactivationStatusNotificationDiv.BasicService);
chan_misdn_log(1, bc->port, " --> ServedUserNr:\n");
print_facility_ServedUserNr(2, &fac->u.DeactivationStatusNotificationDiv.ServedUser, bc);
break;
case Fac_InterrogationDiversion:
chan_misdn_log(1, bc->port, " --> InterrogationDiversion: InvokeID:%d\n",
fac->u.InterrogationDiversion.InvokeID);
switch (fac->u.InterrogationDiversion.ComponentType) {
case FacComponent_Invoke:
chan_misdn_log(1, bc->port, " --> Invoke: Procedure:%d BasicService:%d\n",
fac->u.InterrogationDiversion.Component.Invoke.Procedure,
fac->u.InterrogationDiversion.Component.Invoke.BasicService);
chan_misdn_log(1, bc->port, " --> ServedUserNr:\n");
print_facility_ServedUserNr(3, &fac->u.InterrogationDiversion.Component.Invoke.ServedUser, bc);
break;
case FacComponent_Result:
chan_misdn_log(1, bc->port, " --> Result:\n");
if (fac->u.InterrogationDiversion.Component.Result.NumRecords) {
for (Index = 0; Index < fac->u.InterrogationDiversion.Component.Result.NumRecords; ++Index) {
chan_misdn_log(1, bc->port, " --> IntResult[%d]:\n", Index);
print_facility_IntResult(3, &fac->u.InterrogationDiversion.Component.Result.List[Index], bc);
}
}
break;
default:
break;
}
break;
case Fac_DiversionInformation:
chan_misdn_log(1, bc->port, " --> DiversionInformation: InvokeID:%d Reason:%d BasicService:%d\n",
fac->u.DiversionInformation.InvokeID,
fac->u.DiversionInformation.DiversionReason,
fac->u.DiversionInformation.BasicService);
if (fac->u.DiversionInformation.ServedUserSubaddress.Length) {
chan_misdn_log(1, bc->port, " --> ServedUserSubaddress:\n");
print_facility_Subaddress(2, &fac->u.DiversionInformation.ServedUserSubaddress, bc);
}
if (fac->u.DiversionInformation.CallingAddressPresent) {
chan_misdn_log(1, bc->port, " --> CallingAddress:\n");
print_facility_PresentedAddressScreened(2, &fac->u.DiversionInformation.CallingAddress, bc);
}
if (fac->u.DiversionInformation.OriginalCalledPresent) {
chan_misdn_log(1, bc->port, " --> OriginalCalledNr:\n");
print_facility_PresentedNumberUnscreened(2, &fac->u.DiversionInformation.OriginalCalled, bc);
}
if (fac->u.DiversionInformation.LastDivertingPresent) {
chan_misdn_log(1, bc->port, " --> LastDivertingNr:\n");
print_facility_PresentedNumberUnscreened(2, &fac->u.DiversionInformation.LastDiverting, bc);
}
if (fac->u.DiversionInformation.LastDivertingReasonPresent) {
chan_misdn_log(1, bc->port, " --> LastDivertingReason:%d\n", fac->u.DiversionInformation.LastDivertingReason);
}
if (fac->u.DiversionInformation.UserInfo.Length) {
chan_misdn_log(1, bc->port, " --> UserInfo Length:%d\n", fac->u.DiversionInformation.UserInfo.Length);
}
break;
case Fac_CallDeflection:
chan_misdn_log(1, bc->port, " --> CallDeflection: InvokeID:%d\n",
fac->u.CallDeflection.InvokeID);
switch (fac->u.CallDeflection.ComponentType) {
case FacComponent_Invoke:
chan_misdn_log(1, bc->port, " --> Invoke:\n");
if (fac->u.CallDeflection.Component.Invoke.PresentationAllowedToDivertedToUserPresent) {
chan_misdn_log(1, bc->port, " --> PresentationAllowed:%d\n",
fac->u.CallDeflection.Component.Invoke.PresentationAllowedToDivertedToUser);
}
chan_misdn_log(1, bc->port, " --> DeflectionAddress:\n");
print_facility_Address(3, &fac->u.CallDeflection.Component.Invoke.Deflection, bc);
break;
case FacComponent_Result:
chan_misdn_log(1, bc->port, " --> Result\n");
break;
default:
break;
}
break;
case Fac_CallRerouteing:
chan_misdn_log(1, bc->port, " --> CallRerouteing: InvokeID:%d\n",
fac->u.CallRerouteing.InvokeID);
switch (fac->u.CallRerouteing.ComponentType) {
case FacComponent_Invoke:
chan_misdn_log(1, bc->port, " --> Invoke: Reason:%d Counter:%d\n",
fac->u.CallRerouteing.Component.Invoke.ReroutingReason,
fac->u.CallRerouteing.Component.Invoke.ReroutingCounter);
chan_misdn_log(1, bc->port, " --> CalledAddress:\n");
print_facility_Address(3, &fac->u.CallRerouteing.Component.Invoke.CalledAddress, bc);
print_facility_Q931_Bc_Hlc_Llc_Uu(2, &fac->u.CallRerouteing.Component.Invoke.Q931ie, bc);
chan_misdn_log(1, bc->port, " --> LastReroutingNr:\n");
print_facility_PresentedNumberUnscreened(3, &fac->u.CallRerouteing.Component.Invoke.LastRerouting, bc);
chan_misdn_log(1, bc->port, " --> SubscriptionOption:%d\n",
fac->u.CallRerouteing.Component.Invoke.SubscriptionOption);
if (fac->u.CallRerouteing.Component.Invoke.CallingPartySubaddress.Length) {
chan_misdn_log(1, bc->port, " --> CallingParty:\n");
print_facility_Subaddress(3, &fac->u.CallRerouteing.Component.Invoke.CallingPartySubaddress, bc);
}
break;
case FacComponent_Result:
chan_misdn_log(1, bc->port, " --> Result\n");
break;
default:
break;
}
break;
case Fac_InterrogateServedUserNumbers:
chan_misdn_log(1, bc->port, " --> InterrogateServedUserNumbers: InvokeID:%d\n",
fac->u.InterrogateServedUserNumbers.InvokeID);
switch (fac->u.InterrogateServedUserNumbers.ComponentType) {
case FacComponent_Invoke:
chan_misdn_log(1, bc->port, " --> Invoke\n");
break;
case FacComponent_Result:
chan_misdn_log(1, bc->port, " --> Result:\n");
if (fac->u.InterrogateServedUserNumbers.Component.Result.NumRecords) {
for (Index = 0; Index < fac->u.InterrogateServedUserNumbers.Component.Result.NumRecords; ++Index) {
chan_misdn_log(1, bc->port, " --> ServedUserNr[%d]:\n", Index);
print_facility_PartyNumber(3, &fac->u.InterrogateServedUserNumbers.Component.Result.List[Index], bc);
}
}
break;
default:
break;
}
break;
case Fac_DivertingLegInformation1:
chan_misdn_log(1, bc->port, " --> DivertingLegInformation1: InvokeID:%d Reason:%d SubscriptionOption:%d\n",
fac->u.DivertingLegInformation1.InvokeID,
fac->u.DivertingLegInformation1.DiversionReason,
fac->u.DivertingLegInformation1.SubscriptionOption);
if (fac->u.DivertingLegInformation1.DivertedToPresent) {
chan_misdn_log(1, bc->port, " --> DivertedToNr:\n");
print_facility_PresentedNumberUnscreened(2, &fac->u.DivertingLegInformation1.DivertedTo, bc);
}
break;
case Fac_DivertingLegInformation2:
chan_misdn_log(1, bc->port, " --> DivertingLegInformation2: InvokeID:%d Reason:%d Count:%d\n",
fac->u.DivertingLegInformation2.InvokeID,
fac->u.DivertingLegInformation2.DiversionReason,
fac->u.DivertingLegInformation2.DiversionCounter);
if (fac->u.DivertingLegInformation2.DivertingPresent) {
chan_misdn_log(1, bc->port, " --> DivertingNr:\n");
print_facility_PresentedNumberUnscreened(2, &fac->u.DivertingLegInformation2.Diverting, bc);
}
if (fac->u.DivertingLegInformation2.OriginalCalledPresent) {
chan_misdn_log(1, bc->port, " --> OriginalCalledNr:\n");
print_facility_PresentedNumberUnscreened(2, &fac->u.DivertingLegInformation2.OriginalCalled, bc);
}
break;
case Fac_DivertingLegInformation3:
chan_misdn_log(1, bc->port, " --> DivertingLegInformation3: InvokeID:%d PresentationAllowed:%d\n",
fac->u.DivertingLegInformation3.InvokeID,
fac->u.DivertingLegInformation3.PresentationAllowedIndicator);
break;
#else /* !defined(AST_MISDN_ENHANCEMENTS) */
case Fac_CD:
chan_misdn_log(1, bc->port, " --> calldeflect to: %s, presentable: %s\n", fac->u.CDeflection.DeflectedToNumber,
fac->u.CDeflection.PresentationAllowed ? "yes" : "no");
break;
#endif /* !defined(AST_MISDN_ENHANCEMENTS) */
case Fac_AOCDCurrency:
if (fac->u.AOCDcur.chargeNotAvailable) {
chan_misdn_log(1, bc->port, " --> AOCD currency: charge not available\n");
} else if (fac->u.AOCDcur.freeOfCharge) {
chan_misdn_log(1, bc->port, " --> AOCD currency: free of charge\n");
} else if (fac->u.AOCDchu.billingId >= 0) {
chan_misdn_log(1, bc->port, " --> AOCD currency: currency:%s amount:%d multiplier:%d typeOfChargingInfo:%s billingId:%d\n",
fac->u.AOCDcur.currency, fac->u.AOCDcur.currencyAmount, fac->u.AOCDcur.multiplier,
(fac->u.AOCDcur.typeOfChargingInfo == 0) ? "subTotal" : "total", fac->u.AOCDcur.billingId);
} else {
chan_misdn_log(1, bc->port, " --> AOCD currency: currency:%s amount:%d multiplier:%d typeOfChargingInfo:%s\n",
fac->u.AOCDcur.currency, fac->u.AOCDcur.currencyAmount, fac->u.AOCDcur.multiplier,
(fac->u.AOCDcur.typeOfChargingInfo == 0) ? "subTotal" : "total");
}
break;
case Fac_AOCDChargingUnit:
if (fac->u.AOCDchu.chargeNotAvailable) {
chan_misdn_log(1, bc->port, " --> AOCD charging unit: charge not available\n");
} else if (fac->u.AOCDchu.freeOfCharge) {
chan_misdn_log(1, bc->port, " --> AOCD charging unit: free of charge\n");
} else if (fac->u.AOCDchu.billingId >= 0) {
chan_misdn_log(1, bc->port, " --> AOCD charging unit: recordedUnits:%d typeOfChargingInfo:%s billingId:%d\n",
fac->u.AOCDchu.recordedUnits, (fac->u.AOCDchu.typeOfChargingInfo == 0) ? "subTotal" : "total", fac->u.AOCDchu.billingId);
} else {
chan_misdn_log(1, bc->port, " --> AOCD charging unit: recordedUnits:%d typeOfChargingInfo:%s\n",
fac->u.AOCDchu.recordedUnits, (fac->u.AOCDchu.typeOfChargingInfo == 0) ? "subTotal" : "total");
}
break;
#if defined(AST_MISDN_ENHANCEMENTS)
case Fac_ERROR:
chan_misdn_log(1, bc->port, " --> ERROR: InvokeID:%d, Code:0x%02x\n",
fac->u.ERROR.invokeId, fac->u.ERROR.errorValue);
break;
case Fac_RESULT:
chan_misdn_log(1, bc->port, " --> RESULT: InvokeID:%d\n",
fac->u.RESULT.InvokeID);
break;
case Fac_REJECT:
if (fac->u.REJECT.InvokeIDPresent) {
chan_misdn_log(1, bc->port, " --> REJECT: InvokeID:%d, Code:0x%02x\n",
fac->u.REJECT.InvokeID, fac->u.REJECT.Code);
} else {
chan_misdn_log(1, bc->port, " --> REJECT: Code:0x%02x\n",
fac->u.REJECT.Code);
}
break;
case Fac_EctExecute:
chan_misdn_log(1, bc->port, " --> EctExecute: InvokeID:%d\n",
fac->u.EctExecute.InvokeID);
break;
case Fac_ExplicitEctExecute:
chan_misdn_log(1, bc->port, " --> ExplicitEctExecute: InvokeID:%d LinkID:%d\n",
fac->u.ExplicitEctExecute.InvokeID,
fac->u.ExplicitEctExecute.LinkID);
break;
case Fac_RequestSubaddress:
chan_misdn_log(1, bc->port, " --> RequestSubaddress: InvokeID:%d\n",
fac->u.RequestSubaddress.InvokeID);
break;
case Fac_SubaddressTransfer:
chan_misdn_log(1, bc->port, " --> SubaddressTransfer: InvokeID:%d\n",
fac->u.SubaddressTransfer.InvokeID);
print_facility_Subaddress(1, &fac->u.SubaddressTransfer.Subaddress, bc);
break;
case Fac_EctLinkIdRequest:
chan_misdn_log(1, bc->port, " --> EctLinkIdRequest: InvokeID:%d\n",
fac->u.EctLinkIdRequest.InvokeID);
switch (fac->u.EctLinkIdRequest.ComponentType) {
case FacComponent_Invoke:
chan_misdn_log(1, bc->port, " --> Invoke\n");
break;
case FacComponent_Result:
chan_misdn_log(1, bc->port, " --> Result: LinkID:%d\n",
fac->u.EctLinkIdRequest.Component.Result.LinkID);
break;
default:
break;
}
break;
case Fac_EctInform:
chan_misdn_log(1, bc->port, " --> EctInform: InvokeID:%d Status:%d\n",
fac->u.EctInform.InvokeID,
fac->u.EctInform.Status);
if (fac->u.EctInform.RedirectionPresent) {
chan_misdn_log(1, bc->port, " --> Redirection Number\n");
print_facility_PresentedNumberUnscreened(2, &fac->u.EctInform.Redirection, bc);
}
break;
case Fac_EctLoopTest:
chan_misdn_log(1, bc->port, " --> EctLoopTest: InvokeID:%d\n",
fac->u.EctLoopTest.InvokeID);
switch (fac->u.EctLoopTest.ComponentType) {
case FacComponent_Invoke:
chan_misdn_log(1, bc->port, " --> Invoke: CallTransferID:%d\n",
fac->u.EctLoopTest.Component.Invoke.CallTransferID);
break;
case FacComponent_Result:
chan_misdn_log(1, bc->port, " --> Result: LoopResult:%d\n",
fac->u.EctLoopTest.Component.Result.LoopResult);
break;
default:
break;
}
break;
case Fac_StatusRequest:
chan_misdn_log(1, bc->port, " --> StatusRequest: InvokeID:%d\n",
fac->u.StatusRequest.InvokeID);
switch (fac->u.StatusRequest.ComponentType) {
case FacComponent_Invoke:
chan_misdn_log(1, bc->port, " --> Invoke: Compatibility:%d\n",
fac->u.StatusRequest.Component.Invoke.CompatibilityMode);
break;
case FacComponent_Result:
chan_misdn_log(1, bc->port, " --> Result: Status:%d\n",
fac->u.StatusRequest.Component.Result.Status);
break;
default:
break;
}
break;
case Fac_CallInfoRetain:
chan_misdn_log(1, bc->port, " --> CallInfoRetain: InvokeID:%d, LinkageID:%d\n",
fac->u.CallInfoRetain.InvokeID, fac->u.CallInfoRetain.CallLinkageID);
break;
case Fac_CCBSDeactivate:
chan_misdn_log(1, bc->port, " --> CCBSDeactivate: InvokeID:%d\n",
fac->u.CCBSDeactivate.InvokeID);
switch (fac->u.CCBSDeactivate.ComponentType) {
case FacComponent_Invoke:
chan_misdn_log(1, bc->port, " --> Invoke: CCBSReference:%d\n",
fac->u.CCBSDeactivate.Component.Invoke.CCBSReference);
break;
case FacComponent_Result:
chan_misdn_log(1, bc->port, " --> Result\n");
break;
default:
break;
}
break;
case Fac_CCBSErase:
chan_misdn_log(1, bc->port, " --> CCBSErase: InvokeID:%d, CCBSReference:%d RecallMode:%d, Reason:%d\n",
fac->u.CCBSErase.InvokeID, fac->u.CCBSErase.CCBSReference,
fac->u.CCBSErase.RecallMode, fac->u.CCBSErase.Reason);
chan_misdn_log(1, bc->port, " --> AddressOfB\n");
print_facility_Address(2, &fac->u.CCBSErase.AddressOfB, bc);
print_facility_Q931_Bc_Hlc_Llc(1, &fac->u.CCBSErase.Q931ie, bc);
break;
case Fac_CCBSRemoteUserFree:
chan_misdn_log(1, bc->port, " --> CCBSRemoteUserFree: InvokeID:%d, CCBSReference:%d RecallMode:%d\n",
fac->u.CCBSRemoteUserFree.InvokeID, fac->u.CCBSRemoteUserFree.CCBSReference,
fac->u.CCBSRemoteUserFree.RecallMode);
chan_misdn_log(1, bc->port, " --> AddressOfB\n");
print_facility_Address(2, &fac->u.CCBSRemoteUserFree.AddressOfB, bc);
print_facility_Q931_Bc_Hlc_Llc(1, &fac->u.CCBSRemoteUserFree.Q931ie, bc);
break;
case Fac_CCBSCall:
chan_misdn_log(1, bc->port, " --> CCBSCall: InvokeID:%d, CCBSReference:%d\n",
fac->u.CCBSCall.InvokeID, fac->u.CCBSCall.CCBSReference);
break;
case Fac_CCBSStatusRequest:
chan_misdn_log(1, bc->port, " --> CCBSStatusRequest: InvokeID:%d\n",
fac->u.CCBSStatusRequest.InvokeID);
switch (fac->u.CCBSStatusRequest.ComponentType) {
case FacComponent_Invoke:
chan_misdn_log(1, bc->port, " --> Invoke: CCBSReference:%d RecallMode:%d\n",
fac->u.CCBSStatusRequest.Component.Invoke.CCBSReference,
fac->u.CCBSStatusRequest.Component.Invoke.RecallMode);
print_facility_Q931_Bc_Hlc_Llc(2, &fac->u.CCBSStatusRequest.Component.Invoke.Q931ie, bc);
break;
case FacComponent_Result:
chan_misdn_log(1, bc->port, " --> Result: Free:%d\n",
fac->u.CCBSStatusRequest.Component.Result.Free);
break;
default:
break;
}
break;
case Fac_CCBSBFree:
chan_misdn_log(1, bc->port, " --> CCBSBFree: InvokeID:%d, CCBSReference:%d RecallMode:%d\n",
fac->u.CCBSBFree.InvokeID, fac->u.CCBSBFree.CCBSReference,
fac->u.CCBSBFree.RecallMode);
chan_misdn_log(1, bc->port, " --> AddressOfB\n");
print_facility_Address(2, &fac->u.CCBSBFree.AddressOfB, bc);
print_facility_Q931_Bc_Hlc_Llc(1, &fac->u.CCBSBFree.Q931ie, bc);
break;
case Fac_EraseCallLinkageID:
chan_misdn_log(1, bc->port, " --> EraseCallLinkageID: InvokeID:%d, LinkageID:%d\n",
fac->u.EraseCallLinkageID.InvokeID, fac->u.EraseCallLinkageID.CallLinkageID);
break;
case Fac_CCBSStopAlerting:
chan_misdn_log(1, bc->port, " --> CCBSStopAlerting: InvokeID:%d, CCBSReference:%d\n",
fac->u.CCBSStopAlerting.InvokeID, fac->u.CCBSStopAlerting.CCBSReference);
break;
case Fac_CCBSRequest:
chan_misdn_log(1, bc->port, " --> CCBSRequest: InvokeID:%d\n",
fac->u.CCBSRequest.InvokeID);
switch (fac->u.CCBSRequest.ComponentType) {
case FacComponent_Invoke:
chan_misdn_log(1, bc->port, " --> Invoke: LinkageID:%d\n",
fac->u.CCBSRequest.Component.Invoke.CallLinkageID);
break;
case FacComponent_Result:
chan_misdn_log(1, bc->port, " --> Result: CCBSReference:%d RecallMode:%d\n",
fac->u.CCBSRequest.Component.Result.CCBSReference,
fac->u.CCBSRequest.Component.Result.RecallMode);
break;
default:
break;
}
break;
case Fac_CCBSInterrogate:
chan_misdn_log(1, bc->port, " --> CCBSInterrogate: InvokeID:%d\n",
fac->u.CCBSInterrogate.InvokeID);
switch (fac->u.CCBSInterrogate.ComponentType) {
case FacComponent_Invoke:
chan_misdn_log(1, bc->port, " --> Invoke\n");
if (fac->u.CCBSInterrogate.Component.Invoke.CCBSReferencePresent) {
chan_misdn_log(1, bc->port, " --> CCBSReference:%d\n",
fac->u.CCBSInterrogate.Component.Invoke.CCBSReference);
}
if (fac->u.CCBSInterrogate.Component.Invoke.AParty.LengthOfNumber) {
chan_misdn_log(1, bc->port, " --> AParty\n");
print_facility_PartyNumber(3, &fac->u.CCBSInterrogate.Component.Invoke.AParty, bc);
}
break;
case FacComponent_Result:
chan_misdn_log(1, bc->port, " --> Result: RecallMode:%d\n",
fac->u.CCBSInterrogate.Component.Result.RecallMode);
if (fac->u.CCBSInterrogate.Component.Result.NumRecords) {
for (Index = 0; Index < fac->u.CCBSInterrogate.Component.Result.NumRecords; ++Index) {
chan_misdn_log(1, bc->port, " --> CallDetails[%d]:\n", Index);
print_facility_CallInformation(3, &fac->u.CCBSInterrogate.Component.Result.CallDetails[Index], bc);
}
}
break;
default:
break;
}
break;
case Fac_CCNRRequest:
chan_misdn_log(1, bc->port, " --> CCNRRequest: InvokeID:%d\n",
fac->u.CCNRRequest.InvokeID);
switch (fac->u.CCNRRequest.ComponentType) {
case FacComponent_Invoke:
chan_misdn_log(1, bc->port, " --> Invoke: LinkageID:%d\n",
fac->u.CCNRRequest.Component.Invoke.CallLinkageID);
break;
case FacComponent_Result:
chan_misdn_log(1, bc->port, " --> Result: CCBSReference:%d RecallMode:%d\n",
fac->u.CCNRRequest.Component.Result.CCBSReference,
fac->u.CCNRRequest.Component.Result.RecallMode);
break;
default:
break;
}
break;
case Fac_CCNRInterrogate:
chan_misdn_log(1, bc->port, " --> CCNRInterrogate: InvokeID:%d\n",
fac->u.CCNRInterrogate.InvokeID);
switch (fac->u.CCNRInterrogate.ComponentType) {
case FacComponent_Invoke:
chan_misdn_log(1, bc->port, " --> Invoke\n");
if (fac->u.CCNRInterrogate.Component.Invoke.CCBSReferencePresent) {
chan_misdn_log(1, bc->port, " --> CCBSReference:%d\n",
fac->u.CCNRInterrogate.Component.Invoke.CCBSReference);
}
if (fac->u.CCNRInterrogate.Component.Invoke.AParty.LengthOfNumber) {
chan_misdn_log(1, bc->port, " --> AParty\n");
print_facility_PartyNumber(3, &fac->u.CCNRInterrogate.Component.Invoke.AParty, bc);
}
break;
case FacComponent_Result:
chan_misdn_log(1, bc->port, " --> Result: RecallMode:%d\n",
fac->u.CCNRInterrogate.Component.Result.RecallMode);
if (fac->u.CCNRInterrogate.Component.Result.NumRecords) {
for (Index = 0; Index < fac->u.CCNRInterrogate.Component.Result.NumRecords; ++Index) {
chan_misdn_log(1, bc->port, " --> CallDetails[%d]:\n", Index);
print_facility_CallInformation(3, &fac->u.CCNRInterrogate.Component.Result.CallDetails[Index], bc);
}
}
break;
default:
break;
}
break;
case Fac_CCBS_T_Call:
chan_misdn_log(1, bc->port, " --> CCBS_T_Call: InvokeID:%d\n",
fac->u.CCBS_T_Call.InvokeID);
break;
case Fac_CCBS_T_Suspend:
chan_misdn_log(1, bc->port, " --> CCBS_T_Suspend: InvokeID:%d\n",
fac->u.CCBS_T_Suspend.InvokeID);
break;
case Fac_CCBS_T_Resume:
chan_misdn_log(1, bc->port, " --> CCBS_T_Resume: InvokeID:%d\n",
fac->u.CCBS_T_Resume.InvokeID);
break;
case Fac_CCBS_T_RemoteUserFree:
chan_misdn_log(1, bc->port, " --> CCBS_T_RemoteUserFree: InvokeID:%d\n",
fac->u.CCBS_T_RemoteUserFree.InvokeID);
break;
case Fac_CCBS_T_Available:
chan_misdn_log(1, bc->port, " --> CCBS_T_Available: InvokeID:%d\n",
fac->u.CCBS_T_Available.InvokeID);
break;
case Fac_CCBS_T_Request:
chan_misdn_log(1, bc->port, " --> CCBS_T_Request: InvokeID:%d\n",
fac->u.CCBS_T_Request.InvokeID);
switch (fac->u.CCBS_T_Request.ComponentType) {
case FacComponent_Invoke:
chan_misdn_log(1, bc->port, " --> Invoke\n");
chan_misdn_log(1, bc->port, " --> DestinationAddress:\n");
print_facility_Address(3, &fac->u.CCBS_T_Request.Component.Invoke.Destination, bc);
print_facility_Q931_Bc_Hlc_Llc(2, &fac->u.CCBS_T_Request.Component.Invoke.Q931ie, bc);
if (fac->u.CCBS_T_Request.Component.Invoke.RetentionSupported) {
chan_misdn_log(1, bc->port, " --> RetentionSupported:1\n");
}
if (fac->u.CCBS_T_Request.Component.Invoke.PresentationAllowedIndicatorPresent) {
chan_misdn_log(1, bc->port, " --> PresentationAllowed:%d\n",
fac->u.CCBS_T_Request.Component.Invoke.PresentationAllowedIndicator);
}
if (fac->u.CCBS_T_Request.Component.Invoke.Originating.Party.LengthOfNumber) {
chan_misdn_log(1, bc->port, " --> OriginatingAddress:\n");
print_facility_Address(3, &fac->u.CCBS_T_Request.Component.Invoke.Originating, bc);
}
break;
case FacComponent_Result:
chan_misdn_log(1, bc->port, " --> Result: RetentionSupported:%d\n",
fac->u.CCBS_T_Request.Component.Result.RetentionSupported);
break;
default:
break;
}
break;
case Fac_CCNR_T_Request:
chan_misdn_log(1, bc->port, " --> CCNR_T_Request: InvokeID:%d\n",
fac->u.CCNR_T_Request.InvokeID);
switch (fac->u.CCNR_T_Request.ComponentType) {
case FacComponent_Invoke:
chan_misdn_log(1, bc->port, " --> Invoke\n");
chan_misdn_log(1, bc->port, " --> DestinationAddress:\n");
print_facility_Address(3, &fac->u.CCNR_T_Request.Component.Invoke.Destination, bc);
print_facility_Q931_Bc_Hlc_Llc(2, &fac->u.CCNR_T_Request.Component.Invoke.Q931ie, bc);
if (fac->u.CCNR_T_Request.Component.Invoke.RetentionSupported) {
chan_misdn_log(1, bc->port, " --> RetentionSupported:1\n");
}
if (fac->u.CCNR_T_Request.Component.Invoke.PresentationAllowedIndicatorPresent) {
chan_misdn_log(1, bc->port, " --> PresentationAllowed:%d\n",
fac->u.CCNR_T_Request.Component.Invoke.PresentationAllowedIndicator);
}
if (fac->u.CCNR_T_Request.Component.Invoke.Originating.Party.LengthOfNumber) {
chan_misdn_log(1, bc->port, " --> OriginatingAddress:\n");
print_facility_Address(3, &fac->u.CCNR_T_Request.Component.Invoke.Originating, bc);
}
break;
case FacComponent_Result:
chan_misdn_log(1, bc->port, " --> Result: RetentionSupported:%d\n",
fac->u.CCNR_T_Request.Component.Result.RetentionSupported);
break;
default:
break;
}
break;
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
case Fac_None:
/* No facility so print nothing */
break;
default:
chan_misdn_log(1, bc->port, " --> unknown facility\n");
break;
}
}
static void print_bearer(struct misdn_bchannel *bc)
{
chan_misdn_log(2, bc->port, " --> Bearer: %s\n", bearer2str(bc->capability));
switch(bc->law) {
case INFO_CODEC_ALAW:
chan_misdn_log(2, bc->port, " --> Codec: Alaw\n");
break;
case INFO_CODEC_ULAW:
chan_misdn_log(2, bc->port, " --> Codec: Ulaw\n");
break;
}
}
/*!
* \internal
* \brief Prefix a string to another string in place.
*
* \param str_prefix String to prefix to the main string.
* \param str_main String to get the prefix added to it.
* \param size Buffer size of the main string (Includes null terminator).
*
* \note The str_main buffer size must be greater than one.
*
* \return Nothing
*/
static void misdn_prefix_string(const char *str_prefix, char *str_main, size_t size)
{
size_t len_over;
size_t len_total;
size_t len_main;
size_t len_prefix;
len_prefix = strlen(str_prefix);
if (!len_prefix) {
/* There is no prefix to prepend. */
return;
}
len_main = strlen(str_main);
len_total = len_prefix + len_main;
if (size <= len_total) {
/* We need to truncate since the buffer is too small. */
len_over = len_total + 1 - size;
if (len_over <= len_main) {
len_main -= len_over;
} else {
len_over -= len_main;
len_main = 0;
len_prefix -= len_over;
}
}
if (len_main) {
memmove(str_main + len_prefix, str_main, len_main);
}
memcpy(str_main, str_prefix, len_prefix);
str_main[len_prefix + len_main] = '\0';
}
/*!
* \internal
* \brief Add a configured prefix to the given number.
*
* \param port Logical port number
* \param number_type Type-of-number passed in.
* \param number Given number string to add prefix
* \param size Buffer size number string occupies.
*
* \return Nothing
*/
static void misdn_add_number_prefix(int port, enum mISDN_NUMBER_TYPE number_type, char *number, size_t size)
{
enum misdn_cfg_elements type_prefix;
char num_prefix[MISDN_MAX_NUMBER_LEN];
/* Get prefix string. */
switch (number_type) {
case NUMTYPE_UNKNOWN:
type_prefix = MISDN_CFG_TON_PREFIX_UNKNOWN;
break;
case NUMTYPE_INTERNATIONAL:
type_prefix = MISDN_CFG_TON_PREFIX_INTERNATIONAL;
break;
case NUMTYPE_NATIONAL:
type_prefix = MISDN_CFG_TON_PREFIX_NATIONAL;
break;
case NUMTYPE_NETWORK_SPECIFIC:
type_prefix = MISDN_CFG_TON_PREFIX_NETWORK_SPECIFIC;
break;
case NUMTYPE_SUBSCRIBER:
type_prefix = MISDN_CFG_TON_PREFIX_SUBSCRIBER;
break;
case NUMTYPE_ABBREVIATED:
type_prefix = MISDN_CFG_TON_PREFIX_ABBREVIATED;
break;
default:
/* Type-of-number does not have a prefix that can be added. */
return;
}
misdn_cfg_get(port, type_prefix, num_prefix, sizeof(num_prefix));
misdn_prefix_string(num_prefix, number, size);
}
static void export_aoc_vars(int originator, struct ast_channel *ast, struct misdn_bchannel *bc)
{
RAII_VAR(struct ast_channel *, chan, NULL, ast_channel_cleanup);
char buf[128];
if (!bc->AOCD_need_export || !ast) {
return;
}
if (originator == ORG_AST) {
chan = ast_channel_bridge_peer(ast);
if (!chan) {
return;
}
} else {
chan = ast_channel_ref(ast);
}
switch (bc->AOCDtype) {
case Fac_AOCDCurrency:
pbx_builtin_setvar_helper(chan, "AOCD_Type", "currency");
if (bc->AOCD.currency.chargeNotAvailable) {
pbx_builtin_setvar_helper(chan, "AOCD_ChargeAvailable", "no");
} else {
pbx_builtin_setvar_helper(chan, "AOCD_ChargeAvailable", "yes");
if (bc->AOCD.currency.freeOfCharge) {
pbx_builtin_setvar_helper(chan, "AOCD_FreeOfCharge", "yes");
} else {
pbx_builtin_setvar_helper(chan, "AOCD_FreeOfCharge", "no");
if (snprintf(buf, sizeof(buf), "%d %s", bc->AOCD.currency.currencyAmount * bc->AOCD.currency.multiplier, bc->AOCD.currency.currency) < sizeof(buf)) {
pbx_builtin_setvar_helper(chan, "AOCD_Amount", buf);
if (bc->AOCD.currency.billingId >= 0 && snprintf(buf, sizeof(buf), "%d", bc->AOCD.currency.billingId) < sizeof(buf)) {
pbx_builtin_setvar_helper(chan, "AOCD_BillingId", buf);
}
}
}
}
break;
case Fac_AOCDChargingUnit:
pbx_builtin_setvar_helper(chan, "AOCD_Type", "charging_unit");
if (bc->AOCD.chargingUnit.chargeNotAvailable) {
pbx_builtin_setvar_helper(chan, "AOCD_ChargeAvailable", "no");
} else {
pbx_builtin_setvar_helper(chan, "AOCD_ChargeAvailable", "yes");
if (bc->AOCD.chargingUnit.freeOfCharge) {
pbx_builtin_setvar_helper(chan, "AOCD_FreeOfCharge", "yes");
} else {
pbx_builtin_setvar_helper(chan, "AOCD_FreeOfCharge", "no");
if (snprintf(buf, sizeof(buf), "%d", bc->AOCD.chargingUnit.recordedUnits) < sizeof(buf)) {
pbx_builtin_setvar_helper(chan, "AOCD_RecordedUnits", buf);
if (bc->AOCD.chargingUnit.billingId >= 0 && snprintf(buf, sizeof(buf), "%d", bc->AOCD.chargingUnit.billingId) < sizeof(buf)) {
pbx_builtin_setvar_helper(chan, "AOCD_BillingId", buf);
}
}
}
}
break;
default:
break;
}
bc->AOCD_need_export = 0;
}
/*************** Helpers END *************/
static void sighandler(int sig)
{
}
static void *misdn_tasks_thread_func(void *data)
{
int wait;
struct sigaction sa;
sa.sa_handler = sighandler;
sa.sa_flags = SA_NODEFER;
sigemptyset(&sa.sa_mask);
sigaddset(&sa.sa_mask, SIGUSR1);
sigaction(SIGUSR1, &sa, NULL);
sem_post((sem_t *)data);
while (1) {
wait = ast_sched_wait(misdn_tasks);
if (wait < 0) {
wait = 8000;
}
if (poll(NULL, 0, wait) < 0) {
chan_misdn_log(4, 0, "Waking up misdn_tasks thread\n");
}
ast_sched_runq(misdn_tasks);
}
return NULL;
}
static void misdn_tasks_init(void)
{
sem_t blocker;
int i = 5;
if (sem_init(&blocker, 0, 0)) {
perror("chan_misdn: Failed to initialize semaphore!");
exit(1);
}
chan_misdn_log(4, 0, "Starting misdn_tasks thread\n");
misdn_tasks = ast_sched_context_create();
pthread_create(&misdn_tasks_thread, NULL, misdn_tasks_thread_func, &blocker);
while (sem_wait(&blocker) && --i) {
}
sem_destroy(&blocker);
}
static void misdn_tasks_destroy(void)
{
if (misdn_tasks) {
chan_misdn_log(4, 0, "Killing misdn_tasks thread\n");
if (pthread_cancel(misdn_tasks_thread) == 0) {
cb_log(4, 0, "Joining misdn_tasks thread\n");
pthread_join(misdn_tasks_thread, NULL);
}
ast_sched_context_destroy(misdn_tasks);
}
}
static inline void misdn_tasks_wakeup(void)
{
pthread_kill(misdn_tasks_thread, SIGUSR1);
}
static inline int _misdn_tasks_add_variable(int timeout, ast_sched_cb callback, const void *data, int variable)
{
int task_id;
if (!misdn_tasks) {
misdn_tasks_init();
}
task_id = ast_sched_add_variable(misdn_tasks, timeout, callback, data, variable);
misdn_tasks_wakeup();
return task_id;
}
static int misdn_tasks_add(int timeout, ast_sched_cb callback, const void *data)
{
return _misdn_tasks_add_variable(timeout, callback, data, 0);
}
static int misdn_tasks_add_variable(int timeout, ast_sched_cb callback, const void *data)
{
return _misdn_tasks_add_variable(timeout, callback, data, 1);
}
static void misdn_tasks_remove(int task_id)
{
AST_SCHED_DEL(misdn_tasks, task_id);
}
static int misdn_l1_task(const void *vdata)
{
const int *data = vdata;
misdn_lib_isdn_l1watcher(*data);
chan_misdn_log(5, *data, "L1watcher timeout\n");
return 1;
}
static int misdn_overlap_dial_task(const void *data)
{
struct timeval tv_end, tv_now;
int diff;
struct chan_list *ch = (struct chan_list *) data;
char *dad;
chan_misdn_log(4, ch->bc->port, "overlap dial task, chan_state: %d\n", ch->state);
if (ch->state != MISDN_WAITING4DIGS) {
ch->overlap_dial_task = -1;
return 0;
}
ast_mutex_lock(&ch->overlap_tv_lock);
tv_end = ch->overlap_tv;
ast_mutex_unlock(&ch->overlap_tv_lock);
tv_end.tv_sec += ch->overlap_dial;
tv_now = ast_tvnow();
diff = ast_tvdiff_ms(tv_end, tv_now);
if (100 < diff) {
return diff;
}
/* if we are 100ms near the timeout, we are satisfied.. */
stop_indicate(ch);
if (ast_strlen_zero(ch->bc->dialed.number)) {
dad = "s";
ast_channel_exten_set(ch->ast, dad);
} else {
dad = ch->bc->dialed.number;
}
if (ast_exists_extension(ch->ast, ch->context, dad, 1, ch->bc->caller.number)) {
ch->state = MISDN_DIALING;
if (pbx_start_chan(ch) < 0) {
chan_misdn_log(-1, ch->bc->port, "ast_pbx_start returned < 0 in misdn_overlap_dial_task\n");
goto misdn_overlap_dial_task_disconnect;
}
} else {
misdn_overlap_dial_task_disconnect:
hanguptone_indicate(ch);
ch->bc->out_cause = AST_CAUSE_UNALLOCATED;
ch->state = MISDN_CLEANING;
misdn_lib_send_event(ch->bc, EVENT_DISCONNECT);
}
ch->overlap_dial_task = -1;
return 0;
}
static void send_digit_to_chan(struct chan_list *cl, char digit)
{
static const char * const dtmf_tones[] = {
/* *INDENT-OFF* */
"!941+1336/100,!0/100", /* 0 */
"!697+1209/100,!0/100", /* 1 */
"!697+1336/100,!0/100", /* 2 */
"!697+1477/100,!0/100", /* 3 */
"!770+1209/100,!0/100", /* 4 */
"!770+1336/100,!0/100", /* 5 */
"!770+1477/100,!0/100", /* 6 */
"!852+1209/100,!0/100", /* 7 */
"!852+1336/100,!0/100", /* 8 */
"!852+1477/100,!0/100", /* 9 */
"!697+1633/100,!0/100", /* A */
"!770+1633/100,!0/100", /* B */
"!852+1633/100,!0/100", /* C */
"!941+1633/100,!0/100", /* D */
"!941+1209/100,!0/100", /* * */
"!941+1477/100,!0/100", /* # */
/* *INDENT-ON* */
};
struct ast_channel *chan = cl->ast;
if (digit >= '0' && digit <='9') {
ast_playtones_start(chan, 0, dtmf_tones[digit - '0'], 0);
} else if (digit >= 'A' && digit <= 'D') {
ast_playtones_start(chan, 0, dtmf_tones[digit - 'A' + 10], 0);
} else if (digit == '*') {
ast_playtones_start(chan, 0, dtmf_tones[14], 0);
} else if (digit == '#') {
ast_playtones_start(chan, 0, dtmf_tones[15], 0);
} else {
/* not handled */
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
ast_debug(1, "Unable to handle DTMF tone '%c' for '%s'\n", digit, ast_channel_name(chan));
}
}
/*** CLI HANDLING ***/
static char *handle_cli_misdn_set_debug(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
int level;
switch (cmd) {
case CLI_INIT:
e->command = "misdn set debug [on|off]";
e->usage =
"Usage: misdn set debug {on|off|<level>} [only] | [port <port> [only]]\n"
" Set the debug level of the mISDN channel.\n";
return NULL;
case CLI_GENERATE:
return complete_debug_port(a);
}
if (a->argc < 4 || a->argc > 7) {
return CLI_SHOWUSAGE;
}
if (!strcasecmp(a->argv[3], "on")) {
level = 1;
} else if (!strcasecmp(a->argv[3], "off")) {
level = 0;
} else if (isdigit(a->argv[3][0])) {
level = atoi(a->argv[3]);
} else {
return CLI_SHOWUSAGE;
}
switch (a->argc) {
case 4:
case 5:
{
int i;
int only = 0;
if (a->argc == 5) {
if (strncasecmp(a->argv[4], "only", strlen(a->argv[4]))) {
return CLI_SHOWUSAGE;
} else {
only = 1;
}
}
for (i = 0; i <= max_ports; i++) {
misdn_debug[i] = level;
misdn_debug_only[i] = only;
}
ast_cli(a->fd, "changing debug level for all ports to %d%s\n", misdn_debug[0], only ? " (only)" : "");
}
break;
case 6:
case 7:
{
int port;
if (strncasecmp(a->argv[4], "port", strlen(a->argv[4])))
return CLI_SHOWUSAGE;
port = atoi(a->argv[5]);
if (port <= 0 || port > max_ports) {
switch (max_ports) {
case 0:
ast_cli(a->fd, "port number not valid! no ports available so you won't get lucky with any number here...\n");
break;
case 1:
ast_cli(a->fd, "port number not valid! only port 1 is available.\n");
break;
default:
ast_cli(a->fd, "port number not valid! only ports 1 to %d are available.\n", max_ports);
}
return 0;
}
if (a->argc == 7) {
if (strncasecmp(a->argv[6], "only", strlen(a->argv[6]))) {
return CLI_SHOWUSAGE;
} else {
misdn_debug_only[port] = 1;
}
} else {
misdn_debug_only[port] = 0;
}
misdn_debug[port] = level;
ast_cli(a->fd, "changing debug level to %d%s for port %d\n", misdn_debug[port], misdn_debug_only[port] ? " (only)" : "", port);
}
}
return CLI_SUCCESS;
}
static char *handle_cli_misdn_set_crypt_debug(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
switch (cmd) {
case CLI_INIT:
e->command = "misdn set crypt debug";
e->usage =
"Usage: misdn set crypt debug <level>\n"
" Set the crypt debug level of the mISDN channel. Level\n"
" must be 1 or 2.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != 5) {
return CLI_SHOWUSAGE;
}
/* XXX Is this supposed to not do anything? XXX */
return CLI_SUCCESS;
}
static char *handle_cli_misdn_port_block(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
switch (cmd) {
case CLI_INIT:
e->command = "misdn port block";
e->usage =
"Usage: misdn port block <port>\n"
" Block the specified port by <port>.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != 4) {
return CLI_SHOWUSAGE;
}
misdn_lib_port_block(atoi(a->argv[3]));
return CLI_SUCCESS;
}
static char *handle_cli_misdn_port_unblock(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
switch (cmd) {
case CLI_INIT:
e->command = "misdn port unblock";
e->usage =
"Usage: misdn port unblock <port>\n"
" Unblock the port specified by <port>.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != 4) {
return CLI_SHOWUSAGE;
}
misdn_lib_port_unblock(atoi(a->argv[3]));
return CLI_SUCCESS;
}
static char *handle_cli_misdn_restart_port(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
switch (cmd) {
case CLI_INIT:
e->command = "misdn restart port";
e->usage =
"Usage: misdn restart port <port>\n"
" Restart the given port.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != 4) {
return CLI_SHOWUSAGE;
}
misdn_lib_port_restart(atoi(a->argv[3]));
return CLI_SUCCESS;
}
static char *handle_cli_misdn_restart_pid(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
Merged revisions 49313 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r49313 | crichter | 2007-01-03 10:06:50 +0100 (Mi, 03 Jan 2007) | 41 lines Merged revisions 48319,48321,48467,48552,48576,49135,49303 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) | 1 line changed a few debugs to higher debug levels ........ r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) | 1 line added the export and import of the MISDN_ADDRESS_COMPLETE Variable to inidcate wether the extension is already completely dialed or if there might come additional digits by information elements. also added some docs for that. ........ r48467 | crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line removed FIXUP state. added check for channel allocation conflict when we create a setup while the other site creates a setup on the same channel, besides the check we resolve this conflict. ........ r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18 Dez 2006) | 1 line when our PTP Partner sends us a SETUP with a preselected channel we just accept it, even when we're NT. added some checks for segfaults. ........ r48576 | crichter | 2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line when we reject a channel, because it's in use already, we shouldn't process the setup anymore. made the channel allocation a bit easier and more understandable, removed a few unused lines ........ r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02 Jan 2007) | 1 line added check for channel ranges in the set/empty channel functions. set pmp_l1_check default to no. added misdn restart pid cli command. added cleaning of channel when we send a RELEASE_COMPLETE. ........ r49303 | crichter | 2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines * Added check for bridging in misdn_call to avoid setting echocancellation when 2 mISDN channels are involved and when bridging is set. That lead to a kernel panic before under different situations, because we switched about 2 times between hardware bridging and echocancelation * readded MISDN_URATE variable which got lost before, this should make app_v110 work again * fixed typo ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49321 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-03 11:15:02 +00:00
{
switch (cmd) {
case CLI_INIT:
e->command = "misdn restart pid";
e->usage =
"Usage: misdn restart pid <pid>\n"
" Restart the given pid\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != 4) {
return CLI_SHOWUSAGE;
}
Merged revisions 49313 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r49313 | crichter | 2007-01-03 10:06:50 +0100 (Mi, 03 Jan 2007) | 41 lines Merged revisions 48319,48321,48467,48552,48576,49135,49303 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) | 1 line changed a few debugs to higher debug levels ........ r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) | 1 line added the export and import of the MISDN_ADDRESS_COMPLETE Variable to inidcate wether the extension is already completely dialed or if there might come additional digits by information elements. also added some docs for that. ........ r48467 | crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line removed FIXUP state. added check for channel allocation conflict when we create a setup while the other site creates a setup on the same channel, besides the check we resolve this conflict. ........ r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18 Dez 2006) | 1 line when our PTP Partner sends us a SETUP with a preselected channel we just accept it, even when we're NT. added some checks for segfaults. ........ r48576 | crichter | 2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line when we reject a channel, because it's in use already, we shouldn't process the setup anymore. made the channel allocation a bit easier and more understandable, removed a few unused lines ........ r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02 Jan 2007) | 1 line added check for channel ranges in the set/empty channel functions. set pmp_l1_check default to no. added misdn restart pid cli command. added cleaning of channel when we send a RELEASE_COMPLETE. ........ r49303 | crichter | 2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines * Added check for bridging in misdn_call to avoid setting echocancellation when 2 mISDN channels are involved and when bridging is set. That lead to a kernel panic before under different situations, because we switched about 2 times between hardware bridging and echocancelation * readded MISDN_URATE variable which got lost before, this should make app_v110 work again * fixed typo ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49321 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-03 11:15:02 +00:00
misdn_lib_pid_restart(atoi(a->argv[3]));
Merged revisions 49313 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r49313 | crichter | 2007-01-03 10:06:50 +0100 (Mi, 03 Jan 2007) | 41 lines Merged revisions 48319,48321,48467,48552,48576,49135,49303 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) | 1 line changed a few debugs to higher debug levels ........ r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) | 1 line added the export and import of the MISDN_ADDRESS_COMPLETE Variable to inidcate wether the extension is already completely dialed or if there might come additional digits by information elements. also added some docs for that. ........ r48467 | crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line removed FIXUP state. added check for channel allocation conflict when we create a setup while the other site creates a setup on the same channel, besides the check we resolve this conflict. ........ r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18 Dez 2006) | 1 line when our PTP Partner sends us a SETUP with a preselected channel we just accept it, even when we're NT. added some checks for segfaults. ........ r48576 | crichter | 2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line when we reject a channel, because it's in use already, we shouldn't process the setup anymore. made the channel allocation a bit easier and more understandable, removed a few unused lines ........ r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02 Jan 2007) | 1 line added check for channel ranges in the set/empty channel functions. set pmp_l1_check default to no. added misdn restart pid cli command. added cleaning of channel when we send a RELEASE_COMPLETE. ........ r49303 | crichter | 2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines * Added check for bridging in misdn_call to avoid setting echocancellation when 2 mISDN channels are involved and when bridging is set. That lead to a kernel panic before under different situations, because we switched about 2 times between hardware bridging and echocancelation * readded MISDN_URATE variable which got lost before, this should make app_v110 work again * fixed typo ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49321 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-03 11:15:02 +00:00
return CLI_SUCCESS;
Merged revisions 49313 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r49313 | crichter | 2007-01-03 10:06:50 +0100 (Mi, 03 Jan 2007) | 41 lines Merged revisions 48319,48321,48467,48552,48576,49135,49303 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) | 1 line changed a few debugs to higher debug levels ........ r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) | 1 line added the export and import of the MISDN_ADDRESS_COMPLETE Variable to inidcate wether the extension is already completely dialed or if there might come additional digits by information elements. also added some docs for that. ........ r48467 | crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line removed FIXUP state. added check for channel allocation conflict when we create a setup while the other site creates a setup on the same channel, besides the check we resolve this conflict. ........ r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18 Dez 2006) | 1 line when our PTP Partner sends us a SETUP with a preselected channel we just accept it, even when we're NT. added some checks for segfaults. ........ r48576 | crichter | 2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line when we reject a channel, because it's in use already, we shouldn't process the setup anymore. made the channel allocation a bit easier and more understandable, removed a few unused lines ........ r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02 Jan 2007) | 1 line added check for channel ranges in the set/empty channel functions. set pmp_l1_check default to no. added misdn restart pid cli command. added cleaning of channel when we send a RELEASE_COMPLETE. ........ r49303 | crichter | 2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines * Added check for bridging in misdn_call to avoid setting echocancellation when 2 mISDN channels are involved and when bridging is set. That lead to a kernel panic before under different situations, because we switched about 2 times between hardware bridging and echocancelation * readded MISDN_URATE variable which got lost before, this should make app_v110 work again * fixed typo ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49321 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-03 11:15:02 +00:00
}
static char *handle_cli_misdn_port_up(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
switch (cmd) {
case CLI_INIT:
e->command = "misdn port up";
e->usage =
"Usage: misdn port up <port>\n"
" Try to establish L1 on the given port.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != 4) {
return CLI_SHOWUSAGE;
}
misdn_lib_get_port_up(atoi(a->argv[3]));
return CLI_SUCCESS;
}
static char *handle_cli_misdn_port_down(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
switch (cmd) {
case CLI_INIT:
e->command = "misdn port down";
e->usage =
"Usage: misdn port down <port>\n"
" Try to deactivate the L1 on the given port.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != 4) {
return CLI_SHOWUSAGE;
}
misdn_lib_get_port_down(atoi(a->argv[3]));
return CLI_SUCCESS;
}
static inline void show_config_description(int fd, enum misdn_cfg_elements elem)
{
char section[BUFFERSIZE];
char name[BUFFERSIZE];
char desc[BUFFERSIZE];
char def[BUFFERSIZE];
char tmp[BUFFERSIZE];
misdn_cfg_get_name(elem, tmp, sizeof(tmp));
term_color(name, tmp, COLOR_BRWHITE, 0, sizeof(tmp));
misdn_cfg_get_desc(elem, desc, sizeof(desc), def, sizeof(def));
if (elem < MISDN_CFG_LAST) {
term_color(section, "PORTS SECTION", COLOR_YELLOW, 0, sizeof(section));
} else {
term_color(section, "GENERAL SECTION", COLOR_YELLOW, 0, sizeof(section));
}
if (*def) {
ast_cli(fd, "[%s] %s (Default: %s)\n\t%s\n", section, name, def, desc);
} else {
ast_cli(fd, "[%s] %s\n\t%s\n", section, name, desc);
}
}
static char *handle_cli_misdn_show_config(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
char buffer[BUFFERSIZE];
enum misdn_cfg_elements elem;
int linebreak;
int onlyport = -1;
int ok = 0;
switch (cmd) {
case CLI_INIT:
e->command = "misdn show config";
e->usage =
"Usage: misdn show config [<port> | description <config element> | descriptions [general|ports]]\n"
" Use 0 for <port> to only print the general config.\n";
return NULL;
case CLI_GENERATE:
return complete_show_config(a);
}
if (a->argc >= 4) {
if (!strcmp(a->argv[3], "description")) {
if (a->argc == 5) {
enum misdn_cfg_elements elem = misdn_cfg_get_elem(a->argv[4]);
if (elem == MISDN_CFG_FIRST) {
ast_cli(a->fd, "Unknown element: %s\n", a->argv[4]);
} else {
show_config_description(a->fd, elem);
}
return CLI_SUCCESS;
}
return CLI_SHOWUSAGE;
} else if (!strcmp(a->argv[3], "descriptions")) {
if ((a->argc == 4) || ((a->argc == 5) && !strcmp(a->argv[4], "general"))) {
for (elem = MISDN_GEN_FIRST + 1; elem < MISDN_GEN_LAST; ++elem) {
show_config_description(a->fd, elem);
ast_cli(a->fd, "\n");
}
ok = 1;
}
if ((a->argc == 4) || ((a->argc == 5) && !strcmp(a->argv[4], "ports"))) {
for (elem = MISDN_CFG_FIRST + 1; elem < MISDN_CFG_LAST - 1 /* the ptp hack, remove the -1 when ptp is gone */; ++elem) {
show_config_description(a->fd, elem);
ast_cli(a->fd, "\n");
}
ok = 1;
}
return ok ? CLI_SUCCESS : CLI_SHOWUSAGE;
} else if (!sscanf(a->argv[3], "%5d", &onlyport) || onlyport < 0) {
ast_cli(a->fd, "Unknown option: %s\n", a->argv[3]);
return CLI_SHOWUSAGE;
}
}
if (a->argc == 3 || onlyport == 0) {
ast_cli(a->fd, "mISDN General-Config:\n");
for (elem = MISDN_GEN_FIRST + 1, linebreak = 1; elem < MISDN_GEN_LAST; elem++, linebreak++) {
misdn_cfg_get_config_string(0, elem, buffer, sizeof(buffer));
ast_cli(a->fd, "%-36s%s", buffer, !(linebreak % 2) ? "\n" : "");
}
ast_cli(a->fd, "\n");
}
if (onlyport < 0) {
int port = misdn_cfg_get_next_port(0);
for (; port > 0; port = misdn_cfg_get_next_port(port)) {
ast_cli(a->fd, "\n[PORT %d]\n", port);
for (elem = MISDN_CFG_FIRST + 1, linebreak = 1; elem < MISDN_CFG_LAST; elem++, linebreak++) {
misdn_cfg_get_config_string(port, elem, buffer, sizeof(buffer));
ast_cli(a->fd, "%-36s%s", buffer, !(linebreak % 2) ? "\n" : "");
}
ast_cli(a->fd, "\n");
}
}
if (onlyport > 0) {
if (misdn_cfg_is_port_valid(onlyport)) {
ast_cli(a->fd, "[PORT %d]\n", onlyport);
for (elem = MISDN_CFG_FIRST + 1, linebreak = 1; elem < MISDN_CFG_LAST; elem++, linebreak++) {
misdn_cfg_get_config_string(onlyport, elem, buffer, sizeof(buffer));
ast_cli(a->fd, "%-36s%s", buffer, !(linebreak % 2) ? "\n" : "");
}
ast_cli(a->fd, "\n");
} else {
ast_cli(a->fd, "Port %d is not active!\n", onlyport);
}
}
return CLI_SUCCESS;
}
struct state_struct {
enum misdn_chan_state state;
char txt[255];
};
static const struct state_struct state_array[] = {
/* *INDENT-OFF* */
{ MISDN_NOTHING, "NOTHING" }, /* at beginning */
{ MISDN_WAITING4DIGS, "WAITING4DIGS" }, /* when waiting for infos */
{ MISDN_EXTCANTMATCH, "EXTCANTMATCH" }, /* when asterisk couldn't match our ext */
{ MISDN_INCOMING_SETUP, "INCOMING SETUP" }, /* when pbx_start */
{ MISDN_DIALING, "DIALING" }, /* when pbx_start */
{ MISDN_PROGRESS, "PROGRESS" }, /* when pbx_start */
{ MISDN_PROCEEDING, "PROCEEDING" }, /* when pbx_start */
{ MISDN_CALLING, "CALLING" }, /* when misdn_call is called */
{ MISDN_CALLING_ACKNOWLEDGE, "CALLING_ACKNOWLEDGE" }, /* when misdn_call is called */
{ MISDN_ALERTING, "ALERTING" }, /* when Alerting */
{ MISDN_BUSY, "BUSY" }, /* when BUSY */
{ MISDN_CONNECTED, "CONNECTED" }, /* when connected */
{ MISDN_DISCONNECTED, "DISCONNECTED" }, /* when connected */
{ MISDN_CLEANING, "CLEANING" }, /* when hangup from * but we were connected before */
/* *INDENT-ON* */
};
static const char *misdn_get_ch_state(struct chan_list *p)
{
int i;
static char state[8];
if (!p) {
return NULL;
}
for (i = 0; i < ARRAY_LEN(state_array); i++) {
if (state_array[i].state == p->state) {
return state_array[i].txt;
}
}
snprintf(state, sizeof(state), "%d", p->state) ;
return state;
}
static void reload_config(void)
{
int i, cfg_debug;
if (!g_config_initialized) {
ast_log(LOG_WARNING, "chan_misdn is not initialized properly, still reloading ?\n");
return ;
}
free_robin_list();
misdn_cfg_reload();
misdn_cfg_update_ptp();
misdn_cfg_get(0, MISDN_GEN_TRACEFILE, global_tracefile, sizeof(global_tracefile));
misdn_cfg_get(0, MISDN_GEN_DEBUG, &cfg_debug, sizeof(cfg_debug));
for (i = 0; i <= max_ports; i++) {
misdn_debug[i] = cfg_debug;
misdn_debug_only[i] = 0;
}
}
static char *handle_cli_misdn_reload(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
switch (cmd) {
case CLI_INIT:
e->command = "misdn reload";
e->usage =
"Usage: misdn reload\n"
" Reload internal mISDN config, read from the config\n"
" file.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != 2) {
return CLI_SHOWUSAGE;
}
ast_cli(a->fd, "Reloading mISDN configuration\n");
reload_config();
return CLI_SUCCESS;
}
static void print_bc_info(int fd, struct chan_list *help, struct misdn_bchannel *bc)
{
struct ast_channel *ast = help->ast;
ast_cli(fd,
"* Pid:%d Port:%d Ch:%d Mode:%s Orig:%s dialed:%s\n"
" --> caller:\"%s\" <%s>\n"
" --> redirecting-from:\"%s\" <%s>\n"
" --> redirecting-to:\"%s\" <%s>\n"
" --> context:%s state:%s\n",
bc->pid,
bc->port,
bc->channel,
bc->nt ? "NT" : "TE",
help->originator == ORG_AST ? "*" : "I",
ast ? ast_channel_exten(ast) : "",
(ast && ast_channel_caller(ast)->id.name.valid && ast_channel_caller(ast)->id.name.str)
? ast_channel_caller(ast)->id.name.str : "",
(ast && ast_channel_caller(ast)->id.number.valid && ast_channel_caller(ast)->id.number.str)
? ast_channel_caller(ast)->id.number.str : "",
bc->redirecting.from.name,
bc->redirecting.from.number,
bc->redirecting.to.name,
bc->redirecting.to.number,
ast ? ast_channel_context(ast) : "",
misdn_get_ch_state(help));
if (misdn_debug[bc->port] > 0) {
ast_cli(fd,
" --> astname: %s\n"
" --> ch_l3id: %x\n"
" --> ch_addr: %x\n"
" --> bc_addr: %x\n"
" --> bc_l3id: %x\n"
" --> display: %s\n"
" --> activated: %d\n"
" --> state: %s\n"
" --> capability: %s\n"
#ifdef MISDN_1_2
" --> pipeline: %s\n"
#else
" --> echo_cancel: %d\n"
#endif
" --> notone : rx %d tx:%d\n"
" --> bc_hold: %d\n",
Merged revisions 374515-374535 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ................ r374515 | rmudgett | 2012-10-04 17:52:36 -0500 (Thu, 04 Oct 2012) | 10 lines chan_misdn: Remove some deadcode * Made setup_bc() static. Patches: patch1_unused-code.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882 ................ r374516 | rmudgett | 2012-10-04 18:01:01 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused bchan states Patches: patch2_unused-states.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374517 | rmudgett | 2012-10-04 18:17:51 -0500 (Thu, 04 Oct 2012) | 16 lines chan_misdn: Remove unnecessary null pointer checks and checks for stack->nt * cleanup_bc() is always called with valid bc (or it would've crashed before). * Value of stack->nt is known in advance at some places. * Rename handle_event() to handle_event_te(), handle_frm() to handle_frm_te(). Patches: patch3_checks.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882 ................ r374518 | rmudgett | 2012-10-04 18:21:59 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Fix spelling in log messages Patches: patch4_spelling.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374519 | rmudgett | 2012-10-04 18:31:59 -0500 (Thu, 04 Oct 2012) | 15 lines chan_misdn: Don't cleanup a bc twice. In handle_frm_te() after calling misdn_lib_send_event(bc, EVENT_RELEASE_COMPLETE) bc is emptied, cleaned and set not in use, although misdn_lib_send_event() already did the same. This is bad. When it's not in use we are not allowed to touch it. * Moved log message in front of the resulting actions and fixed it to match the case. Patches: patch5_bccleanup.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374520 | rmudgett | 2012-10-04 18:43:56 -0500 (Thu, 04 Oct 2012) | 12 lines chan_misdn: Fix memory leaks, bc, chan not cleaned up etc., really bad stuff. * Fix return codes of cb_events() for EVENT_SETUP to use caller's cleanup mechanisms. * Move cl_queue_chan() call after bearer check. Patches: patch6_leaks.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374521 | rmudgett | 2012-10-04 18:48:38 -0500 (Thu, 04 Oct 2012) | 11 lines chan_misdn: We must initialize cause on sending a DISCONNECT. We must initialize cause on sending a DISCONNECT, so it is later correctly indicated to ast_channel in case the answer (RELEASE/RELEASE_COMPLETE) does not include one. Patches: patch7_hangupcause.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374522 | rmudgett | 2012-10-04 19:03:56 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused code for upqueue Patches: patch8_unused-upqueue.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374523 | rmudgett | 2012-10-04 19:11:50 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Improve debugging (port number, messages fixed, dups removed) Patches: patch9_debug.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374533 | rmudgett | 2012-10-05 12:17:18 -0500 (Fri, 05 Oct 2012) | 8 lines chan_misdn: Better debug: we can print_bc_info even if there's no ast leg. Patches: patch10_debug-bc-2.diff (license #6372) patch uploaded by Guenther Kelleter Modified. JIRA ABE-2882 ................ r374534 | rmudgett | 2012-10-05 12:34:10 -0500 (Fri, 05 Oct 2012) | 16 lines chan_misdn: setup_bc() is called too early for an incoming SETUP on TE. This prevents the B channel from being setup for HDLC mode when requested by the bearer capability and config option hdlc=yes. It violates ETS300102 Ch.5.2.3.2: "The user, in any case, must not connect to the channel until a CONNECT ACKNOWLEDGE message has been received." * Call setup_bc() on receipt of CONNECT_ACKNOWLEGDE for PTMP, and on first response to SETUP for PTP. Patches: abe-2881-2.diff (license #6372) patch uploaded by Guenther Kelleter Modified. JIRA ABE-2881 ................ r374535 | rmudgett | 2012-10-05 12:41:05 -0500 (Fri, 05 Oct 2012) | 2 lines chan_misdn: Remove some more deadcode. ................ ........ Merged revisions 374536 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374537 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374538 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374539 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-05 18:42:14 +00:00
ast ? ast_channel_name(ast) : "",
help->l3id,
help->addr,
bc->addr,
bc->l3_id,
bc->display,
bc->active,
bc_state2str(bc->bc_state),
bearer2str(bc->capability),
#ifdef MISDN_1_2
bc->pipeline,
#else
bc->ec_enable,
#endif
help->norxtone, help->notxtone,
bc->holded);
}
}
static char *handle_cli_misdn_show_channels(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct chan_list *help;
switch (cmd) {
case CLI_INIT:
e->command = "misdn show channels";
e->usage =
"Usage: misdn show channels\n"
" Show the internal mISDN channel list\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != 3) {
return CLI_SHOWUSAGE;
}
ast_cli(a->fd, "Channel List: %p\n", cl_te);
/*
* Walking the list and dumping the channel information could
* take awhile. With the list locked for the duration, the
* channel driver cannot process signaling messages. However,
* since this is a CLI command it should not be run very often.
*/
ast_mutex_lock(&cl_te_lock);
for (help = cl_te; help; help = help->next) {
struct misdn_bchannel *bc = help->bc;
struct ast_channel *ast = help->ast;
if (!ast) {
if (!bc) {
ast_cli(a->fd, "chan_list obj. with l3id:%x has no bc and no ast Leg\n", help->l3id);
continue;
}
ast_cli(a->fd, "bc with pid:%d has no Ast Leg\n", bc->pid);
}
if (misdn_debug[0] > 2) {
ast_cli(a->fd, "Bc:%p Ast:%p\n", bc, ast);
}
if (bc) {
print_bc_info(a->fd, help, bc);
} else {
if (help->hold.state != MISDN_HOLD_IDLE) {
ast_cli(a->fd, "ITS A HELD CALL BC:\n");
ast_cli(a->fd, " --> l3_id: %x\n"
" --> dialed:%s\n"
" --> caller:\"%s\" <%s>\n"
" --> hold_port: %d\n"
" --> hold_channel: %d\n",
help->l3id,
ast_channel_exten(ast),
S_COR(ast_channel_caller(ast)->id.name.valid, ast_channel_caller(ast)->id.name.str, ""),
S_COR(ast_channel_caller(ast)->id.number.valid, ast_channel_caller(ast)->id.number.str, ""),
help->hold.port,
help->hold.channel
);
} else {
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
ast_cli(a->fd, "* Channel in unknown STATE !!! Exten:%s, Callerid:%s\n",
ast_channel_exten(ast),
S_COR(ast_channel_caller(ast)->id.number.valid, ast_channel_caller(ast)->id.number.str, ""));
}
}
}
ast_mutex_unlock(&cl_te_lock);
misdn_dump_chanlist();
return CLI_SUCCESS;
}
static char *handle_cli_misdn_show_channel(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct chan_list *help;
switch (cmd) {
case CLI_INIT:
e->command = "misdn show channel";
e->usage =
"Usage: misdn show channel <channel>\n"
" Show an internal mISDN channel\n.";
return NULL;
case CLI_GENERATE:
return complete_ch(a);
}
if (a->argc != 4) {
return CLI_SHOWUSAGE;
}
ast_mutex_lock(&cl_te_lock);
for (help = cl_te; help; help = help->next) {
struct misdn_bchannel *bc = help->bc;
struct ast_channel *ast = help->ast;
if (bc && ast) {
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
if (!strcasecmp(ast_channel_name(ast), a->argv[3])) {
print_bc_info(a->fd, help, bc);
break;
}
}
}
ast_mutex_unlock(&cl_te_lock);
return CLI_SUCCESS;
}
static char *handle_cli_misdn_set_tics(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
switch (cmd) {
case CLI_INIT:
e->command = "misdn set tics";
e->usage =
"Usage: misdn set tics <value>\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != 4) {
return CLI_SHOWUSAGE;
}
/* XXX Wow, this does... a whole lot of nothing... XXX */
MAXTICS = atoi(a->argv[3]);
return CLI_SUCCESS;
}
static char *handle_cli_misdn_show_stacks(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
int port;
switch (cmd) {
case CLI_INIT:
e->command = "misdn show stacks";
e->usage =
"Usage: misdn show stacks\n"
" Show internal mISDN stack_list.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != 3) {
return CLI_SHOWUSAGE;
}
ast_cli(a->fd, "BEGIN STACK_LIST:\n");
for (port = misdn_cfg_get_next_port(0); port > 0;
port = misdn_cfg_get_next_port(port)) {
char buf[128];
get_show_stack_details(port, buf);
ast_cli(a->fd, " %s Debug:%d%s\n", buf, misdn_debug[port], misdn_debug_only[port] ? "(only)" : "");
}
return CLI_SUCCESS;
}
static char *handle_cli_misdn_show_ports_stats(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
int port;
switch (cmd) {
case CLI_INIT:
e->command = "misdn show ports stats";
e->usage =
"Usage: misdn show ports stats\n"
" Show mISDNs channel's call statistics per port.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != 4) {
return CLI_SHOWUSAGE;
}
ast_cli(a->fd, "Port\tin_calls\tout_calls\n");
for (port = misdn_cfg_get_next_port(0); port > 0;
port = misdn_cfg_get_next_port(port)) {
ast_cli(a->fd, "%d\t%d\t\t%d\n", port, misdn_in_calls[port], misdn_out_calls[port]);
}
ast_cli(a->fd, "\n");
return CLI_SUCCESS;
}
static char *handle_cli_misdn_show_port(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
int port;
char buf[128];
switch (cmd) {
case CLI_INIT:
e->command = "misdn show port";
e->usage =
"Usage: misdn show port <port>\n"
" Show detailed information for given port.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != 4) {
return CLI_SHOWUSAGE;
}
port = atoi(a->argv[3]);
ast_cli(a->fd, "BEGIN STACK_LIST:\n");
get_show_stack_details(port, buf);
ast_cli(a->fd, " %s Debug:%d%s\n", buf, misdn_debug[port], misdn_debug_only[port] ? "(only)" : "");
return CLI_SUCCESS;
}
#if defined(AST_MISDN_ENHANCEMENTS) && defined(CCBS_TEST_MESSAGES)
static const struct FacParm Fac_Msgs[] = {
/* *INDENT-OFF* */
[0].Function = Fac_ERROR,
[0].u.ERROR.invokeId = 8,
[0].u.ERROR.errorValue = FacError_CCBS_AlreadyAccepted,
[1].Function = Fac_RESULT,
[1].u.RESULT.InvokeID = 9,
[2].Function = Fac_REJECT,
[2].u.REJECT.Code = FacReject_Gen_BadlyStructuredComponent,
[3].Function = Fac_REJECT,
[3].u.REJECT.InvokeIDPresent = 1,
[3].u.REJECT.InvokeID = 10,
[3].u.REJECT.Code = FacReject_Inv_InitiatorReleasing,
[4].Function = Fac_REJECT,
[4].u.REJECT.InvokeIDPresent = 1,
[4].u.REJECT.InvokeID = 11,
[4].u.REJECT.Code = FacReject_Res_MistypedResult,
[5].Function = Fac_REJECT,
[5].u.REJECT.InvokeIDPresent = 1,
[5].u.REJECT.InvokeID = 12,
[5].u.REJECT.Code = FacReject_Err_ErrorResponseUnexpected,
[6].Function = Fac_StatusRequest,
[6].u.StatusRequest.InvokeID = 13,
[6].u.StatusRequest.ComponentType = FacComponent_Invoke,
[6].u.StatusRequest.Component.Invoke.Q931ie.Bc.Length = 2,
[6].u.StatusRequest.Component.Invoke.Q931ie.Bc.Contents = "AB",
[6].u.StatusRequest.Component.Invoke.Q931ie.Llc.Length = 3,
[6].u.StatusRequest.Component.Invoke.Q931ie.Llc.Contents = "CDE",
[6].u.StatusRequest.Component.Invoke.Q931ie.Hlc.Length = 4,
[6].u.StatusRequest.Component.Invoke.Q931ie.Hlc.Contents = "FGHI",
[6].u.StatusRequest.Component.Invoke.CompatibilityMode = 1,
[7].Function = Fac_StatusRequest,
[7].u.StatusRequest.InvokeID = 14,
[7].u.StatusRequest.ComponentType = FacComponent_Result,
[7].u.StatusRequest.Component.Result.Status = 2,
[8].Function = Fac_CallInfoRetain,
[8].u.CallInfoRetain.InvokeID = 15,
[8].u.CallInfoRetain.CallLinkageID = 115,
[9].Function = Fac_EraseCallLinkageID,
[9].u.EraseCallLinkageID.InvokeID = 16,
[9].u.EraseCallLinkageID.CallLinkageID = 105,
[10].Function = Fac_CCBSDeactivate,
[10].u.CCBSDeactivate.InvokeID = 17,
[10].u.CCBSDeactivate.ComponentType = FacComponent_Invoke,
[10].u.CCBSDeactivate.Component.Invoke.CCBSReference = 2,
[11].Function = Fac_CCBSDeactivate,
[11].u.CCBSDeactivate.InvokeID = 18,
[11].u.CCBSDeactivate.ComponentType = FacComponent_Result,
[12].Function = Fac_CCBSErase,
[12].u.CCBSErase.InvokeID = 19,
[12].u.CCBSErase.Q931ie.Bc.Length = 2,
[12].u.CCBSErase.Q931ie.Bc.Contents = "JK",
[12].u.CCBSErase.AddressOfB.Party.Type = 0,
[12].u.CCBSErase.AddressOfB.Party.LengthOfNumber = 5,
[12].u.CCBSErase.AddressOfB.Party.Number = "33403",
[12].u.CCBSErase.AddressOfB.Subaddress.Type = 0,
[12].u.CCBSErase.AddressOfB.Subaddress.Length = 4,
[12].u.CCBSErase.AddressOfB.Subaddress.u.UserSpecified.Information = "3748",
[12].u.CCBSErase.RecallMode = 1,
[12].u.CCBSErase.CCBSReference = 102,
[12].u.CCBSErase.Reason = 3,
[13].Function = Fac_CCBSErase,
[13].u.CCBSErase.InvokeID = 20,
[13].u.CCBSErase.Q931ie.Bc.Length = 2,
[13].u.CCBSErase.Q931ie.Bc.Contents = "JK",
[13].u.CCBSErase.AddressOfB.Party.Type = 1,
[13].u.CCBSErase.AddressOfB.Party.LengthOfNumber = 11,
[13].u.CCBSErase.AddressOfB.Party.TypeOfNumber = 1,
[13].u.CCBSErase.AddressOfB.Party.Number = "18003020102",
[13].u.CCBSErase.AddressOfB.Subaddress.Type = 0,
[13].u.CCBSErase.AddressOfB.Subaddress.Length = 4,
[13].u.CCBSErase.AddressOfB.Subaddress.u.UserSpecified.OddCountPresent = 1,
[13].u.CCBSErase.AddressOfB.Subaddress.u.UserSpecified.OddCount = 1,
[13].u.CCBSErase.AddressOfB.Subaddress.u.UserSpecified.Information = "3748",
[13].u.CCBSErase.RecallMode = 1,
[13].u.CCBSErase.CCBSReference = 102,
[13].u.CCBSErase.Reason = 3,
[14].Function = Fac_CCBSErase,
[14].u.CCBSErase.InvokeID = 21,
[14].u.CCBSErase.Q931ie.Bc.Length = 2,
[14].u.CCBSErase.Q931ie.Bc.Contents = "JK",
[14].u.CCBSErase.AddressOfB.Party.Type = 2,
[14].u.CCBSErase.AddressOfB.Party.LengthOfNumber = 4,
[14].u.CCBSErase.AddressOfB.Party.Number = "1803",
[14].u.CCBSErase.AddressOfB.Subaddress.Type = 1,
[14].u.CCBSErase.AddressOfB.Subaddress.Length = 4,
[14].u.CCBSErase.AddressOfB.Subaddress.u.Nsap = "6492",
[14].u.CCBSErase.RecallMode = 1,
[14].u.CCBSErase.CCBSReference = 102,
[14].u.CCBSErase.Reason = 3,
[15].Function = Fac_CCBSErase,
[15].u.CCBSErase.InvokeID = 22,
[15].u.CCBSErase.Q931ie.Bc.Length = 2,
[15].u.CCBSErase.Q931ie.Bc.Contents = "JK",
[15].u.CCBSErase.AddressOfB.Party.Type = 3,
[15].u.CCBSErase.AddressOfB.Party.LengthOfNumber = 4,
[15].u.CCBSErase.AddressOfB.Party.Number = "1803",
[15].u.CCBSErase.RecallMode = 1,
[15].u.CCBSErase.CCBSReference = 102,
[15].u.CCBSErase.Reason = 3,
[16].Function = Fac_CCBSErase,
[16].u.CCBSErase.InvokeID = 23,
[16].u.CCBSErase.Q931ie.Bc.Length = 2,
[16].u.CCBSErase.Q931ie.Bc.Contents = "JK",
[16].u.CCBSErase.AddressOfB.Party.Type = 4,
[16].u.CCBSErase.AddressOfB.Party.LengthOfNumber = 4,
[16].u.CCBSErase.AddressOfB.Party.Number = "1803",
[16].u.CCBSErase.RecallMode = 1,
[16].u.CCBSErase.CCBSReference = 102,
[16].u.CCBSErase.Reason = 3,
[17].Function = Fac_CCBSErase,
[17].u.CCBSErase.InvokeID = 24,
[17].u.CCBSErase.Q931ie.Bc.Length = 2,
[17].u.CCBSErase.Q931ie.Bc.Contents = "JK",
[17].u.CCBSErase.AddressOfB.Party.Type = 5,
[17].u.CCBSErase.AddressOfB.Party.LengthOfNumber = 11,
[17].u.CCBSErase.AddressOfB.Party.TypeOfNumber = 4,
[17].u.CCBSErase.AddressOfB.Party.Number = "18003020102",
[17].u.CCBSErase.RecallMode = 1,
[17].u.CCBSErase.CCBSReference = 102,
[17].u.CCBSErase.Reason = 3,
[18].Function = Fac_CCBSErase,
[18].u.CCBSErase.InvokeID = 25,
[18].u.CCBSErase.Q931ie.Bc.Length = 2,
[18].u.CCBSErase.Q931ie.Bc.Contents = "JK",
[18].u.CCBSErase.AddressOfB.Party.Type = 8,
[18].u.CCBSErase.AddressOfB.Party.LengthOfNumber = 4,
[18].u.CCBSErase.AddressOfB.Party.Number = "1803",
[18].u.CCBSErase.RecallMode = 1,
[18].u.CCBSErase.CCBSReference = 102,
[18].u.CCBSErase.Reason = 3,
[19].Function = Fac_CCBSRemoteUserFree,
[19].u.CCBSRemoteUserFree.InvokeID = 26,
[19].u.CCBSRemoteUserFree.Q931ie.Bc.Length = 2,
[19].u.CCBSRemoteUserFree.Q931ie.Bc.Contents = "JK",
[19].u.CCBSRemoteUserFree.AddressOfB.Party.Type = 8,
[19].u.CCBSRemoteUserFree.AddressOfB.Party.LengthOfNumber = 4,
[19].u.CCBSRemoteUserFree.AddressOfB.Party.Number = "1803",
[19].u.CCBSRemoteUserFree.RecallMode = 1,
[19].u.CCBSRemoteUserFree.CCBSReference = 102,
[20].Function = Fac_CCBSCall,
[20].u.CCBSCall.InvokeID = 27,
[20].u.CCBSCall.CCBSReference = 115,
[21].Function = Fac_CCBSStatusRequest,
[21].u.CCBSStatusRequest.InvokeID = 28,
[21].u.CCBSStatusRequest.ComponentType = FacComponent_Invoke,
[21].u.CCBSStatusRequest.Component.Invoke.Q931ie.Bc.Length = 2,
[21].u.CCBSStatusRequest.Component.Invoke.Q931ie.Bc.Contents = "JK",
[21].u.CCBSStatusRequest.Component.Invoke.RecallMode = 1,
[21].u.CCBSStatusRequest.Component.Invoke.CCBSReference = 102,
[22].Function = Fac_CCBSStatusRequest,
[22].u.CCBSStatusRequest.InvokeID = 29,
[22].u.CCBSStatusRequest.ComponentType = FacComponent_Result,
[22].u.CCBSStatusRequest.Component.Result.Free = 1,
[23].Function = Fac_CCBSBFree,
[23].u.CCBSBFree.InvokeID = 30,
[23].u.CCBSBFree.Q931ie.Bc.Length = 2,
[23].u.CCBSBFree.Q931ie.Bc.Contents = "JK",
[23].u.CCBSBFree.AddressOfB.Party.Type = 8,
[23].u.CCBSBFree.AddressOfB.Party.LengthOfNumber = 4,
[23].u.CCBSBFree.AddressOfB.Party.Number = "1803",
[23].u.CCBSBFree.RecallMode = 1,
[23].u.CCBSBFree.CCBSReference = 14,
[24].Function = Fac_CCBSStopAlerting,
[24].u.CCBSStopAlerting.InvokeID = 31,
[24].u.CCBSStopAlerting.CCBSReference = 37,
[25].Function = Fac_CCBSRequest,
[25].u.CCBSRequest.InvokeID = 32,
[25].u.CCBSRequest.ComponentType = FacComponent_Invoke,
[25].u.CCBSRequest.Component.Invoke.CallLinkageID = 57,
[26].Function = Fac_CCBSRequest,
[26].u.CCBSRequest.InvokeID = 33,
[26].u.CCBSRequest.ComponentType = FacComponent_Result,
[26].u.CCBSRequest.Component.Result.RecallMode = 1,
[26].u.CCBSRequest.Component.Result.CCBSReference = 102,
[27].Function = Fac_CCBSInterrogate,
[27].u.CCBSInterrogate.InvokeID = 34,
[27].u.CCBSInterrogate.ComponentType = FacComponent_Invoke,
[27].u.CCBSInterrogate.Component.Invoke.AParty.Type = 8,
[27].u.CCBSInterrogate.Component.Invoke.AParty.LengthOfNumber = 4,
[27].u.CCBSInterrogate.Component.Invoke.AParty.Number = "1803",
[27].u.CCBSInterrogate.Component.Invoke.CCBSReferencePresent = 1,
[27].u.CCBSInterrogate.Component.Invoke.CCBSReference = 76,
[28].Function = Fac_CCBSInterrogate,
[28].u.CCBSInterrogate.InvokeID = 35,
[28].u.CCBSInterrogate.ComponentType = FacComponent_Invoke,
[28].u.CCBSInterrogate.Component.Invoke.AParty.Type = 8,
[28].u.CCBSInterrogate.Component.Invoke.AParty.LengthOfNumber = 4,
[28].u.CCBSInterrogate.Component.Invoke.AParty.Number = "1803",
[29].Function = Fac_CCBSInterrogate,
[29].u.CCBSInterrogate.InvokeID = 36,
[29].u.CCBSInterrogate.ComponentType = FacComponent_Invoke,
[29].u.CCBSInterrogate.Component.Invoke.CCBSReferencePresent = 1,
[29].u.CCBSInterrogate.Component.Invoke.CCBSReference = 76,
[30].Function = Fac_CCBSInterrogate,
[30].u.CCBSInterrogate.InvokeID = 37,
[30].u.CCBSInterrogate.ComponentType = FacComponent_Invoke,
[31].Function = Fac_CCBSInterrogate,
[31].u.CCBSInterrogate.InvokeID = 38,
[31].u.CCBSInterrogate.ComponentType = FacComponent_Result,
[31].u.CCBSInterrogate.Component.Result.RecallMode = 1,
[32].Function = Fac_CCBSInterrogate,
[32].u.CCBSInterrogate.InvokeID = 39,
[32].u.CCBSInterrogate.ComponentType = FacComponent_Result,
[32].u.CCBSInterrogate.Component.Result.RecallMode = 1,
[32].u.CCBSInterrogate.Component.Result.NumRecords = 1,
[32].u.CCBSInterrogate.Component.Result.CallDetails[0].CCBSReference = 12,
[32].u.CCBSInterrogate.Component.Result.CallDetails[0].Q931ie.Bc.Length = 2,
[32].u.CCBSInterrogate.Component.Result.CallDetails[0].Q931ie.Bc.Contents = "JK",
[32].u.CCBSInterrogate.Component.Result.CallDetails[0].AddressOfB.Party.Type = 8,
[32].u.CCBSInterrogate.Component.Result.CallDetails[0].AddressOfB.Party.LengthOfNumber = 4,
[32].u.CCBSInterrogate.Component.Result.CallDetails[0].AddressOfB.Party.Number = "1803",
[32].u.CCBSInterrogate.Component.Result.CallDetails[0].SubaddressOfA.Type = 1,
[32].u.CCBSInterrogate.Component.Result.CallDetails[0].SubaddressOfA.Length = 4,
[32].u.CCBSInterrogate.Component.Result.CallDetails[0].SubaddressOfA.u.Nsap = "6492",
[33].Function = Fac_CCBSInterrogate,
[33].u.CCBSInterrogate.InvokeID = 40,
[33].u.CCBSInterrogate.ComponentType = FacComponent_Result,
[33].u.CCBSInterrogate.Component.Result.RecallMode = 1,
[33].u.CCBSInterrogate.Component.Result.NumRecords = 2,
[33].u.CCBSInterrogate.Component.Result.CallDetails[0].CCBSReference = 12,
[33].u.CCBSInterrogate.Component.Result.CallDetails[0].Q931ie.Bc.Length = 2,
[33].u.CCBSInterrogate.Component.Result.CallDetails[0].Q931ie.Bc.Contents = "JK",
[33].u.CCBSInterrogate.Component.Result.CallDetails[0].AddressOfB.Party.Type = 8,
[33].u.CCBSInterrogate.Component.Result.CallDetails[0].AddressOfB.Party.LengthOfNumber = 4,
[33].u.CCBSInterrogate.Component.Result.CallDetails[0].AddressOfB.Party.Number = "1803",
[33].u.CCBSInterrogate.Component.Result.CallDetails[1].CCBSReference = 102,
[33].u.CCBSInterrogate.Component.Result.CallDetails[1].Q931ie.Bc.Length = 2,
[33].u.CCBSInterrogate.Component.Result.CallDetails[1].Q931ie.Bc.Contents = "LM",
[33].u.CCBSInterrogate.Component.Result.CallDetails[1].AddressOfB.Party.Type = 8,
[33].u.CCBSInterrogate.Component.Result.CallDetails[1].AddressOfB.Party.LengthOfNumber = 4,
[33].u.CCBSInterrogate.Component.Result.CallDetails[1].AddressOfB.Party.Number = "6229",
[33].u.CCBSInterrogate.Component.Result.CallDetails[1].AddressOfB.Subaddress.Type = 1,
[33].u.CCBSInterrogate.Component.Result.CallDetails[1].AddressOfB.Subaddress.Length = 4,
[33].u.CCBSInterrogate.Component.Result.CallDetails[1].AddressOfB.Subaddress.u.Nsap = "8592",
[33].u.CCBSInterrogate.Component.Result.CallDetails[1].SubaddressOfA.Type = 1,
[33].u.CCBSInterrogate.Component.Result.CallDetails[1].SubaddressOfA.Length = 4,
[33].u.CCBSInterrogate.Component.Result.CallDetails[1].SubaddressOfA.u.Nsap = "6492",
[34].Function = Fac_CCNRRequest,
[34].u.CCNRRequest.InvokeID = 512,
[34].u.CCNRRequest.ComponentType = FacComponent_Invoke,
[34].u.CCNRRequest.Component.Invoke.CallLinkageID = 57,
[35].Function = Fac_CCNRRequest,
[35].u.CCNRRequest.InvokeID = 150,
[35].u.CCNRRequest.ComponentType = FacComponent_Result,
[35].u.CCNRRequest.Component.Result.RecallMode = 1,
[35].u.CCNRRequest.Component.Result.CCBSReference = 102,
[36].Function = Fac_CCNRInterrogate,
[36].u.CCNRInterrogate.InvokeID = -129,
[36].u.CCNRInterrogate.ComponentType = FacComponent_Invoke,
[37].Function = Fac_CCNRInterrogate,
[37].u.CCNRInterrogate.InvokeID = -3,
[37].u.CCNRInterrogate.ComponentType = FacComponent_Result,
[37].u.CCNRInterrogate.Component.Result.RecallMode = 1,
[38].Function = Fac_CCBS_T_Call,
[38].u.EctExecute.InvokeID = 41,
[39].Function = Fac_CCBS_T_Suspend,
[39].u.EctExecute.InvokeID = 42,
[40].Function = Fac_CCBS_T_Resume,
[40].u.EctExecute.InvokeID = 43,
[41].Function = Fac_CCBS_T_RemoteUserFree,
[41].u.EctExecute.InvokeID = 44,
[42].Function = Fac_CCBS_T_Available,
[42].u.EctExecute.InvokeID = 45,
[43].Function = Fac_CCBS_T_Request,
[43].u.CCBS_T_Request.InvokeID = 46,
[43].u.CCBS_T_Request.ComponentType = FacComponent_Invoke,
[43].u.CCBS_T_Request.Component.Invoke.Destination.Party.Type = 8,
[43].u.CCBS_T_Request.Component.Invoke.Destination.Party.LengthOfNumber = 4,
[43].u.CCBS_T_Request.Component.Invoke.Destination.Party.Number = "6229",
[43].u.CCBS_T_Request.Component.Invoke.Q931ie.Bc.Length = 2,
[43].u.CCBS_T_Request.Component.Invoke.Q931ie.Bc.Contents = "LM",
[43].u.CCBS_T_Request.Component.Invoke.RetentionSupported = 1,
[43].u.CCBS_T_Request.Component.Invoke.PresentationAllowedIndicatorPresent = 1,
[43].u.CCBS_T_Request.Component.Invoke.PresentationAllowedIndicator = 1,
[43].u.CCBS_T_Request.Component.Invoke.Originating.Party.Type = 8,
[43].u.CCBS_T_Request.Component.Invoke.Originating.Party.LengthOfNumber = 4,
[43].u.CCBS_T_Request.Component.Invoke.Originating.Party.Number = "9864",
[44].Function = Fac_CCBS_T_Request,
[44].u.CCBS_T_Request.InvokeID = 47,
[44].u.CCBS_T_Request.ComponentType = FacComponent_Invoke,
[44].u.CCBS_T_Request.Component.Invoke.Destination.Party.Type = 8,
[44].u.CCBS_T_Request.Component.Invoke.Destination.Party.LengthOfNumber = 4,
[44].u.CCBS_T_Request.Component.Invoke.Destination.Party.Number = "6229",
[44].u.CCBS_T_Request.Component.Invoke.Q931ie.Bc.Length = 2,
[44].u.CCBS_T_Request.Component.Invoke.Q931ie.Bc.Contents = "LM",
[44].u.CCBS_T_Request.Component.Invoke.PresentationAllowedIndicatorPresent = 1,
[44].u.CCBS_T_Request.Component.Invoke.PresentationAllowedIndicator = 1,
[44].u.CCBS_T_Request.Component.Invoke.Originating.Party.Type = 8,
[44].u.CCBS_T_Request.Component.Invoke.Originating.Party.LengthOfNumber = 4,
[44].u.CCBS_T_Request.Component.Invoke.Originating.Party.Number = "9864",
[45].Function = Fac_CCBS_T_Request,
[45].u.CCBS_T_Request.InvokeID = 48,
[45].u.CCBS_T_Request.ComponentType = FacComponent_Invoke,
[45].u.CCBS_T_Request.Component.Invoke.Destination.Party.Type = 8,
[45].u.CCBS_T_Request.Component.Invoke.Destination.Party.LengthOfNumber = 4,
[45].u.CCBS_T_Request.Component.Invoke.Destination.Party.Number = "6229",
[45].u.CCBS_T_Request.Component.Invoke.Q931ie.Bc.Length = 2,
[45].u.CCBS_T_Request.Component.Invoke.Q931ie.Bc.Contents = "LM",
[45].u.CCBS_T_Request.Component.Invoke.Originating.Party.Type = 8,
[45].u.CCBS_T_Request.Component.Invoke.Originating.Party.LengthOfNumber = 4,
[45].u.CCBS_T_Request.Component.Invoke.Originating.Party.Number = "9864",
[46].Function = Fac_CCBS_T_Request,
[46].u.CCBS_T_Request.InvokeID = 49,
[46].u.CCBS_T_Request.ComponentType = FacComponent_Invoke,
[46].u.CCBS_T_Request.Component.Invoke.Destination.Party.Type = 8,
[46].u.CCBS_T_Request.Component.Invoke.Destination.Party.LengthOfNumber = 4,
[46].u.CCBS_T_Request.Component.Invoke.Destination.Party.Number = "6229",
[46].u.CCBS_T_Request.Component.Invoke.Q931ie.Bc.Length = 2,
[46].u.CCBS_T_Request.Component.Invoke.Q931ie.Bc.Contents = "LM",
[46].u.CCBS_T_Request.Component.Invoke.PresentationAllowedIndicatorPresent = 1,
[46].u.CCBS_T_Request.Component.Invoke.PresentationAllowedIndicator = 1,
[47].Function = Fac_CCBS_T_Request,
[47].u.CCBS_T_Request.InvokeID = 50,
[47].u.CCBS_T_Request.ComponentType = FacComponent_Invoke,
[47].u.CCBS_T_Request.Component.Invoke.Destination.Party.Type = 8,
[47].u.CCBS_T_Request.Component.Invoke.Destination.Party.LengthOfNumber = 4,
[47].u.CCBS_T_Request.Component.Invoke.Destination.Party.Number = "6229",
[47].u.CCBS_T_Request.Component.Invoke.Q931ie.Bc.Length = 2,
[47].u.CCBS_T_Request.Component.Invoke.Q931ie.Bc.Contents = "LM",
[48].Function = Fac_CCBS_T_Request,
[48].u.CCBS_T_Request.InvokeID = 51,
[48].u.CCBS_T_Request.ComponentType = FacComponent_Result,
[48].u.CCBS_T_Request.Component.Result.RetentionSupported = 1,
[49].Function = Fac_CCNR_T_Request,
[49].u.CCNR_T_Request.InvokeID = 52,
[49].u.CCNR_T_Request.ComponentType = FacComponent_Invoke,
[49].u.CCNR_T_Request.Component.Invoke.Destination.Party.Type = 8,
[49].u.CCNR_T_Request.Component.Invoke.Destination.Party.LengthOfNumber = 4,
[49].u.CCNR_T_Request.Component.Invoke.Destination.Party.Number = "6229",
[49].u.CCNR_T_Request.Component.Invoke.Q931ie.Bc.Length = 2,
[49].u.CCNR_T_Request.Component.Invoke.Q931ie.Bc.Contents = "LM",
[50].Function = Fac_CCNR_T_Request,
[50].u.CCNR_T_Request.InvokeID = 53,
[50].u.CCNR_T_Request.ComponentType = FacComponent_Result,
[50].u.CCNR_T_Request.Component.Result.RetentionSupported = 1,
[51].Function = Fac_EctExecute,
[51].u.EctExecute.InvokeID = 54,
[52].Function = Fac_ExplicitEctExecute,
[52].u.ExplicitEctExecute.InvokeID = 55,
[52].u.ExplicitEctExecute.LinkID = 23,
[53].Function = Fac_RequestSubaddress,
[53].u.RequestSubaddress.InvokeID = 56,
[54].Function = Fac_SubaddressTransfer,
[54].u.SubaddressTransfer.InvokeID = 57,
[54].u.SubaddressTransfer.Subaddress.Type = 1,
[54].u.SubaddressTransfer.Subaddress.Length = 4,
[54].u.SubaddressTransfer.Subaddress.u.Nsap = "6492",
[55].Function = Fac_EctLinkIdRequest,
[55].u.EctLinkIdRequest.InvokeID = 58,
[55].u.EctLinkIdRequest.ComponentType = FacComponent_Invoke,
[56].Function = Fac_EctLinkIdRequest,
[56].u.EctLinkIdRequest.InvokeID = 59,
[56].u.EctLinkIdRequest.ComponentType = FacComponent_Result,
[56].u.EctLinkIdRequest.Component.Result.LinkID = 76,
[57].Function = Fac_EctInform,
[57].u.EctInform.InvokeID = 60,
[57].u.EctInform.Status = 1,
[57].u.EctInform.RedirectionPresent = 1,
[57].u.EctInform.Redirection.Type = 0,
[57].u.EctInform.Redirection.Unscreened.Type = 8,
[57].u.EctInform.Redirection.Unscreened.LengthOfNumber = 4,
[57].u.EctInform.Redirection.Unscreened.Number = "6229",
[58].Function = Fac_EctInform,
[58].u.EctInform.InvokeID = 61,
[58].u.EctInform.Status = 1,
[58].u.EctInform.RedirectionPresent = 1,
[58].u.EctInform.Redirection.Type = 1,
[59].Function = Fac_EctInform,
[59].u.EctInform.InvokeID = 62,
[59].u.EctInform.Status = 1,
[59].u.EctInform.RedirectionPresent = 1,
[59].u.EctInform.Redirection.Type = 2,
[60].Function = Fac_EctInform,
[60].u.EctInform.InvokeID = 63,
[60].u.EctInform.Status = 1,
[60].u.EctInform.RedirectionPresent = 1,
[60].u.EctInform.Redirection.Type = 3,
[60].u.EctInform.Redirection.Unscreened.Type = 8,
[60].u.EctInform.Redirection.Unscreened.LengthOfNumber = 4,
[60].u.EctInform.Redirection.Unscreened.Number = "3340",
[61].Function = Fac_EctInform,
[61].u.EctInform.InvokeID = 64,
[61].u.EctInform.Status = 1,
[61].u.EctInform.RedirectionPresent = 0,
[62].Function = Fac_EctLoopTest,
[62].u.EctLoopTest.InvokeID = 65,
[62].u.EctLoopTest.ComponentType = FacComponent_Invoke,
[62].u.EctLoopTest.Component.Invoke.CallTransferID = 7,
[63].Function = Fac_EctLoopTest,
[63].u.EctLoopTest.InvokeID = 66,
[63].u.EctLoopTest.ComponentType = FacComponent_Result,
[63].u.EctLoopTest.Component.Result.LoopResult = 2,
[64].Function = Fac_ActivationDiversion,
[64].u.ActivationDiversion.InvokeID = 67,
[64].u.ActivationDiversion.ComponentType = FacComponent_Invoke,
[64].u.ActivationDiversion.Component.Invoke.Procedure = 2,
[64].u.ActivationDiversion.Component.Invoke.BasicService = 3,
[64].u.ActivationDiversion.Component.Invoke.ForwardedTo.Party.Type = 4,
[64].u.ActivationDiversion.Component.Invoke.ForwardedTo.Party.LengthOfNumber = 4,
[64].u.ActivationDiversion.Component.Invoke.ForwardedTo.Party.Number = "1803",
[64].u.ActivationDiversion.Component.Invoke.ServedUser.Type = 4,
[64].u.ActivationDiversion.Component.Invoke.ServedUser.LengthOfNumber = 4,
[64].u.ActivationDiversion.Component.Invoke.ServedUser.Number = "5398",
[65].Function = Fac_ActivationDiversion,
[65].u.ActivationDiversion.InvokeID = 68,
[65].u.ActivationDiversion.ComponentType = FacComponent_Invoke,
[65].u.ActivationDiversion.Component.Invoke.Procedure = 1,
[65].u.ActivationDiversion.Component.Invoke.BasicService = 5,
[65].u.ActivationDiversion.Component.Invoke.ForwardedTo.Party.Type = 4,
[65].u.ActivationDiversion.Component.Invoke.ForwardedTo.Party.LengthOfNumber = 4,
[65].u.ActivationDiversion.Component.Invoke.ForwardedTo.Party.Number = "1803",
[66].Function = Fac_ActivationDiversion,
[66].u.ActivationDiversion.InvokeID = 69,
[66].u.ActivationDiversion.ComponentType = FacComponent_Result,
[67].Function = Fac_DeactivationDiversion,
[67].u.DeactivationDiversion.InvokeID = 70,
[67].u.DeactivationDiversion.ComponentType = FacComponent_Invoke,
[67].u.DeactivationDiversion.Component.Invoke.Procedure = 1,
[67].u.DeactivationDiversion.Component.Invoke.BasicService = 5,
[68].Function = Fac_DeactivationDiversion,
[68].u.DeactivationDiversion.InvokeID = 71,
[68].u.DeactivationDiversion.ComponentType = FacComponent_Result,
[69].Function = Fac_ActivationStatusNotificationDiv,
[69].u.ActivationStatusNotificationDiv.InvokeID = 72,
[69].u.ActivationStatusNotificationDiv.Procedure = 1,
[69].u.ActivationStatusNotificationDiv.BasicService = 5,
[69].u.ActivationStatusNotificationDiv.ForwardedTo.Party.Type = 4,
[69].u.ActivationStatusNotificationDiv.ForwardedTo.Party.LengthOfNumber = 4,
[69].u.ActivationStatusNotificationDiv.ForwardedTo.Party.Number = "1803",
[70].Function = Fac_DeactivationStatusNotificationDiv,
[70].u.DeactivationStatusNotificationDiv.InvokeID = 73,
[70].u.DeactivationStatusNotificationDiv.Procedure = 1,
[70].u.DeactivationStatusNotificationDiv.BasicService = 5,
[71].Function = Fac_InterrogationDiversion,
[71].u.InterrogationDiversion.InvokeID = 74,
[71].u.InterrogationDiversion.ComponentType = FacComponent_Invoke,
[71].u.InterrogationDiversion.Component.Invoke.Procedure = 1,
[71].u.InterrogationDiversion.Component.Invoke.BasicService = 5,
[72].Function = Fac_InterrogationDiversion,
[72].u.InterrogationDiversion.InvokeID = 75,
[72].u.InterrogationDiversion.ComponentType = FacComponent_Invoke,
[72].u.InterrogationDiversion.Component.Invoke.Procedure = 1,
[73].Function = Fac_InterrogationDiversion,
[73].u.InterrogationDiversion.InvokeID = 76,
[73].u.InterrogationDiversion.ComponentType = FacComponent_Result,
[73].u.InterrogationDiversion.Component.Result.NumRecords = 2,
[73].u.InterrogationDiversion.Component.Result.List[0].Procedure = 2,
[73].u.InterrogationDiversion.Component.Result.List[0].BasicService = 5,
[73].u.InterrogationDiversion.Component.Result.List[0].ForwardedTo.Party.Type = 4,
[73].u.InterrogationDiversion.Component.Result.List[0].ForwardedTo.Party.LengthOfNumber = 4,
[73].u.InterrogationDiversion.Component.Result.List[0].ForwardedTo.Party.Number = "1803",
[73].u.InterrogationDiversion.Component.Result.List[1].Procedure = 1,
[73].u.InterrogationDiversion.Component.Result.List[1].BasicService = 3,
[73].u.InterrogationDiversion.Component.Result.List[1].ForwardedTo.Party.Type = 4,
[73].u.InterrogationDiversion.Component.Result.List[1].ForwardedTo.Party.LengthOfNumber = 4,
[73].u.InterrogationDiversion.Component.Result.List[1].ForwardedTo.Party.Number = "1903",
[73].u.InterrogationDiversion.Component.Result.List[1].ServedUser.Type = 4,
[73].u.InterrogationDiversion.Component.Result.List[1].ServedUser.LengthOfNumber = 4,
[73].u.InterrogationDiversion.Component.Result.List[1].ServedUser.Number = "5398",
[74].Function = Fac_DiversionInformation,
[74].u.DiversionInformation.InvokeID = 77,
[74].u.DiversionInformation.DiversionReason = 3,
[74].u.DiversionInformation.BasicService = 5,
[74].u.DiversionInformation.ServedUserSubaddress.Type = 1,
[74].u.DiversionInformation.ServedUserSubaddress.Length = 4,
[74].u.DiversionInformation.ServedUserSubaddress.u.Nsap = "6492",
[74].u.DiversionInformation.CallingAddressPresent = 1,
[74].u.DiversionInformation.CallingAddress.Type = 0,
[74].u.DiversionInformation.CallingAddress.Address.ScreeningIndicator = 3,
[74].u.DiversionInformation.CallingAddress.Address.Party.Type = 4,
[74].u.DiversionInformation.CallingAddress.Address.Party.LengthOfNumber = 4,
[74].u.DiversionInformation.CallingAddress.Address.Party.Number = "1803",
[74].u.DiversionInformation.OriginalCalledPresent = 1,
[74].u.DiversionInformation.OriginalCalled.Type = 1,
[74].u.DiversionInformation.LastDivertingPresent = 1,
[74].u.DiversionInformation.LastDiverting.Type = 2,
[74].u.DiversionInformation.LastDivertingReasonPresent = 1,
[74].u.DiversionInformation.LastDivertingReason = 3,
[74].u.DiversionInformation.UserInfo.Length = 5,
[74].u.DiversionInformation.UserInfo.Contents = "79828",
[75].Function = Fac_DiversionInformation,
[75].u.DiversionInformation.InvokeID = 78,
[75].u.DiversionInformation.DiversionReason = 3,
[75].u.DiversionInformation.BasicService = 5,
[75].u.DiversionInformation.CallingAddressPresent = 1,
[75].u.DiversionInformation.CallingAddress.Type = 1,
[75].u.DiversionInformation.OriginalCalledPresent = 1,
[75].u.DiversionInformation.OriginalCalled.Type = 2,
[75].u.DiversionInformation.LastDivertingPresent = 1,
[75].u.DiversionInformation.LastDiverting.Type = 1,
[76].Function = Fac_DiversionInformation,
[76].u.DiversionInformation.InvokeID = 79,
[76].u.DiversionInformation.DiversionReason = 2,
[76].u.DiversionInformation.BasicService = 3,
[76].u.DiversionInformation.CallingAddressPresent = 1,
[76].u.DiversionInformation.CallingAddress.Type = 2,
[77].Function = Fac_DiversionInformation,
[77].u.DiversionInformation.InvokeID = 80,
[77].u.DiversionInformation.DiversionReason = 3,
[77].u.DiversionInformation.BasicService = 5,
[77].u.DiversionInformation.CallingAddressPresent = 1,
[77].u.DiversionInformation.CallingAddress.Type = 3,
[77].u.DiversionInformation.CallingAddress.Address.ScreeningIndicator = 2,
[77].u.DiversionInformation.CallingAddress.Address.Party.Type = 4,
[77].u.DiversionInformation.CallingAddress.Address.Party.LengthOfNumber = 4,
[77].u.DiversionInformation.CallingAddress.Address.Party.Number = "1803",
[78].Function = Fac_DiversionInformation,
[78].u.DiversionInformation.InvokeID = 81,
[78].u.DiversionInformation.DiversionReason = 2,
[78].u.DiversionInformation.BasicService = 4,
[78].u.DiversionInformation.UserInfo.Length = 5,
[78].u.DiversionInformation.UserInfo.Contents = "79828",
[79].Function = Fac_DiversionInformation,
[79].u.DiversionInformation.InvokeID = 82,
[79].u.DiversionInformation.DiversionReason = 2,
[79].u.DiversionInformation.BasicService = 4,
[80].Function = Fac_CallDeflection,
[80].u.CallDeflection.InvokeID = 83,
[80].u.CallDeflection.ComponentType = FacComponent_Invoke,
[80].u.CallDeflection.Component.Invoke.Deflection.Party.Type = 4,
[80].u.CallDeflection.Component.Invoke.Deflection.Party.LengthOfNumber = 4,
[80].u.CallDeflection.Component.Invoke.Deflection.Party.Number = "1803",
[80].u.CallDeflection.Component.Invoke.PresentationAllowedToDivertedToUserPresent = 1,
[80].u.CallDeflection.Component.Invoke.PresentationAllowedToDivertedToUser = 1,
[81].Function = Fac_CallDeflection,
[81].u.CallDeflection.InvokeID = 84,
[81].u.CallDeflection.ComponentType = FacComponent_Invoke,
[81].u.CallDeflection.Component.Invoke.Deflection.Party.Type = 4,
[81].u.CallDeflection.Component.Invoke.Deflection.Party.LengthOfNumber = 4,
[81].u.CallDeflection.Component.Invoke.Deflection.Party.Number = "1803",
[81].u.CallDeflection.Component.Invoke.PresentationAllowedToDivertedToUserPresent = 1,
[81].u.CallDeflection.Component.Invoke.PresentationAllowedToDivertedToUser = 0,
[82].Function = Fac_CallDeflection,
[82].u.CallDeflection.InvokeID = 85,
[82].u.CallDeflection.ComponentType = FacComponent_Invoke,
[82].u.CallDeflection.Component.Invoke.Deflection.Party.Type = 4,
[82].u.CallDeflection.Component.Invoke.Deflection.Party.LengthOfNumber = 4,
[82].u.CallDeflection.Component.Invoke.Deflection.Party.Number = "1803",
[83].Function = Fac_CallDeflection,
[83].u.CallDeflection.InvokeID = 86,
[83].u.CallDeflection.ComponentType = FacComponent_Result,
[84].Function = Fac_CallRerouteing,
[84].u.CallRerouteing.InvokeID = 87,
[84].u.CallRerouteing.ComponentType = FacComponent_Invoke,
[84].u.CallRerouteing.Component.Invoke.ReroutingReason = 3,
[84].u.CallRerouteing.Component.Invoke.ReroutingCounter = 2,
[84].u.CallRerouteing.Component.Invoke.CalledAddress.Party.Type = 4,
[84].u.CallRerouteing.Component.Invoke.CalledAddress.Party.LengthOfNumber = 4,
[84].u.CallRerouteing.Component.Invoke.CalledAddress.Party.Number = "1803",
[84].u.CallRerouteing.Component.Invoke.Q931ie.Bc.Length = 2,
[84].u.CallRerouteing.Component.Invoke.Q931ie.Bc.Contents = "RT",
[84].u.CallRerouteing.Component.Invoke.Q931ie.Hlc.Length = 3,
[84].u.CallRerouteing.Component.Invoke.Q931ie.Hlc.Contents = "RTG",
[84].u.CallRerouteing.Component.Invoke.Q931ie.Llc.Length = 2,
[84].u.CallRerouteing.Component.Invoke.Q931ie.Llc.Contents = "MY",
[84].u.CallRerouteing.Component.Invoke.Q931ie.UserInfo.Length = 5,
[84].u.CallRerouteing.Component.Invoke.Q931ie.UserInfo.Contents = "YEHAW",
[84].u.CallRerouteing.Component.Invoke.LastRerouting.Type = 1,
[84].u.CallRerouteing.Component.Invoke.SubscriptionOption = 2,
[84].u.CallRerouteing.Component.Invoke.CallingPartySubaddress.Type = 1,
[84].u.CallRerouteing.Component.Invoke.CallingPartySubaddress.Length = 4,
[84].u.CallRerouteing.Component.Invoke.CallingPartySubaddress.u.Nsap = "6492",
[85].Function = Fac_CallRerouteing,
[85].u.CallRerouteing.InvokeID = 88,
[85].u.CallRerouteing.ComponentType = FacComponent_Invoke,
[85].u.CallRerouteing.Component.Invoke.ReroutingReason = 3,
[85].u.CallRerouteing.Component.Invoke.ReroutingCounter = 2,
[85].u.CallRerouteing.Component.Invoke.CalledAddress.Party.Type = 4,
[85].u.CallRerouteing.Component.Invoke.CalledAddress.Party.LengthOfNumber = 4,
[85].u.CallRerouteing.Component.Invoke.CalledAddress.Party.Number = "1803",
[85].u.CallRerouteing.Component.Invoke.Q931ie.Bc.Length = 2,
[85].u.CallRerouteing.Component.Invoke.Q931ie.Bc.Contents = "RT",
[85].u.CallRerouteing.Component.Invoke.LastRerouting.Type = 1,
[85].u.CallRerouteing.Component.Invoke.SubscriptionOption = 2,
[86].Function = Fac_CallRerouteing,
[86].u.CallRerouteing.InvokeID = 89,
[86].u.CallRerouteing.ComponentType = FacComponent_Invoke,
[86].u.CallRerouteing.Component.Invoke.ReroutingReason = 3,
[86].u.CallRerouteing.Component.Invoke.ReroutingCounter = 2,
[86].u.CallRerouteing.Component.Invoke.CalledAddress.Party.Type = 4,
[86].u.CallRerouteing.Component.Invoke.CalledAddress.Party.LengthOfNumber = 4,
[86].u.CallRerouteing.Component.Invoke.CalledAddress.Party.Number = "1803",
[86].u.CallRerouteing.Component.Invoke.Q931ie.Bc.Length = 2,
[86].u.CallRerouteing.Component.Invoke.Q931ie.Bc.Contents = "RT",
[86].u.CallRerouteing.Component.Invoke.LastRerouting.Type = 2,
[87].Function = Fac_CallRerouteing,
[87].u.CallRerouteing.InvokeID = 90,
[87].u.CallRerouteing.ComponentType = FacComponent_Result,
[88].Function = Fac_InterrogateServedUserNumbers,
[88].u.InterrogateServedUserNumbers.InvokeID = 91,
[88].u.InterrogateServedUserNumbers.ComponentType = FacComponent_Invoke,
[89].Function = Fac_InterrogateServedUserNumbers,
[89].u.InterrogateServedUserNumbers.InvokeID = 92,
[89].u.InterrogateServedUserNumbers.ComponentType = FacComponent_Result,
[89].u.InterrogateServedUserNumbers.Component.Result.NumRecords = 2,
[89].u.InterrogateServedUserNumbers.Component.Result.List[0].Type = 4,
[89].u.InterrogateServedUserNumbers.Component.Result.List[0].LengthOfNumber = 4,
[89].u.InterrogateServedUserNumbers.Component.Result.List[0].Number = "1803",
[89].u.InterrogateServedUserNumbers.Component.Result.List[1].Type = 4,
[89].u.InterrogateServedUserNumbers.Component.Result.List[1].LengthOfNumber = 4,
[89].u.InterrogateServedUserNumbers.Component.Result.List[1].Number = "5786",
[90].Function = Fac_DivertingLegInformation1,
[90].u.DivertingLegInformation1.InvokeID = 93,
[90].u.DivertingLegInformation1.DiversionReason = 4,
[90].u.DivertingLegInformation1.SubscriptionOption = 1,
[90].u.DivertingLegInformation1.DivertedToPresent = 1,
[90].u.DivertingLegInformation1.DivertedTo.Type = 2,
[91].Function = Fac_DivertingLegInformation1,
[91].u.DivertingLegInformation1.InvokeID = 94,
[91].u.DivertingLegInformation1.DiversionReason = 4,
[91].u.DivertingLegInformation1.SubscriptionOption = 1,
[92].Function = Fac_DivertingLegInformation2,
[92].u.DivertingLegInformation2.InvokeID = 95,
[92].u.DivertingLegInformation2.DiversionCounter = 3,
[92].u.DivertingLegInformation2.DiversionReason = 2,
[92].u.DivertingLegInformation2.DivertingPresent = 1,
[92].u.DivertingLegInformation2.Diverting.Type = 2,
[92].u.DivertingLegInformation2.OriginalCalledPresent = 1,
[92].u.DivertingLegInformation2.OriginalCalled.Type = 1,
[93].Function = Fac_DivertingLegInformation2,
[93].u.DivertingLegInformation2.InvokeID = 96,
[93].u.DivertingLegInformation2.DiversionCounter = 3,
[93].u.DivertingLegInformation2.DiversionReason = 2,
[93].u.DivertingLegInformation2.OriginalCalledPresent = 1,
[93].u.DivertingLegInformation2.OriginalCalled.Type = 1,
[94].Function = Fac_DivertingLegInformation2,
[94].u.DivertingLegInformation2.InvokeID = 97,
[94].u.DivertingLegInformation2.DiversionCounter = 1,
[94].u.DivertingLegInformation2.DiversionReason = 2,
[95].Function = Fac_DivertingLegInformation3,
[95].u.DivertingLegInformation3.InvokeID = 98,
[95].u.DivertingLegInformation3.PresentationAllowedIndicator = 1,
/* *INDENT-ON* */
};
#endif /* defined(AST_MISDN_ENHANCEMENTS) && defined(CCBS_TEST_MESSAGES) */
static char *handle_cli_misdn_send_facility(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
const char *channame;
const char *nr;
struct chan_list *tmp;
int port;
const char *served_nr;
struct misdn_bchannel dummy, *bc=&dummy;
unsigned max_len;
switch (cmd) {
case CLI_INIT:
e->command = "misdn send facility";
e->usage = "Usage: misdn send facility <type> <channel|port> \"<args>\" \n"
"\t type is one of:\n"
"\t - calldeflect\n"
#if defined(AST_MISDN_ENHANCEMENTS)
"\t - callrerouting\n"
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
"\t - CFActivate\n"
"\t - CFDeactivate\n";
return NULL;
case CLI_GENERATE:
return complete_ch(a);
}
if (a->argc < 5) {
return CLI_SHOWUSAGE;
}
if (strstr(a->argv[3], "calldeflect")) {
if (a->argc < 6) {
ast_verbose("calldeflect requires 1 arg: ToNumber\n\n");
return 0;
}
channame = a->argv[4];
nr = a->argv[5];
ast_verbose("Sending Calldeflection (%s) to %s\n", nr, channame);
tmp = get_chan_by_ast_name(channame);
if (!tmp) {
ast_verbose("Sending CD with nr %s to %s failed: Channel does not exist.\n", nr, channame);
return 0;
}
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
ao2_lock(tmp);
#if defined(AST_MISDN_ENHANCEMENTS)
max_len = sizeof(tmp->bc->fac_out.u.CallDeflection.Component.Invoke.Deflection.Party.Number) - 1;
if (max_len < strlen(nr)) {
ast_verbose("Sending CD with nr %s to %s failed: Number too long (up to %u digits are allowed).\n",
nr, channame, max_len);
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
ao2_unlock(tmp);
chan_list_unref(tmp, "Number too long");
return 0;
}
tmp->bc->fac_out.Function = Fac_CallDeflection;
tmp->bc->fac_out.u.CallDeflection.InvokeID = ++misdn_invoke_id;
tmp->bc->fac_out.u.CallDeflection.ComponentType = FacComponent_Invoke;
tmp->bc->fac_out.u.CallDeflection.Component.Invoke.PresentationAllowedToDivertedToUserPresent = 1;
tmp->bc->fac_out.u.CallDeflection.Component.Invoke.PresentationAllowedToDivertedToUser = 0;
tmp->bc->fac_out.u.CallDeflection.Component.Invoke.Deflection.Party.Type = 0;/* unknown */
tmp->bc->fac_out.u.CallDeflection.Component.Invoke.Deflection.Party.LengthOfNumber = strlen(nr);
strcpy((char *) tmp->bc->fac_out.u.CallDeflection.Component.Invoke.Deflection.Party.Number, nr);
tmp->bc->fac_out.u.CallDeflection.Component.Invoke.Deflection.Subaddress.Length = 0;
#else /* !defined(AST_MISDN_ENHANCEMENTS) */
max_len = sizeof(tmp->bc->fac_out.u.CDeflection.DeflectedToNumber) - 1;
if (max_len < strlen(nr)) {
ast_verbose("Sending CD with nr %s to %s failed: Number too long (up to %u digits are allowed).\n",
nr, channame, max_len);
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
ao2_unlock(tmp);
chan_list_unref(tmp, "Number too long");
return 0;
}
tmp->bc->fac_out.Function = Fac_CD;
tmp->bc->fac_out.u.CDeflection.PresentationAllowed = 0;
//tmp->bc->fac_out.u.CDeflection.DeflectedToSubaddress[0] = 0;
strcpy((char *) tmp->bc->fac_out.u.CDeflection.DeflectedToNumber, nr);
#endif /* !defined(AST_MISDN_ENHANCEMENTS) */
/* Send message */
print_facility(&tmp->bc->fac_out, tmp->bc);
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
ao2_unlock(tmp);
misdn_lib_send_event(tmp->bc, EVENT_FACILITY);
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
chan_list_unref(tmp, "Send facility complete");
#if defined(AST_MISDN_ENHANCEMENTS)
} else if (strstr(a->argv[3], "callrerouteing") || strstr(a->argv[3], "callrerouting")) {
if (a->argc < 6) {
ast_verbose("callrerouting requires 1 arg: ToNumber\n\n");
return 0;
}
channame = a->argv[4];
nr = a->argv[5];
ast_verbose("Sending Callrerouting (%s) to %s\n", nr, channame);
tmp = get_chan_by_ast_name(channame);
if (!tmp) {
ast_verbose("Sending Call Rerouting with nr %s to %s failed: Channel does not exist.\n", nr, channame);
return 0;
}
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
ao2_lock(tmp);
max_len = sizeof(tmp->bc->fac_out.u.CallRerouteing.Component.Invoke.CalledAddress.Party.Number) - 1;
if (max_len < strlen(nr)) {
ast_verbose("Sending Call Rerouting with nr %s to %s failed: Number too long (up to %u digits are allowed).\n",
nr, channame, max_len);
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
ao2_unlock(tmp);
chan_list_unref(tmp, "Number too long");
return 0;
}
tmp->bc->fac_out.Function = Fac_CallRerouteing;
tmp->bc->fac_out.u.CallRerouteing.InvokeID = ++misdn_invoke_id;
tmp->bc->fac_out.u.CallRerouteing.ComponentType = FacComponent_Invoke;
tmp->bc->fac_out.u.CallRerouteing.Component.Invoke.ReroutingReason = 0;/* unknown */
tmp->bc->fac_out.u.CallRerouteing.Component.Invoke.ReroutingCounter = 1;
tmp->bc->fac_out.u.CallRerouteing.Component.Invoke.CalledAddress.Party.Type = 0;/* unknown */
tmp->bc->fac_out.u.CallRerouteing.Component.Invoke.CalledAddress.Party.LengthOfNumber = strlen(nr);
strcpy((char *) tmp->bc->fac_out.u.CallRerouteing.Component.Invoke.CalledAddress.Party.Number, nr);
tmp->bc->fac_out.u.CallRerouteing.Component.Invoke.CalledAddress.Subaddress.Length = 0;
tmp->bc->fac_out.u.CallRerouteing.Component.Invoke.CallingPartySubaddress.Length = 0;
/* 0x90 0x90 0xa3 3.1 kHz audio, circuit mode, 64kbit/sec, level1/a-Law */
tmp->bc->fac_out.u.CallRerouteing.Component.Invoke.Q931ie.Bc.Length = 3;
tmp->bc->fac_out.u.CallRerouteing.Component.Invoke.Q931ie.Bc.Contents[0] = 0x90;
tmp->bc->fac_out.u.CallRerouteing.Component.Invoke.Q931ie.Bc.Contents[1] = 0x90;
tmp->bc->fac_out.u.CallRerouteing.Component.Invoke.Q931ie.Bc.Contents[2] = 0xa3;
tmp->bc->fac_out.u.CallRerouteing.Component.Invoke.Q931ie.Hlc.Length = 0;
tmp->bc->fac_out.u.CallRerouteing.Component.Invoke.Q931ie.Llc.Length = 0;
tmp->bc->fac_out.u.CallRerouteing.Component.Invoke.Q931ie.UserInfo.Length = 0;
tmp->bc->fac_out.u.CallRerouteing.Component.Invoke.LastRerouting.Type = 1;/* presentationRestricted */
tmp->bc->fac_out.u.CallRerouteing.Component.Invoke.SubscriptionOption = 0;/* no notification to caller */
/* Send message */
print_facility(&tmp->bc->fac_out, tmp->bc);
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
ao2_unlock(tmp);
misdn_lib_send_event(tmp->bc, EVENT_FACILITY);
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
chan_list_unref(tmp, "Send facility complete");
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
} else if (strstr(a->argv[3], "CFActivate")) {
if (a->argc < 7) {
ast_verbose("CFActivate requires 2 args: 1.FromNumber, 2.ToNumber\n\n");
return 0;
}
port = atoi(a->argv[4]);
served_nr = a->argv[5];
nr = a->argv[6];
misdn_make_dummy(bc, port, 0, misdn_lib_port_is_nt(port), 0);
ast_verbose("Sending CFActivate Port:(%d) FromNr. (%s) to Nr. (%s)\n", port, served_nr, nr);
#if defined(AST_MISDN_ENHANCEMENTS)
bc->fac_out.Function = Fac_ActivationDiversion;
bc->fac_out.u.ActivationDiversion.InvokeID = ++misdn_invoke_id;
bc->fac_out.u.ActivationDiversion.ComponentType = FacComponent_Invoke;
bc->fac_out.u.ActivationDiversion.Component.Invoke.BasicService = 0;/* allServices */
bc->fac_out.u.ActivationDiversion.Component.Invoke.Procedure = 0;/* cfu (Call Forward Unconditional) */
ast_copy_string((char *) bc->fac_out.u.ActivationDiversion.Component.Invoke.ServedUser.Number,
served_nr, sizeof(bc->fac_out.u.ActivationDiversion.Component.Invoke.ServedUser.Number));
bc->fac_out.u.ActivationDiversion.Component.Invoke.ServedUser.LengthOfNumber =
strlen((char *) bc->fac_out.u.ActivationDiversion.Component.Invoke.ServedUser.Number);
bc->fac_out.u.ActivationDiversion.Component.Invoke.ServedUser.Type = 0;/* unknown */
ast_copy_string((char *) bc->fac_out.u.ActivationDiversion.Component.Invoke.ForwardedTo.Party.Number,
nr, sizeof(bc->fac_out.u.ActivationDiversion.Component.Invoke.ForwardedTo.Party.Number));
bc->fac_out.u.ActivationDiversion.Component.Invoke.ForwardedTo.Party.LengthOfNumber =
strlen((char *) bc->fac_out.u.ActivationDiversion.Component.Invoke.ForwardedTo.Party.Number);
bc->fac_out.u.ActivationDiversion.Component.Invoke.ForwardedTo.Party.Type = 0;/* unknown */
bc->fac_out.u.ActivationDiversion.Component.Invoke.ForwardedTo.Subaddress.Length = 0;
#else /* !defined(AST_MISDN_ENHANCEMENTS) */
bc->fac_out.Function = Fac_CFActivate;
bc->fac_out.u.CFActivate.BasicService = 0; /* All Services */
bc->fac_out.u.CFActivate.Procedure = 0; /* Unconditional */
ast_copy_string((char *) bc->fac_out.u.CFActivate.ServedUserNumber, served_nr, sizeof(bc->fac_out.u.CFActivate.ServedUserNumber));
ast_copy_string((char *) bc->fac_out.u.CFActivate.ForwardedToNumber, nr, sizeof(bc->fac_out.u.CFActivate.ForwardedToNumber));
#endif /* !defined(AST_MISDN_ENHANCEMENTS) */
/* Send message */
print_facility(&bc->fac_out, bc);
misdn_lib_send_event(bc, EVENT_FACILITY);
} else if (strstr(a->argv[3], "CFDeactivate")) {
if (a->argc < 6) {
ast_verbose("CFDeactivate requires 1 arg: FromNumber\n\n");
return 0;
}
port = atoi(a->argv[4]);
served_nr = a->argv[5];
misdn_make_dummy(bc, port, 0, misdn_lib_port_is_nt(port), 0);
ast_verbose("Sending CFDeactivate Port:(%d) FromNr. (%s)\n", port, served_nr);
#if defined(AST_MISDN_ENHANCEMENTS)
bc->fac_out.Function = Fac_DeactivationDiversion;
bc->fac_out.u.DeactivationDiversion.InvokeID = ++misdn_invoke_id;
bc->fac_out.u.DeactivationDiversion.ComponentType = FacComponent_Invoke;
bc->fac_out.u.DeactivationDiversion.Component.Invoke.BasicService = 0;/* allServices */
bc->fac_out.u.DeactivationDiversion.Component.Invoke.Procedure = 0;/* cfu (Call Forward Unconditional) */
ast_copy_string((char *) bc->fac_out.u.DeactivationDiversion.Component.Invoke.ServedUser.Number,
served_nr, sizeof(bc->fac_out.u.DeactivationDiversion.Component.Invoke.ServedUser.Number));
bc->fac_out.u.DeactivationDiversion.Component.Invoke.ServedUser.LengthOfNumber =
strlen((char *) bc->fac_out.u.DeactivationDiversion.Component.Invoke.ServedUser.Number);
bc->fac_out.u.DeactivationDiversion.Component.Invoke.ServedUser.Type = 0;/* unknown */
#else /* !defined(AST_MISDN_ENHANCEMENTS) */
bc->fac_out.Function = Fac_CFDeactivate;
bc->fac_out.u.CFDeactivate.BasicService = 0; /* All Services */
bc->fac_out.u.CFDeactivate.Procedure = 0; /* Unconditional */
ast_copy_string((char *) bc->fac_out.u.CFActivate.ServedUserNumber, served_nr, sizeof(bc->fac_out.u.CFActivate.ServedUserNumber));
#endif /* !defined(AST_MISDN_ENHANCEMENTS) */
/* Send message */
print_facility(&bc->fac_out, bc);
misdn_lib_send_event(bc, EVENT_FACILITY);
#if defined(AST_MISDN_ENHANCEMENTS) && defined(CCBS_TEST_MESSAGES)
} else if (strstr(a->argv[3], "test")) {
int msg_number;
if (a->argc < 5) {
ast_verbose("test (<port> [<msg#>]) | (<channel-name> <msg#>)\n\n");
return 0;
}
port = atoi(a->argv[4]);
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
channame = a->argv[4];
tmp = get_chan_by_ast_name(channame);
if (tmp) {
/* We are going to send this FACILITY message out on an existing connection */
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
msg_number = atoi(a->argv[5]);
if (msg_number < ARRAY_LEN(Fac_Msgs)) {
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
ao2_lock(tmp);
tmp->bc->fac_out = Fac_Msgs[msg_number];
/* Send message */
print_facility(&tmp->bc->fac_out, tmp->bc);
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
ao2_unlock(tmp);
misdn_lib_send_event(tmp->bc, EVENT_FACILITY);
} else {
ast_verbose("test <channel-name> <msg#>\n\n");
}
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
chan_list_unref(tmp, "Facility test done");
} else if (a->argc < 6) {
for (msg_number = 0; msg_number < ARRAY_LEN(Fac_Msgs); ++msg_number) {
misdn_make_dummy(bc, port, 0, misdn_lib_port_is_nt(port), 0);
bc->fac_out = Fac_Msgs[msg_number];
/* Send message */
print_facility(&bc->fac_out, bc);
misdn_lib_send_event(bc, EVENT_FACILITY);
sleep(1);
}
} else {
msg_number = atoi(a->argv[5]);
if (msg_number < ARRAY_LEN(Fac_Msgs)) {
misdn_make_dummy(bc, port, 0, misdn_lib_port_is_nt(port), 0);
bc->fac_out = Fac_Msgs[msg_number];
/* Send message */
print_facility(&bc->fac_out, bc);
misdn_lib_send_event(bc, EVENT_FACILITY);
} else {
ast_verbose("test <port> [<msg#>]\n\n");
}
}
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
} else if (strstr(a->argv[3], "register")) {
if (a->argc < 5) {
ast_verbose("register <port>\n\n");
return 0;
}
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
port = atoi(a->argv[4]);
bc = misdn_lib_get_register_bc(port);
if (!bc) {
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
ast_verbose("Could not allocate REGISTER bc struct\n\n");
return 0;
}
bc->fac_out = Fac_Msgs[45];
/* Send message */
print_facility(&bc->fac_out, bc);
misdn_lib_send_event(bc, EVENT_REGISTER);
#endif /* defined(AST_MISDN_ENHANCEMENTS) && defined(CCBS_TEST_MESSAGES) */
}
return CLI_SUCCESS;
}
static char *handle_cli_misdn_send_restart(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
int port;
int channel;
switch (cmd) {
case CLI_INIT:
e->command = "misdn send restart";
e->usage =
"Usage: misdn send restart [port [channel]]\n"
" Send a restart for every bchannel on the given port.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc < 4 || a->argc > 5) {
return CLI_SHOWUSAGE;
}
port = atoi(a->argv[3]);
if (a->argc == 5) {
channel = atoi(a->argv[4]);
misdn_lib_send_restart(port, channel);
} else {
misdn_lib_send_restart(port, -1);
}
return CLI_SUCCESS;
}
static char *handle_cli_misdn_send_digit(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
const char *channame;
const char *msg;
struct chan_list *tmp;
int i, msglen;
switch (cmd) {
case CLI_INIT:
e->command = "misdn send digit";
e->usage =
"Usage: misdn send digit <channel> \"<msg>\" \n"
" Send <digit> to <channel> as DTMF Tone\n"
" when channel is a mISDN channel\n";
return NULL;
case CLI_GENERATE:
return complete_ch(a);
}
if (a->argc != 5) {
return CLI_SHOWUSAGE;
}
channame = a->argv[3];
msg = a->argv[4];
msglen = strlen(msg);
ast_cli(a->fd, "Sending %s to %s\n", msg, channame);
tmp = get_chan_by_ast_name(channame);
if (!tmp) {
ast_cli(a->fd, "Sending %s to %s failed Channel does not exist\n", msg, channame);
return CLI_SUCCESS;
}
#if 1
for (i = 0; i < msglen; i++) {
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
if (!tmp->ast) {
break;
}
ast_cli(a->fd, "Sending: %c\n", msg[i]);
send_digit_to_chan(tmp, msg[i]);
/* res = ast_safe_sleep(tmp->ast, 250); */
usleep(250000);
/* res = ast_waitfor(tmp->ast,100); */
}
#else
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
if (tmp->ast) {
ast_dtmf_stream(tmp->ast, NULL, msg, 250);
}
#endif
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
chan_list_unref(tmp, "Digit(s) sent");
return CLI_SUCCESS;
}
static char *handle_cli_misdn_toggle_echocancel(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
const char *channame;
struct chan_list *tmp;
switch (cmd) {
case CLI_INIT:
e->command = "misdn toggle echocancel";
e->usage =
"Usage: misdn toggle echocancel <channel>\n"
" Toggle EchoCancel on mISDN Channel.\n";
return NULL;
case CLI_GENERATE:
return complete_ch(a);
}
if (a->argc != 4) {
return CLI_SHOWUSAGE;
}
channame = a->argv[3];
ast_cli(a->fd, "Toggling EchoCancel on %s\n", channame);
tmp = get_chan_by_ast_name(channame);
if (!tmp) {
ast_cli(a->fd, "Toggling EchoCancel %s failed Channel does not exist\n", channame);
return CLI_SUCCESS;
}
tmp->toggle_ec = tmp->toggle_ec ? 0 : 1;
if (tmp->toggle_ec) {
#ifdef MISDN_1_2
update_pipeline_config(tmp->bc);
#else
update_ec_config(tmp->bc);
#endif
manager_ec_enable(tmp->bc);
} else {
manager_ec_disable(tmp->bc);
}
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
chan_list_unref(tmp, "Done toggling echo cancel");
return CLI_SUCCESS;
}
static char *handle_cli_misdn_send_display(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
const char *channame;
const char *msg;
struct chan_list *tmp;
switch (cmd) {
case CLI_INIT:
e->command = "misdn send display";
e->usage =
"Usage: misdn send display <channel> \"<msg>\" \n"
" Send <msg> to <channel> as Display Message\n"
" when channel is a mISDN channel\n";
return NULL;
case CLI_GENERATE:
return complete_ch(a);
}
if (a->argc != 5) {
return CLI_SHOWUSAGE;
}
channame = a->argv[3];
msg = a->argv[4];
ast_cli(a->fd, "Sending %s to %s\n", msg, channame);
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
tmp = get_chan_by_ast_name(channame);
if (tmp && tmp->bc) {
ast_copy_string(tmp->bc->display, msg, sizeof(tmp->bc->display));
misdn_lib_send_event(tmp->bc, EVENT_INFORMATION);
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
chan_list_unref(tmp, "Done sending display");
} else {
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
if (tmp) {
chan_list_unref(tmp, "Display failed");
}
ast_cli(a->fd, "No such channel %s\n", channame);
return CLI_SUCCESS;
}
return CLI_SUCCESS;
}
static char *complete_ch(struct ast_cli_args *a)
{
return ast_complete_channels(a->line, a->word, a->pos, a->n, 3);
}
static char *complete_debug_port(struct ast_cli_args *a)
{
if (a->n) {
return NULL;
}
switch (a->pos) {
case 4:
if (a->word[0] == 'p') {
return ast_strdup("port");
} else if (a->word[0] == 'o') {
return ast_strdup("only");
}
break;
case 6:
if (a->word[0] == 'o') {
return ast_strdup("only");
}
break;
}
return NULL;
}
static char *complete_show_config(struct ast_cli_args *a)
{
char buffer[BUFFERSIZE];
enum misdn_cfg_elements elem;
int wordlen = strlen(a->word);
int which = 0;
int port = 0;
switch (a->pos) {
case 3:
if ((!strncmp(a->word, "description", wordlen)) && (++which > a->n)) {
return ast_strdup("description");
}
if ((!strncmp(a->word, "descriptions", wordlen)) && (++which > a->n)) {
return ast_strdup("descriptions");
}
if ((!strncmp(a->word, "0", wordlen)) && (++which > a->n)) {
return ast_strdup("0");
}
while ((port = misdn_cfg_get_next_port(port)) != -1) {
snprintf(buffer, sizeof(buffer), "%d", port);
if ((!strncmp(a->word, buffer, wordlen)) && (++which > a->n)) {
return ast_strdup(buffer);
}
}
break;
case 4:
if (strstr(a->line, "description ")) {
for (elem = MISDN_CFG_FIRST + 1; elem < MISDN_GEN_LAST; ++elem) {
if ((elem == MISDN_CFG_LAST) || (elem == MISDN_GEN_FIRST)) {
continue;
}
misdn_cfg_get_name(elem, buffer, sizeof(buffer));
if (!wordlen || !strncmp(a->word, buffer, wordlen)) {
if (++which > a->n) {
return ast_strdup(buffer);
}
}
}
} else if (strstr(a->line, "descriptions ")) {
if ((!wordlen || !strncmp(a->word, "general", wordlen)) && (++which > a->n)) {
return ast_strdup("general");
}
if ((!wordlen || !strncmp(a->word, "ports", wordlen)) && (++which > a->n)) {
return ast_strdup("ports");
}
}
break;
}
return NULL;
}
static struct ast_cli_entry chan_misdn_clis[] = {
/* *INDENT-OFF* */
AST_CLI_DEFINE(handle_cli_misdn_port_block, "Block the given port"),
AST_CLI_DEFINE(handle_cli_misdn_port_down, "Try to deactivate the L1 on the given port"),
AST_CLI_DEFINE(handle_cli_misdn_port_unblock, "Unblock the given port"),
AST_CLI_DEFINE(handle_cli_misdn_port_up, "Try to establish L1 on the given port"),
AST_CLI_DEFINE(handle_cli_misdn_reload, "Reload internal mISDN config, read from the config file"),
AST_CLI_DEFINE(handle_cli_misdn_restart_pid, "Restart the given pid"),
AST_CLI_DEFINE(handle_cli_misdn_restart_port, "Restart the given port"),
AST_CLI_DEFINE(handle_cli_misdn_show_channel, "Show an internal mISDN channel"),
AST_CLI_DEFINE(handle_cli_misdn_show_channels, "Show the internal mISDN channel list"),
AST_CLI_DEFINE(handle_cli_misdn_show_config, "Show internal mISDN config, read from the config file"),
AST_CLI_DEFINE(handle_cli_misdn_show_port, "Show detailed information for given port"),
AST_CLI_DEFINE(handle_cli_misdn_show_ports_stats, "Show mISDNs channel's call statistics per port"),
AST_CLI_DEFINE(handle_cli_misdn_show_stacks, "Show internal mISDN stack_list"),
AST_CLI_DEFINE(handle_cli_misdn_send_facility, "Sends a Facility Message to the mISDN Channel"),
AST_CLI_DEFINE(handle_cli_misdn_send_digit, "Send DTMF digit to mISDN Channel"),
AST_CLI_DEFINE(handle_cli_misdn_send_display, "Send Text to mISDN Channel"),
AST_CLI_DEFINE(handle_cli_misdn_send_restart, "Send a restart for every bchannel on the given port"),
AST_CLI_DEFINE(handle_cli_misdn_set_crypt_debug, "Set CryptDebuglevel of chan_misdn, at the moment, level={1,2}"),
AST_CLI_DEFINE(handle_cli_misdn_set_debug, "Set Debuglevel of chan_misdn"),
AST_CLI_DEFINE(handle_cli_misdn_set_tics, "???"),
AST_CLI_DEFINE(handle_cli_misdn_toggle_echocancel, "Toggle EchoCancel on mISDN Channel"),
/* *INDENT-ON* */
};
/*! \brief Updates caller ID information from config */
static void update_config(struct chan_list *ch)
{
struct ast_channel *ast;
struct misdn_bchannel *bc;
int port;
int hdlc = 0;
int pres;
int screen;
if (!ch) {
ast_log(LOG_WARNING, "Cannot configure without chanlist\n");
return;
}
ast = ch->ast;
bc = ch->bc;
if (! ast || ! bc) {
ast_log(LOG_WARNING, "Cannot configure without ast || bc\n");
return;
}
port = bc->port;
chan_misdn_log(7, port, "update_config: Getting Config\n");
misdn_cfg_get(port, MISDN_CFG_HDLC, &hdlc, sizeof(int));
if (hdlc) {
switch (bc->capability) {
case INFO_CAPABILITY_DIGITAL_UNRESTRICTED:
case INFO_CAPABILITY_DIGITAL_RESTRICTED:
chan_misdn_log(1, bc->port, " --> CONF HDLC\n");
bc->hdlc = 1;
break;
}
}
misdn_cfg_get(port, MISDN_CFG_PRES, &pres, sizeof(pres));
misdn_cfg_get(port, MISDN_CFG_SCREEN, &screen, sizeof(screen));
chan_misdn_log(2, port, " --> pres: %d screen: %d\n", pres, screen);
if (pres < 0 || screen < 0) {
chan_misdn_log(2, port, " --> pres: %x\n", ast_channel_connected(ast)->id.number.presentation);
bc->caller.presentation = ast_to_misdn_pres(ast_channel_connected(ast)->id.number.presentation);
chan_misdn_log(2, port, " --> PRES: %s(%d)\n", misdn_to_str_pres(bc->caller.presentation), bc->caller.presentation);
bc->caller.screening = ast_to_misdn_screen(ast_channel_connected(ast)->id.number.presentation);
chan_misdn_log(2, port, " --> SCREEN: %s(%d)\n", misdn_to_str_screen(bc->caller.screening), bc->caller.screening);
} else {
bc->caller.screening = screen;
bc->caller.presentation = pres;
}
}
static void config_jitterbuffer(struct chan_list *ch)
{
struct misdn_bchannel *bc = ch->bc;
int len = ch->jb_len;
int threshold = ch->jb_upper_threshold;
chan_misdn_log(5, bc->port, "config_jb: Called\n");
if (!len) {
chan_misdn_log(1, bc->port, "config_jb: Deactivating Jitterbuffer\n");
bc->nojitter = 1;
} else {
if (len <= 100 || len > 8000) {
chan_misdn_log(0, bc->port, "config_jb: Jitterbuffer out of Bounds, setting to 1000\n");
len = 1000;
}
if (threshold > len) {
chan_misdn_log(0, bc->port, "config_jb: Jitterbuffer Threshold > Jitterbuffer setting to Jitterbuffer -1\n");
}
if (ch->jb) {
cb_log(0, bc->port, "config_jb: We've got a Jitterbuffer Already on this port.\n");
misdn_jb_destroy(ch->jb);
ch->jb = NULL;
}
ch->jb = misdn_jb_init(len, threshold);
if (!ch->jb) {
bc->nojitter = 1;
}
}
}
void debug_numtype(int port, int numtype, char *type)
{
switch (numtype) {
case NUMTYPE_UNKNOWN:
chan_misdn_log(2, port, " --> %s: Unknown\n", type);
break;
case NUMTYPE_INTERNATIONAL:
chan_misdn_log(2, port, " --> %s: International\n", type);
break;
case NUMTYPE_NATIONAL:
chan_misdn_log(2, port, " --> %s: National\n", type);
break;
case NUMTYPE_NETWORK_SPECIFIC:
chan_misdn_log(2, port, " --> %s: Network Specific\n", type);
break;
case NUMTYPE_SUBSCRIBER:
chan_misdn_log(2, port, " --> %s: Subscriber\n", type);
break;
case NUMTYPE_ABBREVIATED:
chan_misdn_log(2, port, " --> %s: Abbreviated\n", type);
break;
/* Maybe we should cut off the prefix if present ? */
default:
chan_misdn_log(0, port, " --> !!!! Wrong dialplan setting, please see the misdn.conf sample file\n ");
break;
}
}
#ifdef MISDN_1_2
static int update_pipeline_config(struct misdn_bchannel *bc)
{
int ec;
misdn_cfg_get(bc->port, MISDN_CFG_PIPELINE, bc->pipeline, sizeof(bc->pipeline));
if (*bc->pipeline) {
return 0;
}
misdn_cfg_get(bc->port, MISDN_CFG_ECHOCANCEL, &ec, sizeof(ec));
if (ec == 1) {
ast_copy_string(bc->pipeline, "mg2ec", sizeof(bc->pipeline));
} else if (ec > 1) {
snprintf(bc->pipeline, sizeof(bc->pipeline), "mg2ec(deftaps=%d)", ec);
}
return 0;
}
#else
static int update_ec_config(struct misdn_bchannel *bc)
{
int ec;
int port = bc->port;
misdn_cfg_get(port, MISDN_CFG_ECHOCANCEL, &ec, sizeof(ec));
if (ec == 1) {
bc->ec_enable = 1;
} else if (ec > 1) {
bc->ec_enable = 1;
bc->ec_deftaps = ec;
}
return 0;
}
#endif
static int read_config(struct chan_list *ch)
{
struct ast_channel *ast;
struct misdn_bchannel *bc;
int port;
int hdlc = 0;
char lang[BUFFERSIZE + 1];
char faxdetect[BUFFERSIZE + 1];
char buf[256];
char buf2[256];
ast_group_t pg;
ast_group_t cg;
struct ast_namedgroups *npg;
struct ast_namedgroups *ncg;
struct ast_str *tmp_str;
if (!ch) {
ast_log(LOG_WARNING, "Cannot configure without chanlist\n");
return -1;
}
ast = ch->ast;
bc = ch->bc;
if (! ast || ! bc) {
ast_log(LOG_WARNING, "Cannot configure without ast || bc\n");
return -1;
}
port = bc->port;
chan_misdn_log(1, port, "read_config: Getting Config\n");
misdn_cfg_get(port, MISDN_CFG_LANGUAGE, lang, sizeof(lang));
ast_channel_lock(ast);
ast_channel_language_set(ast, lang);
ast_channel_unlock(ast);
misdn_cfg_get(port, MISDN_CFG_MUSICCLASS, ch->mohinterpret, sizeof(ch->mohinterpret));
misdn_cfg_get(port, MISDN_CFG_TXGAIN, &bc->txgain, sizeof(bc->txgain));
misdn_cfg_get(port, MISDN_CFG_RXGAIN, &bc->rxgain, sizeof(bc->rxgain));
misdn_cfg_get(port, MISDN_CFG_INCOMING_EARLY_AUDIO, &ch->incoming_early_audio, sizeof(ch->incoming_early_audio));
misdn_cfg_get(port, MISDN_CFG_SENDDTMF, &bc->send_dtmf, sizeof(bc->send_dtmf));
misdn_cfg_get(port, MISDN_CFG_ASTDTMF, &ch->ast_dsp, sizeof(int));
if (ch->ast_dsp) {
ch->ignore_dtmf = 1;
}
misdn_cfg_get(port, MISDN_CFG_NEED_MORE_INFOS, &bc->need_more_infos, sizeof(bc->need_more_infos));
misdn_cfg_get(port, MISDN_CFG_NTTIMEOUT, &ch->nttimeout, sizeof(ch->nttimeout));
misdn_cfg_get(port, MISDN_CFG_NOAUTORESPOND_ON_SETUP, &ch->noautorespond_on_setup, sizeof(ch->noautorespond_on_setup));
misdn_cfg_get(port, MISDN_CFG_FAR_ALERTING, &ch->far_alerting, sizeof(ch->far_alerting));
misdn_cfg_get(port, MISDN_CFG_ALLOWED_BEARERS, &ch->allowed_bearers, sizeof(ch->allowed_bearers));
misdn_cfg_get(port, MISDN_CFG_FAXDETECT, faxdetect, sizeof(faxdetect));
misdn_cfg_get(port, MISDN_CFG_HDLC, &hdlc, sizeof(hdlc));
if (hdlc) {
switch (bc->capability) {
case INFO_CAPABILITY_DIGITAL_UNRESTRICTED:
case INFO_CAPABILITY_DIGITAL_RESTRICTED:
chan_misdn_log(1, bc->port, " --> CONF HDLC\n");
bc->hdlc = 1;
break;
}
}
/*Initialize new Jitterbuffer*/
misdn_cfg_get(port, MISDN_CFG_JITTERBUFFER, &ch->jb_len, sizeof(ch->jb_len));
misdn_cfg_get(port, MISDN_CFG_JITTERBUFFER_UPPER_THRESHOLD, &ch->jb_upper_threshold, sizeof(ch->jb_upper_threshold));
config_jitterbuffer(ch);
misdn_cfg_get(bc->port, MISDN_CFG_CONTEXT, ch->context, sizeof(ch->context));
ast_channel_lock(ast);
ast_channel_context_set(ast, ch->context);
ast_channel_unlock(ast);
#ifdef MISDN_1_2
update_pipeline_config(bc);
#else
update_ec_config(bc);
#endif
misdn_cfg_get(bc->port, MISDN_CFG_EARLY_BCONNECT, &bc->early_bconnect, sizeof(bc->early_bconnect));
misdn_cfg_get(port, MISDN_CFG_DISPLAY_CONNECTED, &bc->display_connected, sizeof(bc->display_connected));
misdn_cfg_get(port, MISDN_CFG_DISPLAY_SETUP, &bc->display_setup, sizeof(bc->display_setup));
misdn_cfg_get(port, MISDN_CFG_OUTGOING_COLP, &bc->outgoing_colp, sizeof(bc->outgoing_colp));
misdn_cfg_get(port, MISDN_CFG_PICKUPGROUP, &pg, sizeof(pg));
misdn_cfg_get(port, MISDN_CFG_CALLGROUP, &cg, sizeof(cg));
chan_misdn_log(5, port, " --> * CallGrp:%s PickupGrp:%s\n", ast_print_group(buf, sizeof(buf), cg), ast_print_group(buf2, sizeof(buf2), pg));
ast_channel_lock(ast);
ast_channel_pickupgroup_set(ast, pg);
ast_channel_callgroup_set(ast, cg);
ast_channel_unlock(ast);
misdn_cfg_get(port, MISDN_CFG_NAMEDPICKUPGROUP, &npg, sizeof(npg));
misdn_cfg_get(port, MISDN_CFG_NAMEDCALLGROUP, &ncg, sizeof(ncg));
tmp_str = ast_str_create(1024);
if (tmp_str) {
chan_misdn_log(5, port, " --> * NamedCallGrp:%s\n", ast_print_namedgroups(&tmp_str, ncg));
ast_str_reset(tmp_str);
chan_misdn_log(5, port, " --> * NamedPickupGrp:%s\n", ast_print_namedgroups(&tmp_str, npg));
ast_free(tmp_str);
}
ast_channel_lock(ast);
ast_channel_named_pickupgroups_set(ast, npg);
ast_channel_named_callgroups_set(ast, ncg);
ast_channel_unlock(ast);
if (ch->originator == ORG_AST) {
char callerid[BUFFERSIZE + 1];
/* ORIGINATOR Asterisk (outgoing call) */
misdn_cfg_get(port, MISDN_CFG_TE_CHOOSE_CHANNEL, &(bc->te_choose_channel), sizeof(bc->te_choose_channel));
if (strstr(faxdetect, "outgoing") || strstr(faxdetect, "both")) {
ch->faxdetect = strstr(faxdetect, "nojump") ? 2 : 1;
}
misdn_cfg_get(port, MISDN_CFG_CALLERID, callerid, sizeof(callerid));
if (!ast_strlen_zero(callerid)) {
char *cid_name = NULL;
char *cid_num = NULL;
ast_callerid_parse(callerid, &cid_name, &cid_num);
if (cid_name) {
ast_copy_string(bc->caller.name, cid_name, sizeof(bc->caller.name));
} else {
bc->caller.name[0] = '\0';
}
if (cid_num) {
ast_copy_string(bc->caller.number, cid_num, sizeof(bc->caller.number));
} else {
bc->caller.number[0] = '\0';
}
chan_misdn_log(1, port, " --> * Setting caller to \"%s\" <%s>\n", bc->caller.name, bc->caller.number);
}
misdn_cfg_get(port, MISDN_CFG_DIALPLAN, &bc->dialed.number_type, sizeof(bc->dialed.number_type));
bc->dialed.number_plan = NUMPLAN_ISDN;
debug_numtype(port, bc->dialed.number_type, "TON");
ch->overlap_dial = 0;
} else {
/* ORIGINATOR MISDN (incoming call) */
if (strstr(faxdetect, "incoming") || strstr(faxdetect, "both")) {
ch->faxdetect = (strstr(faxdetect, "nojump")) ? 2 : 1;
}
/* Add configured prefix to caller.number */
misdn_add_number_prefix(bc->port, bc->caller.number_type, bc->caller.number, sizeof(bc->caller.number));
if (ast_strlen_zero(bc->dialed.number) && !ast_strlen_zero(bc->keypad)) {
ast_copy_string(bc->dialed.number, bc->keypad, sizeof(bc->dialed.number));
}
/* Add configured prefix to dialed.number */
misdn_add_number_prefix(bc->port, bc->dialed.number_type, bc->dialed.number, sizeof(bc->dialed.number));
ast_channel_lock(ast);
ast_channel_exten_set(ast, bc->dialed.number);
ast_channel_unlock(ast);
misdn_cfg_get(bc->port, MISDN_CFG_OVERLAP_DIAL, &ch->overlap_dial, sizeof(ch->overlap_dial));
ast_mutex_init(&ch->overlap_tv_lock);
} /* ORIG MISDN END */
Enhancements to connected line and redirecting work. From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
misdn_cfg_get(port, MISDN_CFG_INCOMING_CALLERID_TAG, bc->incoming_cid_tag, sizeof(bc->incoming_cid_tag));
if (!ast_strlen_zero(bc->incoming_cid_tag)) {
chan_misdn_log(1, port, " --> * Setting incoming caller id tag to \"%s\"\n", bc->incoming_cid_tag);
}
ch->overlap_dial_task = -1;
if (ch->faxdetect || ch->ast_dsp) {
misdn_cfg_get(port, MISDN_CFG_FAXDETECT_TIMEOUT, &ch->faxdetect_timeout, sizeof(ch->faxdetect_timeout));
if (!ch->dsp) {
ch->dsp = ast_dsp_new();
}
if (ch->dsp) {
ast_dsp_set_features(ch->dsp, DSP_FEATURE_DIGIT_DETECT | (ch->faxdetect ? DSP_FEATURE_FAX_DETECT : 0));
}
}
/* AOCD initialization */
bc->AOCDtype = Fac_None;
return 0;
}
/*!
* \internal
* \brief Send a connected line update to the other channel
*
* \param ast Current Asterisk channel
* \param id Party id information to send to the other side
* \param source Why are we sending this update
* \param cid_tag User tag to apply to the party id.
*
* \return Nothing
*/
Enhancements to connected line and redirecting work. From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
static void misdn_queue_connected_line_update(struct ast_channel *ast, const struct misdn_party_id *id, enum AST_CONNECTED_LINE_UPDATE_SOURCE source, char *cid_tag)
{
struct ast_party_connected_line connected;
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
struct ast_set_party_connected_line update_connected;
ast_party_connected_line_init(&connected);
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
memset(&update_connected, 0, sizeof(update_connected));
update_connected.id.number = 1;
connected.id.number.valid = 1;
connected.id.number.str = (char *) id->number;
connected.id.number.plan = misdn_to_ast_ton(id->number_type)
| misdn_to_ast_plan(id->number_plan);
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
connected.id.number.presentation = misdn_to_ast_pres(id->presentation)
| misdn_to_ast_screen(id->screening);
Add private representation of caller, connected and redirecting party ids. This patch adds the feature "Private representation of caller, connected and redirecting party ids", as previously discussed with us (DATUS) and Digium. 1. Feature motivation Until now it is quite difficult to modify a party number or name which can only be seen by exactly one particular instantiated technology channel subscriber. One example where a modified party number or name on one channel is spread over several channels are supplementary services like call transfer or pickup. To implement these features Asterisk internally copies caller and connected ids from one channel to another. Another example are extension subscriptions. The monitoring entities (watchers) are notified of state changes and - if desired - of party numbers or names which represent the involving call parties. One major feature where a private representation of party names is essentially needed, i.e. where a party name shall be exclusively signaled to only one particular user, is a private user-specific name resolution for party numbers. A lookup in a private destination-dependent telephone book shall provide party names which cannot be seen by any other user at any time. 2. Feature Description This feature comes along with the implementation of additional private party id elements for caller id, connected id and redirecting ids inside Asterisk channels. The private party id elements can be read or set by the user using Asterisk dialplan functions. When a technology channel is initiating a call, receives an internal connected-line update event, or receives an internal redirecting update event, it merges the corresponding public id with the private id to create an effective party id. The effective party id is then used for protocol signaling. The channel technologies which initially support the private id representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and PRI (chan_dahdi). Once a private name or number on a channel is set and (implicitly) made valid, it is generally used for any further protocol signaling until it is rewritten or invalidated. To simplify the invalidation of private ids all internally generated connected/redirecting update events and also all connected/redirecting update events which are generated by technology channels -- receiving regarding protocol information - automatically trigger the invalidation of private ids. If not using the private party id representation feature at all, i.e. if using only the 'regular' caller-id, connected and redirecting related functions, the current characteristic of Asterisk is not affected by the new extended functionality. 3. User interface Description To grant access to the private name and number representation from the Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan functions are extended by the following data types. The formats of these data types are equal to the corresponding regular 'non-private' already existing data types: CALLERID: priv-all priv-name priv-name-valid priv-name-charset priv-name-pres priv-num priv-num-valid priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE: priv-name priv-name-valid priv-name-pres priv-name-charset priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag REDIRECTING: priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd priv-orig-tag priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd priv-to-tag Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2030/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-10 19:54:55 +00:00
/*
* Make sure that any earlier private connected id
* representation at the remote end is invalidated
*/
ast_set_party_id_all(&update_connected.priv);
Enhancements to connected line and redirecting work. From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
connected.id.tag = cid_tag;
connected.source = source;
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
ast_channel_queue_connected_line_update(ast, &connected, &update_connected);
}
/*!
* \internal
* \brief Update the caller id party on this channel.
*
* \param ast Current Asterisk channel
* \param id Remote party id information to update.
* \param cid_tag User tag to apply to the party id.
*
* \return Nothing
*/
static void misdn_update_caller_id(struct ast_channel *ast, const struct misdn_party_id *id, char *cid_tag)
{
struct ast_party_caller caller;
struct ast_set_party_caller update_caller;
memset(&update_caller, 0, sizeof(update_caller));
update_caller.id.number = 1;
update_caller.ani.number = 1;
ast_channel_lock(ast);
ast_party_caller_set_init(&caller, ast_channel_caller(ast));
caller.id.number.valid = 1;
caller.id.number.str = (char *) id->number;
caller.id.number.plan = misdn_to_ast_ton(id->number_type)
| misdn_to_ast_plan(id->number_plan);
caller.id.number.presentation = misdn_to_ast_pres(id->presentation)
| misdn_to_ast_screen(id->screening);
caller.ani.number = caller.id.number;
caller.id.tag = cid_tag;
caller.ani.tag = cid_tag;
ast_channel_set_caller_event(ast, &caller, &update_caller);
ast_channel_unlock(ast);
}
/*!
* \internal
* \brief Update the remote party id information.
*
* \param ast Current Asterisk channel
* \param id Remote party id information to update.
* \param source Why are we sending this update
* \param cid_tag User tag to apply to the party id.
*
* \return Nothing
*/
static void misdn_update_remote_party(struct ast_channel *ast, const struct misdn_party_id *id, enum AST_CONNECTED_LINE_UPDATE_SOURCE source, char *cid_tag)
{
misdn_update_caller_id(ast, id, cid_tag);
misdn_queue_connected_line_update(ast, id, source, cid_tag);
}
/*!
* \internal
* \brief Get the connected line information out of the Asterisk channel.
*
* \param ast Current Asterisk channel
* \param bc Associated B channel
* \param originator Who originally created this channel. ORG_AST or ORG_MISDN
*
* \return Nothing
*/
static void misdn_get_connected_line(struct ast_channel *ast, struct misdn_bchannel *bc, int originator)
{
int number_type;
Add private representation of caller, connected and redirecting party ids. This patch adds the feature "Private representation of caller, connected and redirecting party ids", as previously discussed with us (DATUS) and Digium. 1. Feature motivation Until now it is quite difficult to modify a party number or name which can only be seen by exactly one particular instantiated technology channel subscriber. One example where a modified party number or name on one channel is spread over several channels are supplementary services like call transfer or pickup. To implement these features Asterisk internally copies caller and connected ids from one channel to another. Another example are extension subscriptions. The monitoring entities (watchers) are notified of state changes and - if desired - of party numbers or names which represent the involving call parties. One major feature where a private representation of party names is essentially needed, i.e. where a party name shall be exclusively signaled to only one particular user, is a private user-specific name resolution for party numbers. A lookup in a private destination-dependent telephone book shall provide party names which cannot be seen by any other user at any time. 2. Feature Description This feature comes along with the implementation of additional private party id elements for caller id, connected id and redirecting ids inside Asterisk channels. The private party id elements can be read or set by the user using Asterisk dialplan functions. When a technology channel is initiating a call, receives an internal connected-line update event, or receives an internal redirecting update event, it merges the corresponding public id with the private id to create an effective party id. The effective party id is then used for protocol signaling. The channel technologies which initially support the private id representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and PRI (chan_dahdi). Once a private name or number on a channel is set and (implicitly) made valid, it is generally used for any further protocol signaling until it is rewritten or invalidated. To simplify the invalidation of private ids all internally generated connected/redirecting update events and also all connected/redirecting update events which are generated by technology channels -- receiving regarding protocol information - automatically trigger the invalidation of private ids. If not using the private party id representation feature at all, i.e. if using only the 'regular' caller-id, connected and redirecting related functions, the current characteristic of Asterisk is not affected by the new extended functionality. 3. User interface Description To grant access to the private name and number representation from the Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan functions are extended by the following data types. The formats of these data types are equal to the corresponding regular 'non-private' already existing data types: CALLERID: priv-all priv-name priv-name-valid priv-name-charset priv-name-pres priv-num priv-num-valid priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE: priv-name priv-name-valid priv-name-pres priv-name-charset priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag REDIRECTING: priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd priv-orig-tag priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd priv-to-tag Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2030/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-10 19:54:55 +00:00
struct ast_party_id connected_id = ast_channel_connected_effective_id(ast);
if (originator == ORG_MISDN) {
/* ORIGINATOR MISDN (incoming call) */
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
ast_copy_string(bc->connected.name,
Add private representation of caller, connected and redirecting party ids. This patch adds the feature "Private representation of caller, connected and redirecting party ids", as previously discussed with us (DATUS) and Digium. 1. Feature motivation Until now it is quite difficult to modify a party number or name which can only be seen by exactly one particular instantiated technology channel subscriber. One example where a modified party number or name on one channel is spread over several channels are supplementary services like call transfer or pickup. To implement these features Asterisk internally copies caller and connected ids from one channel to another. Another example are extension subscriptions. The monitoring entities (watchers) are notified of state changes and - if desired - of party numbers or names which represent the involving call parties. One major feature where a private representation of party names is essentially needed, i.e. where a party name shall be exclusively signaled to only one particular user, is a private user-specific name resolution for party numbers. A lookup in a private destination-dependent telephone book shall provide party names which cannot be seen by any other user at any time. 2. Feature Description This feature comes along with the implementation of additional private party id elements for caller id, connected id and redirecting ids inside Asterisk channels. The private party id elements can be read or set by the user using Asterisk dialplan functions. When a technology channel is initiating a call, receives an internal connected-line update event, or receives an internal redirecting update event, it merges the corresponding public id with the private id to create an effective party id. The effective party id is then used for protocol signaling. The channel technologies which initially support the private id representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and PRI (chan_dahdi). Once a private name or number on a channel is set and (implicitly) made valid, it is generally used for any further protocol signaling until it is rewritten or invalidated. To simplify the invalidation of private ids all internally generated connected/redirecting update events and also all connected/redirecting update events which are generated by technology channels -- receiving regarding protocol information - automatically trigger the invalidation of private ids. If not using the private party id representation feature at all, i.e. if using only the 'regular' caller-id, connected and redirecting related functions, the current characteristic of Asterisk is not affected by the new extended functionality. 3. User interface Description To grant access to the private name and number representation from the Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan functions are extended by the following data types. The formats of these data types are equal to the corresponding regular 'non-private' already existing data types: CALLERID: priv-all priv-name priv-name-valid priv-name-charset priv-name-pres priv-num priv-num-valid priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE: priv-name priv-name-valid priv-name-pres priv-name-charset priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag REDIRECTING: priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd priv-orig-tag priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd priv-to-tag Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2030/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-10 19:54:55 +00:00
S_COR(connected_id.name.valid, connected_id.name.str, ""),
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
sizeof(bc->connected.name));
Add private representation of caller, connected and redirecting party ids. This patch adds the feature "Private representation of caller, connected and redirecting party ids", as previously discussed with us (DATUS) and Digium. 1. Feature motivation Until now it is quite difficult to modify a party number or name which can only be seen by exactly one particular instantiated technology channel subscriber. One example where a modified party number or name on one channel is spread over several channels are supplementary services like call transfer or pickup. To implement these features Asterisk internally copies caller and connected ids from one channel to another. Another example are extension subscriptions. The monitoring entities (watchers) are notified of state changes and - if desired - of party numbers or names which represent the involving call parties. One major feature where a private representation of party names is essentially needed, i.e. where a party name shall be exclusively signaled to only one particular user, is a private user-specific name resolution for party numbers. A lookup in a private destination-dependent telephone book shall provide party names which cannot be seen by any other user at any time. 2. Feature Description This feature comes along with the implementation of additional private party id elements for caller id, connected id and redirecting ids inside Asterisk channels. The private party id elements can be read or set by the user using Asterisk dialplan functions. When a technology channel is initiating a call, receives an internal connected-line update event, or receives an internal redirecting update event, it merges the corresponding public id with the private id to create an effective party id. The effective party id is then used for protocol signaling. The channel technologies which initially support the private id representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and PRI (chan_dahdi). Once a private name or number on a channel is set and (implicitly) made valid, it is generally used for any further protocol signaling until it is rewritten or invalidated. To simplify the invalidation of private ids all internally generated connected/redirecting update events and also all connected/redirecting update events which are generated by technology channels -- receiving regarding protocol information - automatically trigger the invalidation of private ids. If not using the private party id representation feature at all, i.e. if using only the 'regular' caller-id, connected and redirecting related functions, the current characteristic of Asterisk is not affected by the new extended functionality. 3. User interface Description To grant access to the private name and number representation from the Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan functions are extended by the following data types. The formats of these data types are equal to the corresponding regular 'non-private' already existing data types: CALLERID: priv-all priv-name priv-name-valid priv-name-charset priv-name-pres priv-num priv-num-valid priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE: priv-name priv-name-valid priv-name-pres priv-name-charset priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag REDIRECTING: priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd priv-orig-tag priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd priv-to-tag Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2030/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-10 19:54:55 +00:00
if (connected_id.number.valid) {
ast_copy_string(bc->connected.number, S_OR(connected_id.number.str, ""),
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
sizeof(bc->connected.number));
Add private representation of caller, connected and redirecting party ids. This patch adds the feature "Private representation of caller, connected and redirecting party ids", as previously discussed with us (DATUS) and Digium. 1. Feature motivation Until now it is quite difficult to modify a party number or name which can only be seen by exactly one particular instantiated technology channel subscriber. One example where a modified party number or name on one channel is spread over several channels are supplementary services like call transfer or pickup. To implement these features Asterisk internally copies caller and connected ids from one channel to another. Another example are extension subscriptions. The monitoring entities (watchers) are notified of state changes and - if desired - of party numbers or names which represent the involving call parties. One major feature where a private representation of party names is essentially needed, i.e. where a party name shall be exclusively signaled to only one particular user, is a private user-specific name resolution for party numbers. A lookup in a private destination-dependent telephone book shall provide party names which cannot be seen by any other user at any time. 2. Feature Description This feature comes along with the implementation of additional private party id elements for caller id, connected id and redirecting ids inside Asterisk channels. The private party id elements can be read or set by the user using Asterisk dialplan functions. When a technology channel is initiating a call, receives an internal connected-line update event, or receives an internal redirecting update event, it merges the corresponding public id with the private id to create an effective party id. The effective party id is then used for protocol signaling. The channel technologies which initially support the private id representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and PRI (chan_dahdi). Once a private name or number on a channel is set and (implicitly) made valid, it is generally used for any further protocol signaling until it is rewritten or invalidated. To simplify the invalidation of private ids all internally generated connected/redirecting update events and also all connected/redirecting update events which are generated by technology channels -- receiving regarding protocol information - automatically trigger the invalidation of private ids. If not using the private party id representation feature at all, i.e. if using only the 'regular' caller-id, connected and redirecting related functions, the current characteristic of Asterisk is not affected by the new extended functionality. 3. User interface Description To grant access to the private name and number representation from the Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan functions are extended by the following data types. The formats of these data types are equal to the corresponding regular 'non-private' already existing data types: CALLERID: priv-all priv-name priv-name-valid priv-name-charset priv-name-pres priv-num priv-num-valid priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE: priv-name priv-name-valid priv-name-pres priv-name-charset priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag REDIRECTING: priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd priv-orig-tag priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd priv-to-tag Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2030/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-10 19:54:55 +00:00
bc->connected.presentation = ast_to_misdn_pres(connected_id.number.presentation);
bc->connected.screening = ast_to_misdn_screen(connected_id.number.presentation);
bc->connected.number_type = ast_to_misdn_ton(connected_id.number.plan);
bc->connected.number_plan = ast_to_misdn_plan(connected_id.number.plan);
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
} else {
bc->connected.number[0] = '\0';
bc->connected.presentation = 0;/* Allowed */
bc->connected.screening = 0;/* Unscreened */
bc->connected.number_type = NUMTYPE_UNKNOWN;
bc->connected.number_plan = NUMPLAN_UNKNOWN;
}
misdn_cfg_get(bc->port, MISDN_CFG_CPNDIALPLAN, &number_type, sizeof(number_type));
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
if (0 <= number_type) {
/* Force us to send in CONNECT message */
bc->connected.number_type = number_type;
bc->connected.number_plan = NUMPLAN_ISDN;
}
debug_numtype(bc->port, bc->connected.number_type, "CTON");
} else {
/* ORIGINATOR Asterisk (outgoing call) */
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
ast_copy_string(bc->caller.name,
Add private representation of caller, connected and redirecting party ids. This patch adds the feature "Private representation of caller, connected and redirecting party ids", as previously discussed with us (DATUS) and Digium. 1. Feature motivation Until now it is quite difficult to modify a party number or name which can only be seen by exactly one particular instantiated technology channel subscriber. One example where a modified party number or name on one channel is spread over several channels are supplementary services like call transfer or pickup. To implement these features Asterisk internally copies caller and connected ids from one channel to another. Another example are extension subscriptions. The monitoring entities (watchers) are notified of state changes and - if desired - of party numbers or names which represent the involving call parties. One major feature where a private representation of party names is essentially needed, i.e. where a party name shall be exclusively signaled to only one particular user, is a private user-specific name resolution for party numbers. A lookup in a private destination-dependent telephone book shall provide party names which cannot be seen by any other user at any time. 2. Feature Description This feature comes along with the implementation of additional private party id elements for caller id, connected id and redirecting ids inside Asterisk channels. The private party id elements can be read or set by the user using Asterisk dialplan functions. When a technology channel is initiating a call, receives an internal connected-line update event, or receives an internal redirecting update event, it merges the corresponding public id with the private id to create an effective party id. The effective party id is then used for protocol signaling. The channel technologies which initially support the private id representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and PRI (chan_dahdi). Once a private name or number on a channel is set and (implicitly) made valid, it is generally used for any further protocol signaling until it is rewritten or invalidated. To simplify the invalidation of private ids all internally generated connected/redirecting update events and also all connected/redirecting update events which are generated by technology channels -- receiving regarding protocol information - automatically trigger the invalidation of private ids. If not using the private party id representation feature at all, i.e. if using only the 'regular' caller-id, connected and redirecting related functions, the current characteristic of Asterisk is not affected by the new extended functionality. 3. User interface Description To grant access to the private name and number representation from the Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan functions are extended by the following data types. The formats of these data types are equal to the corresponding regular 'non-private' already existing data types: CALLERID: priv-all priv-name priv-name-valid priv-name-charset priv-name-pres priv-num priv-num-valid priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE: priv-name priv-name-valid priv-name-pres priv-name-charset priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag REDIRECTING: priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd priv-orig-tag priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd priv-to-tag Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2030/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-10 19:54:55 +00:00
S_COR(connected_id.name.valid, connected_id.name.str, ""),
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
sizeof(bc->caller.name));
Add private representation of caller, connected and redirecting party ids. This patch adds the feature "Private representation of caller, connected and redirecting party ids", as previously discussed with us (DATUS) and Digium. 1. Feature motivation Until now it is quite difficult to modify a party number or name which can only be seen by exactly one particular instantiated technology channel subscriber. One example where a modified party number or name on one channel is spread over several channels are supplementary services like call transfer or pickup. To implement these features Asterisk internally copies caller and connected ids from one channel to another. Another example are extension subscriptions. The monitoring entities (watchers) are notified of state changes and - if desired - of party numbers or names which represent the involving call parties. One major feature where a private representation of party names is essentially needed, i.e. where a party name shall be exclusively signaled to only one particular user, is a private user-specific name resolution for party numbers. A lookup in a private destination-dependent telephone book shall provide party names which cannot be seen by any other user at any time. 2. Feature Description This feature comes along with the implementation of additional private party id elements for caller id, connected id and redirecting ids inside Asterisk channels. The private party id elements can be read or set by the user using Asterisk dialplan functions. When a technology channel is initiating a call, receives an internal connected-line update event, or receives an internal redirecting update event, it merges the corresponding public id with the private id to create an effective party id. The effective party id is then used for protocol signaling. The channel technologies which initially support the private id representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and PRI (chan_dahdi). Once a private name or number on a channel is set and (implicitly) made valid, it is generally used for any further protocol signaling until it is rewritten or invalidated. To simplify the invalidation of private ids all internally generated connected/redirecting update events and also all connected/redirecting update events which are generated by technology channels -- receiving regarding protocol information - automatically trigger the invalidation of private ids. If not using the private party id representation feature at all, i.e. if using only the 'regular' caller-id, connected and redirecting related functions, the current characteristic of Asterisk is not affected by the new extended functionality. 3. User interface Description To grant access to the private name and number representation from the Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan functions are extended by the following data types. The formats of these data types are equal to the corresponding regular 'non-private' already existing data types: CALLERID: priv-all priv-name priv-name-valid priv-name-charset priv-name-pres priv-num priv-num-valid priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE: priv-name priv-name-valid priv-name-pres priv-name-charset priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag REDIRECTING: priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd priv-orig-tag priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd priv-to-tag Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2030/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-10 19:54:55 +00:00
if (connected_id.number.valid) {
ast_copy_string(bc->caller.number, S_OR(connected_id.number.str, ""),
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
sizeof(bc->caller.number));
Add private representation of caller, connected and redirecting party ids. This patch adds the feature "Private representation of caller, connected and redirecting party ids", as previously discussed with us (DATUS) and Digium. 1. Feature motivation Until now it is quite difficult to modify a party number or name which can only be seen by exactly one particular instantiated technology channel subscriber. One example where a modified party number or name on one channel is spread over several channels are supplementary services like call transfer or pickup. To implement these features Asterisk internally copies caller and connected ids from one channel to another. Another example are extension subscriptions. The monitoring entities (watchers) are notified of state changes and - if desired - of party numbers or names which represent the involving call parties. One major feature where a private representation of party names is essentially needed, i.e. where a party name shall be exclusively signaled to only one particular user, is a private user-specific name resolution for party numbers. A lookup in a private destination-dependent telephone book shall provide party names which cannot be seen by any other user at any time. 2. Feature Description This feature comes along with the implementation of additional private party id elements for caller id, connected id and redirecting ids inside Asterisk channels. The private party id elements can be read or set by the user using Asterisk dialplan functions. When a technology channel is initiating a call, receives an internal connected-line update event, or receives an internal redirecting update event, it merges the corresponding public id with the private id to create an effective party id. The effective party id is then used for protocol signaling. The channel technologies which initially support the private id representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and PRI (chan_dahdi). Once a private name or number on a channel is set and (implicitly) made valid, it is generally used for any further protocol signaling until it is rewritten or invalidated. To simplify the invalidation of private ids all internally generated connected/redirecting update events and also all connected/redirecting update events which are generated by technology channels -- receiving regarding protocol information - automatically trigger the invalidation of private ids. If not using the private party id representation feature at all, i.e. if using only the 'regular' caller-id, connected and redirecting related functions, the current characteristic of Asterisk is not affected by the new extended functionality. 3. User interface Description To grant access to the private name and number representation from the Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan functions are extended by the following data types. The formats of these data types are equal to the corresponding regular 'non-private' already existing data types: CALLERID: priv-all priv-name priv-name-valid priv-name-charset priv-name-pres priv-num priv-num-valid priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE: priv-name priv-name-valid priv-name-pres priv-name-charset priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag REDIRECTING: priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd priv-orig-tag priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd priv-to-tag Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2030/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-10 19:54:55 +00:00
bc->caller.presentation = ast_to_misdn_pres(connected_id.number.presentation);
bc->caller.screening = ast_to_misdn_screen(connected_id.number.presentation);
bc->caller.number_type = ast_to_misdn_ton(connected_id.number.plan);
bc->caller.number_plan = ast_to_misdn_plan(connected_id.number.plan);
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
} else {
bc->caller.number[0] = '\0';
bc->caller.presentation = 0;/* Allowed */
bc->caller.screening = 0;/* Unscreened */
bc->caller.number_type = NUMTYPE_UNKNOWN;
bc->caller.number_plan = NUMPLAN_UNKNOWN;
}
misdn_cfg_get(bc->port, MISDN_CFG_LOCALDIALPLAN, &number_type, sizeof(number_type));
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
if (0 <= number_type) {
/* Force us to send in SETUP message */
bc->caller.number_type = number_type;
bc->caller.number_plan = NUMPLAN_ISDN;
}
debug_numtype(bc->port, bc->caller.number_type, "LTON");
}
}
/*!
* \internal
* \brief Notify peer that the connected line has changed.
*
* \param ast Current Asterisk channel
* \param bc Associated B channel
* \param originator Who originally created this channel. ORG_AST or ORG_MISDN
*
* \return Nothing
*/
static void misdn_update_connected_line(struct ast_channel *ast, struct misdn_bchannel *bc, int originator)
{
struct chan_list *ch;
misdn_get_connected_line(ast, bc, originator);
if (originator == ORG_MISDN) {
bc->redirecting.to = bc->connected;
} else {
bc->redirecting.to = bc->caller;
}
switch (bc->outgoing_colp) {
case 1:/* restricted */
bc->redirecting.to.presentation = 1;/* restricted */
break;
case 2:/* blocked */
/* Don't tell the remote party that the call was transferred. */
return;
default:
break;
}
ch = MISDN_ASTERISK_TECH_PVT(ast);
if (ch->state == MISDN_CONNECTED
|| originator != ORG_MISDN) {
int is_ptmp;
is_ptmp = !misdn_lib_is_ptp(bc->port);
if (is_ptmp) {
Merged revisions 287017 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r287017 | rmudgett | 2010-09-15 15:53:38 -0500 (Wed, 15 Sep 2010) | 65 lines Merged revision 287014 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r287014 | rmudgett | 2010-09-15 15:32:24 -0500 (Wed, 15 Sep 2010) | 58 lines The handling of call transfer signaling for mISDN PTMP is not fully implemented. The handling of call transfer signaling for mISDN PTMP is not fully implemented. The signaling of number updates with ISDN/DSS1 ECT supplementary services (ETS 300 369-1) comes along with a notification indicator IE and redirection number IE for PTMP. The implementation in the current Asterisk mISDN channel unfortunately can handle these information elements only in a NOTIFY message. These information elements are also signaled in a FACILTY message with a RequestSubaddress facility, when the subscriber is already in the active state (see 9.2.4 and 9.2.5 of ETS 300 369-1). ********** abe_2526_ast.patch * Added support to handle the notification indicator IE and redirection number IE with the RequestSubaddress facility. * Made misdn_update_connected_line() send a NOTIFY message if Asterisk originated the call and it is not connected yet. * Made misdn_update_connected_line() send a FACILITY message if the call is already connected. This patch requires the presence of the associated mISDN patches to compile. I had to enhance mISDN to allow the notification indicator IE and the redirection number IE to be used with a FACILITY message. Earlier versions of the Digium enhanced mISDN are no longer going to work. ********** abe_2526_misdn.patch * Made an incoming FACILITY message allow the presence of the notification indicator IE and the redirection number IE. ********** abe_2526_misdnuser_v3.patch * Added support to send and receive a FACILITY message with the notification indicator IE and the redirection number IE. * Added the ability to send a NOTIFY message in PTMP/NT mode to all responding subcalls in Q.931 states 6, 7, 8, 9, and 25. ********** Patches: abe_2526_ast.patch uploaded by rmudgett (license 664) abe_2526_misdn.patch uploaded by rmudgett (license 664) abe_2526_misdnuser_v3.patch uploaded by rmudgett (license 664) Tested by: rmudgett and reporter JIRA SWP-2146 JIRA ABE-2526 .......... ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287018 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-15 20:56:21 +00:00
/*
* We should not send these messages to the network if we are
* the CPE side since phones do not transfer calls within
* themselves. Well... If you consider handing the handset to
* someone else a transfer then how is the network to know?
*/
if (!misdn_lib_port_is_nt(bc->port)) {
return;
}
if (ch->state != MISDN_CONNECTED) {
/* Send NOTIFY(Nie(transfer-active), RDNie(redirecting.to data)) */
bc->redirecting.to_changed = 1;
bc->notify_description_code = mISDN_NOTIFY_CODE_CALL_TRANSFER_ACTIVE;
misdn_lib_send_event(bc, EVENT_NOTIFY);
#if defined(AST_MISDN_ENHANCEMENTS)
} else {
/* Send FACILITY(Fie(RequestSubaddress), Nie(transfer-active), RDNie(redirecting.to data)) */
bc->redirecting.to_changed = 1;
bc->notify_description_code = mISDN_NOTIFY_CODE_CALL_TRANSFER_ACTIVE;
bc->fac_out.Function = Fac_RequestSubaddress;
bc->fac_out.u.RequestSubaddress.InvokeID = ++misdn_invoke_id;
/* Send message */
print_facility(&bc->fac_out, bc);
misdn_lib_send_event(bc, EVENT_FACILITY);
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
}
#if defined(AST_MISDN_ENHANCEMENTS)
} else {
Merged revisions 287017 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r287017 | rmudgett | 2010-09-15 15:53:38 -0500 (Wed, 15 Sep 2010) | 65 lines Merged revision 287014 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r287014 | rmudgett | 2010-09-15 15:32:24 -0500 (Wed, 15 Sep 2010) | 58 lines The handling of call transfer signaling for mISDN PTMP is not fully implemented. The handling of call transfer signaling for mISDN PTMP is not fully implemented. The signaling of number updates with ISDN/DSS1 ECT supplementary services (ETS 300 369-1) comes along with a notification indicator IE and redirection number IE for PTMP. The implementation in the current Asterisk mISDN channel unfortunately can handle these information elements only in a NOTIFY message. These information elements are also signaled in a FACILTY message with a RequestSubaddress facility, when the subscriber is already in the active state (see 9.2.4 and 9.2.5 of ETS 300 369-1). ********** abe_2526_ast.patch * Added support to handle the notification indicator IE and redirection number IE with the RequestSubaddress facility. * Made misdn_update_connected_line() send a NOTIFY message if Asterisk originated the call and it is not connected yet. * Made misdn_update_connected_line() send a FACILITY message if the call is already connected. This patch requires the presence of the associated mISDN patches to compile. I had to enhance mISDN to allow the notification indicator IE and the redirection number IE to be used with a FACILITY message. Earlier versions of the Digium enhanced mISDN are no longer going to work. ********** abe_2526_misdn.patch * Made an incoming FACILITY message allow the presence of the notification indicator IE and the redirection number IE. ********** abe_2526_misdnuser_v3.patch * Added support to send and receive a FACILITY message with the notification indicator IE and the redirection number IE. * Added the ability to send a NOTIFY message in PTMP/NT mode to all responding subcalls in Q.931 states 6, 7, 8, 9, and 25. ********** Patches: abe_2526_ast.patch uploaded by rmudgett (license 664) abe_2526_misdn.patch uploaded by rmudgett (license 664) abe_2526_misdnuser_v3.patch uploaded by rmudgett (license 664) Tested by: rmudgett and reporter JIRA SWP-2146 JIRA ABE-2526 .......... ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287018 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-15 20:56:21 +00:00
/* Send FACILITY(Fie(EctInform(transfer-active, redirecting.to data))) */
bc->fac_out.Function = Fac_EctInform;
bc->fac_out.u.EctInform.InvokeID = ++misdn_invoke_id;
bc->fac_out.u.EctInform.Status = 1;/* active */
bc->fac_out.u.EctInform.RedirectionPresent = 1;/* Must be present when status is active */
misdn_PresentedNumberUnscreened_fill(&bc->fac_out.u.EctInform.Redirection,
&bc->redirecting.to);
/* Send message */
print_facility(&bc->fac_out, bc);
misdn_lib_send_event(bc, EVENT_FACILITY);
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
}
}
}
/*!
* \internal
* \brief Copy the redirecting information out of the Asterisk channel
*
* \param bc Associated B channel
* \param ast Current Asterisk channel
*
* \return Nothing
*/
static void misdn_copy_redirecting_from_ast(struct misdn_bchannel *bc, struct ast_channel *ast)
{
Add private representation of caller, connected and redirecting party ids. This patch adds the feature "Private representation of caller, connected and redirecting party ids", as previously discussed with us (DATUS) and Digium. 1. Feature motivation Until now it is quite difficult to modify a party number or name which can only be seen by exactly one particular instantiated technology channel subscriber. One example where a modified party number or name on one channel is spread over several channels are supplementary services like call transfer or pickup. To implement these features Asterisk internally copies caller and connected ids from one channel to another. Another example are extension subscriptions. The monitoring entities (watchers) are notified of state changes and - if desired - of party numbers or names which represent the involving call parties. One major feature where a private representation of party names is essentially needed, i.e. where a party name shall be exclusively signaled to only one particular user, is a private user-specific name resolution for party numbers. A lookup in a private destination-dependent telephone book shall provide party names which cannot be seen by any other user at any time. 2. Feature Description This feature comes along with the implementation of additional private party id elements for caller id, connected id and redirecting ids inside Asterisk channels. The private party id elements can be read or set by the user using Asterisk dialplan functions. When a technology channel is initiating a call, receives an internal connected-line update event, or receives an internal redirecting update event, it merges the corresponding public id with the private id to create an effective party id. The effective party id is then used for protocol signaling. The channel technologies which initially support the private id representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and PRI (chan_dahdi). Once a private name or number on a channel is set and (implicitly) made valid, it is generally used for any further protocol signaling until it is rewritten or invalidated. To simplify the invalidation of private ids all internally generated connected/redirecting update events and also all connected/redirecting update events which are generated by technology channels -- receiving regarding protocol information - automatically trigger the invalidation of private ids. If not using the private party id representation feature at all, i.e. if using only the 'regular' caller-id, connected and redirecting related functions, the current characteristic of Asterisk is not affected by the new extended functionality. 3. User interface Description To grant access to the private name and number representation from the Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan functions are extended by the following data types. The formats of these data types are equal to the corresponding regular 'non-private' already existing data types: CALLERID: priv-all priv-name priv-name-valid priv-name-charset priv-name-pres priv-num priv-num-valid priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE: priv-name priv-name-valid priv-name-pres priv-name-charset priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag REDIRECTING: priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd priv-orig-tag priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd priv-to-tag Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2030/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-10 19:54:55 +00:00
struct ast_party_id from_id = ast_channel_redirecting_effective_from(ast);
struct ast_party_id to_id = ast_channel_redirecting_effective_to(ast);
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
ast_copy_string(bc->redirecting.from.name,
Add private representation of caller, connected and redirecting party ids. This patch adds the feature "Private representation of caller, connected and redirecting party ids", as previously discussed with us (DATUS) and Digium. 1. Feature motivation Until now it is quite difficult to modify a party number or name which can only be seen by exactly one particular instantiated technology channel subscriber. One example where a modified party number or name on one channel is spread over several channels are supplementary services like call transfer or pickup. To implement these features Asterisk internally copies caller and connected ids from one channel to another. Another example are extension subscriptions. The monitoring entities (watchers) are notified of state changes and - if desired - of party numbers or names which represent the involving call parties. One major feature where a private representation of party names is essentially needed, i.e. where a party name shall be exclusively signaled to only one particular user, is a private user-specific name resolution for party numbers. A lookup in a private destination-dependent telephone book shall provide party names which cannot be seen by any other user at any time. 2. Feature Description This feature comes along with the implementation of additional private party id elements for caller id, connected id and redirecting ids inside Asterisk channels. The private party id elements can be read or set by the user using Asterisk dialplan functions. When a technology channel is initiating a call, receives an internal connected-line update event, or receives an internal redirecting update event, it merges the corresponding public id with the private id to create an effective party id. The effective party id is then used for protocol signaling. The channel technologies which initially support the private id representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and PRI (chan_dahdi). Once a private name or number on a channel is set and (implicitly) made valid, it is generally used for any further protocol signaling until it is rewritten or invalidated. To simplify the invalidation of private ids all internally generated connected/redirecting update events and also all connected/redirecting update events which are generated by technology channels -- receiving regarding protocol information - automatically trigger the invalidation of private ids. If not using the private party id representation feature at all, i.e. if using only the 'regular' caller-id, connected and redirecting related functions, the current characteristic of Asterisk is not affected by the new extended functionality. 3. User interface Description To grant access to the private name and number representation from the Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan functions are extended by the following data types. The formats of these data types are equal to the corresponding regular 'non-private' already existing data types: CALLERID: priv-all priv-name priv-name-valid priv-name-charset priv-name-pres priv-num priv-num-valid priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE: priv-name priv-name-valid priv-name-pres priv-name-charset priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag REDIRECTING: priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd priv-orig-tag priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd priv-to-tag Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2030/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-10 19:54:55 +00:00
S_COR(from_id.name.valid, from_id.name.str, ""),
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
sizeof(bc->redirecting.from.name));
Add private representation of caller, connected and redirecting party ids. This patch adds the feature "Private representation of caller, connected and redirecting party ids", as previously discussed with us (DATUS) and Digium. 1. Feature motivation Until now it is quite difficult to modify a party number or name which can only be seen by exactly one particular instantiated technology channel subscriber. One example where a modified party number or name on one channel is spread over several channels are supplementary services like call transfer or pickup. To implement these features Asterisk internally copies caller and connected ids from one channel to another. Another example are extension subscriptions. The monitoring entities (watchers) are notified of state changes and - if desired - of party numbers or names which represent the involving call parties. One major feature where a private representation of party names is essentially needed, i.e. where a party name shall be exclusively signaled to only one particular user, is a private user-specific name resolution for party numbers. A lookup in a private destination-dependent telephone book shall provide party names which cannot be seen by any other user at any time. 2. Feature Description This feature comes along with the implementation of additional private party id elements for caller id, connected id and redirecting ids inside Asterisk channels. The private party id elements can be read or set by the user using Asterisk dialplan functions. When a technology channel is initiating a call, receives an internal connected-line update event, or receives an internal redirecting update event, it merges the corresponding public id with the private id to create an effective party id. The effective party id is then used for protocol signaling. The channel technologies which initially support the private id representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and PRI (chan_dahdi). Once a private name or number on a channel is set and (implicitly) made valid, it is generally used for any further protocol signaling until it is rewritten or invalidated. To simplify the invalidation of private ids all internally generated connected/redirecting update events and also all connected/redirecting update events which are generated by technology channels -- receiving regarding protocol information - automatically trigger the invalidation of private ids. If not using the private party id representation feature at all, i.e. if using only the 'regular' caller-id, connected and redirecting related functions, the current characteristic of Asterisk is not affected by the new extended functionality. 3. User interface Description To grant access to the private name and number representation from the Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan functions are extended by the following data types. The formats of these data types are equal to the corresponding regular 'non-private' already existing data types: CALLERID: priv-all priv-name priv-name-valid priv-name-charset priv-name-pres priv-num priv-num-valid priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE: priv-name priv-name-valid priv-name-pres priv-name-charset priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag REDIRECTING: priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd priv-orig-tag priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd priv-to-tag Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2030/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-10 19:54:55 +00:00
if (from_id.number.valid) {
ast_copy_string(bc->redirecting.from.number, S_OR(from_id.number.str, ""),
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
sizeof(bc->redirecting.from.number));
Add private representation of caller, connected and redirecting party ids. This patch adds the feature "Private representation of caller, connected and redirecting party ids", as previously discussed with us (DATUS) and Digium. 1. Feature motivation Until now it is quite difficult to modify a party number or name which can only be seen by exactly one particular instantiated technology channel subscriber. One example where a modified party number or name on one channel is spread over several channels are supplementary services like call transfer or pickup. To implement these features Asterisk internally copies caller and connected ids from one channel to another. Another example are extension subscriptions. The monitoring entities (watchers) are notified of state changes and - if desired - of party numbers or names which represent the involving call parties. One major feature where a private representation of party names is essentially needed, i.e. where a party name shall be exclusively signaled to only one particular user, is a private user-specific name resolution for party numbers. A lookup in a private destination-dependent telephone book shall provide party names which cannot be seen by any other user at any time. 2. Feature Description This feature comes along with the implementation of additional private party id elements for caller id, connected id and redirecting ids inside Asterisk channels. The private party id elements can be read or set by the user using Asterisk dialplan functions. When a technology channel is initiating a call, receives an internal connected-line update event, or receives an internal redirecting update event, it merges the corresponding public id with the private id to create an effective party id. The effective party id is then used for protocol signaling. The channel technologies which initially support the private id representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and PRI (chan_dahdi). Once a private name or number on a channel is set and (implicitly) made valid, it is generally used for any further protocol signaling until it is rewritten or invalidated. To simplify the invalidation of private ids all internally generated connected/redirecting update events and also all connected/redirecting update events which are generated by technology channels -- receiving regarding protocol information - automatically trigger the invalidation of private ids. If not using the private party id representation feature at all, i.e. if using only the 'regular' caller-id, connected and redirecting related functions, the current characteristic of Asterisk is not affected by the new extended functionality. 3. User interface Description To grant access to the private name and number representation from the Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan functions are extended by the following data types. The formats of these data types are equal to the corresponding regular 'non-private' already existing data types: CALLERID: priv-all priv-name priv-name-valid priv-name-charset priv-name-pres priv-num priv-num-valid priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE: priv-name priv-name-valid priv-name-pres priv-name-charset priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag REDIRECTING: priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd priv-orig-tag priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd priv-to-tag Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2030/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-10 19:54:55 +00:00
bc->redirecting.from.presentation = ast_to_misdn_pres(from_id.number.presentation);
bc->redirecting.from.screening = ast_to_misdn_screen(from_id.number.presentation);
bc->redirecting.from.number_type = ast_to_misdn_ton(from_id.number.plan);
bc->redirecting.from.number_plan = ast_to_misdn_plan(from_id.number.plan);
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
} else {
bc->redirecting.from.number[0] = '\0';
bc->redirecting.from.presentation = 0;/* Allowed */
bc->redirecting.from.screening = 0;/* Unscreened */
bc->redirecting.from.number_type = NUMTYPE_UNKNOWN;
bc->redirecting.from.number_plan = NUMPLAN_UNKNOWN;
}
ast_copy_string(bc->redirecting.to.name,
Add private representation of caller, connected and redirecting party ids. This patch adds the feature "Private representation of caller, connected and redirecting party ids", as previously discussed with us (DATUS) and Digium. 1. Feature motivation Until now it is quite difficult to modify a party number or name which can only be seen by exactly one particular instantiated technology channel subscriber. One example where a modified party number or name on one channel is spread over several channels are supplementary services like call transfer or pickup. To implement these features Asterisk internally copies caller and connected ids from one channel to another. Another example are extension subscriptions. The monitoring entities (watchers) are notified of state changes and - if desired - of party numbers or names which represent the involving call parties. One major feature where a private representation of party names is essentially needed, i.e. where a party name shall be exclusively signaled to only one particular user, is a private user-specific name resolution for party numbers. A lookup in a private destination-dependent telephone book shall provide party names which cannot be seen by any other user at any time. 2. Feature Description This feature comes along with the implementation of additional private party id elements for caller id, connected id and redirecting ids inside Asterisk channels. The private party id elements can be read or set by the user using Asterisk dialplan functions. When a technology channel is initiating a call, receives an internal connected-line update event, or receives an internal redirecting update event, it merges the corresponding public id with the private id to create an effective party id. The effective party id is then used for protocol signaling. The channel technologies which initially support the private id representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and PRI (chan_dahdi). Once a private name or number on a channel is set and (implicitly) made valid, it is generally used for any further protocol signaling until it is rewritten or invalidated. To simplify the invalidation of private ids all internally generated connected/redirecting update events and also all connected/redirecting update events which are generated by technology channels -- receiving regarding protocol information - automatically trigger the invalidation of private ids. If not using the private party id representation feature at all, i.e. if using only the 'regular' caller-id, connected and redirecting related functions, the current characteristic of Asterisk is not affected by the new extended functionality. 3. User interface Description To grant access to the private name and number representation from the Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan functions are extended by the following data types. The formats of these data types are equal to the corresponding regular 'non-private' already existing data types: CALLERID: priv-all priv-name priv-name-valid priv-name-charset priv-name-pres priv-num priv-num-valid priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE: priv-name priv-name-valid priv-name-pres priv-name-charset priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag REDIRECTING: priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd priv-orig-tag priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd priv-to-tag Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2030/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-10 19:54:55 +00:00
S_COR(to_id.name.valid, to_id.name.str, ""),
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
sizeof(bc->redirecting.to.name));
Add private representation of caller, connected and redirecting party ids. This patch adds the feature "Private representation of caller, connected and redirecting party ids", as previously discussed with us (DATUS) and Digium. 1. Feature motivation Until now it is quite difficult to modify a party number or name which can only be seen by exactly one particular instantiated technology channel subscriber. One example where a modified party number or name on one channel is spread over several channels are supplementary services like call transfer or pickup. To implement these features Asterisk internally copies caller and connected ids from one channel to another. Another example are extension subscriptions. The monitoring entities (watchers) are notified of state changes and - if desired - of party numbers or names which represent the involving call parties. One major feature where a private representation of party names is essentially needed, i.e. where a party name shall be exclusively signaled to only one particular user, is a private user-specific name resolution for party numbers. A lookup in a private destination-dependent telephone book shall provide party names which cannot be seen by any other user at any time. 2. Feature Description This feature comes along with the implementation of additional private party id elements for caller id, connected id and redirecting ids inside Asterisk channels. The private party id elements can be read or set by the user using Asterisk dialplan functions. When a technology channel is initiating a call, receives an internal connected-line update event, or receives an internal redirecting update event, it merges the corresponding public id with the private id to create an effective party id. The effective party id is then used for protocol signaling. The channel technologies which initially support the private id representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and PRI (chan_dahdi). Once a private name or number on a channel is set and (implicitly) made valid, it is generally used for any further protocol signaling until it is rewritten or invalidated. To simplify the invalidation of private ids all internally generated connected/redirecting update events and also all connected/redirecting update events which are generated by technology channels -- receiving regarding protocol information - automatically trigger the invalidation of private ids. If not using the private party id representation feature at all, i.e. if using only the 'regular' caller-id, connected and redirecting related functions, the current characteristic of Asterisk is not affected by the new extended functionality. 3. User interface Description To grant access to the private name and number representation from the Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan functions are extended by the following data types. The formats of these data types are equal to the corresponding regular 'non-private' already existing data types: CALLERID: priv-all priv-name priv-name-valid priv-name-charset priv-name-pres priv-num priv-num-valid priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE: priv-name priv-name-valid priv-name-pres priv-name-charset priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag REDIRECTING: priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd priv-orig-tag priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd priv-to-tag Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2030/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-10 19:54:55 +00:00
if (to_id.number.valid) {
ast_copy_string(bc->redirecting.to.number, S_OR(to_id.number.str, ""),
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
sizeof(bc->redirecting.to.number));
Add private representation of caller, connected and redirecting party ids. This patch adds the feature "Private representation of caller, connected and redirecting party ids", as previously discussed with us (DATUS) and Digium. 1. Feature motivation Until now it is quite difficult to modify a party number or name which can only be seen by exactly one particular instantiated technology channel subscriber. One example where a modified party number or name on one channel is spread over several channels are supplementary services like call transfer or pickup. To implement these features Asterisk internally copies caller and connected ids from one channel to another. Another example are extension subscriptions. The monitoring entities (watchers) are notified of state changes and - if desired - of party numbers or names which represent the involving call parties. One major feature where a private representation of party names is essentially needed, i.e. where a party name shall be exclusively signaled to only one particular user, is a private user-specific name resolution for party numbers. A lookup in a private destination-dependent telephone book shall provide party names which cannot be seen by any other user at any time. 2. Feature Description This feature comes along with the implementation of additional private party id elements for caller id, connected id and redirecting ids inside Asterisk channels. The private party id elements can be read or set by the user using Asterisk dialplan functions. When a technology channel is initiating a call, receives an internal connected-line update event, or receives an internal redirecting update event, it merges the corresponding public id with the private id to create an effective party id. The effective party id is then used for protocol signaling. The channel technologies which initially support the private id representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and PRI (chan_dahdi). Once a private name or number on a channel is set and (implicitly) made valid, it is generally used for any further protocol signaling until it is rewritten or invalidated. To simplify the invalidation of private ids all internally generated connected/redirecting update events and also all connected/redirecting update events which are generated by technology channels -- receiving regarding protocol information - automatically trigger the invalidation of private ids. If not using the private party id representation feature at all, i.e. if using only the 'regular' caller-id, connected and redirecting related functions, the current characteristic of Asterisk is not affected by the new extended functionality. 3. User interface Description To grant access to the private name and number representation from the Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan functions are extended by the following data types. The formats of these data types are equal to the corresponding regular 'non-private' already existing data types: CALLERID: priv-all priv-name priv-name-valid priv-name-charset priv-name-pres priv-num priv-num-valid priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE: priv-name priv-name-valid priv-name-pres priv-name-charset priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag REDIRECTING: priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd priv-orig-tag priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd priv-to-tag Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2030/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-10 19:54:55 +00:00
bc->redirecting.to.presentation = ast_to_misdn_pres(to_id.number.presentation);
bc->redirecting.to.screening = ast_to_misdn_screen(to_id.number.presentation);
bc->redirecting.to.number_type = ast_to_misdn_ton(to_id.number.plan);
bc->redirecting.to.number_plan = ast_to_misdn_plan(to_id.number.plan);
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
} else {
bc->redirecting.to.number[0] = '\0';
bc->redirecting.to.presentation = 0;/* Allowed */
bc->redirecting.to.screening = 0;/* Unscreened */
bc->redirecting.to.number_type = NUMTYPE_UNKNOWN;
bc->redirecting.to.number_plan = NUMPLAN_UNKNOWN;
}
bc->redirecting.reason = ast_to_misdn_reason(ast_channel_redirecting(ast)->reason.code);
bc->redirecting.count = ast_channel_redirecting(ast)->count;
}
/*!
* \internal
* \brief Copy the redirecting info into the Asterisk channel
*
* \param ast Current Asterisk channel
* \param redirect Associated B channel redirecting info
Enhancements to connected line and redirecting work. From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
* \param tag Caller ID tag to set in the redirecting party fields
*
* \return Nothing
*/
Enhancements to connected line and redirecting work. From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
static void misdn_copy_redirecting_to_ast(struct ast_channel *ast, const struct misdn_party_redirecting *redirect, char *tag)
{
struct ast_party_redirecting redirecting;
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
struct ast_set_party_redirecting update_redirecting;
ast_party_redirecting_set_init(&redirecting, ast_channel_redirecting(ast));
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
memset(&update_redirecting, 0, sizeof(update_redirecting));
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
update_redirecting.from.number = 1;
redirecting.from.number.valid = 1;
redirecting.from.number.str = (char *) redirect->from.number;
redirecting.from.number.plan =
misdn_to_ast_ton(redirect->from.number_type)
| misdn_to_ast_plan(redirect->from.number_plan);
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
redirecting.from.number.presentation =
misdn_to_ast_pres(redirect->from.presentation)
| misdn_to_ast_screen(redirect->from.screening);
Enhancements to connected line and redirecting work. From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
redirecting.from.tag = tag;
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
update_redirecting.to.number = 1;
redirecting.to.number.valid = 1;
redirecting.to.number.str = (char *) redirect->to.number;
redirecting.to.number.plan =
misdn_to_ast_ton(redirect->to.number_type)
| misdn_to_ast_plan(redirect->to.number_plan);
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
redirecting.to.number.presentation =
misdn_to_ast_pres(redirect->to.presentation)
| misdn_to_ast_screen(redirect->to.screening);
Enhancements to connected line and redirecting work. From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
redirecting.to.tag = tag;
redirecting.reason.code = misdn_to_ast_reason(redirect->reason);
redirecting.count = redirect->count;
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
ast_channel_set_redirecting(ast, &redirecting, &update_redirecting);
}
/*!
* \internal
* \brief Notify peer that the redirecting information has changed.
*
* \param ast Current Asterisk channel
* \param bc Associated B channel
* \param originator Who originally created this channel. ORG_AST or ORG_MISDN
*
* \return Nothing
*/
static void misdn_update_redirecting(struct ast_channel *ast, struct misdn_bchannel *bc, int originator)
{
int is_ptmp;
misdn_copy_redirecting_from_ast(bc, ast);
switch (bc->outgoing_colp) {
case 1:/* restricted */
bc->redirecting.to.presentation = 1;/* restricted */
break;
case 2:/* blocked */
/* Don't tell the remote party that the call was redirected. */
return;
default:
break;
}
if (originator != ORG_MISDN) {
return;
}
is_ptmp = !misdn_lib_is_ptp(bc->port);
if (is_ptmp) {
/*
* We should not send these messages to the network if we are
* the CPE side since phones do not redirect calls within
* themselves. Well... If you consider someone else picking up
* the handset a redirection then how is the network to know?
*/
if (!misdn_lib_port_is_nt(bc->port)) {
return;
}
/* Send NOTIFY(call-is-diverting, redirecting.to data) */
bc->redirecting.to_changed = 1;
bc->notify_description_code = mISDN_NOTIFY_CODE_CALL_IS_DIVERTING;
misdn_lib_send_event(bc, EVENT_NOTIFY);
#if defined(AST_MISDN_ENHANCEMENTS)
} else {
int match; /* TRUE if the dialed number matches the redirecting to number */
match = (strcmp(ast_channel_exten(ast), bc->redirecting.to.number) == 0) ? 1 : 0;
if (!bc->div_leg_3_tx_pending
|| !match) {
/* Send DivertingLegInformation1 */
bc->fac_out.Function = Fac_DivertingLegInformation1;
bc->fac_out.u.DivertingLegInformation1.InvokeID = ++misdn_invoke_id;
bc->fac_out.u.DivertingLegInformation1.DiversionReason =
misdn_to_diversion_reason(bc->redirecting.reason);
bc->fac_out.u.DivertingLegInformation1.SubscriptionOption = 2;/* notificationWithDivertedToNr */
bc->fac_out.u.DivertingLegInformation1.DivertedToPresent = 1;
misdn_PresentedNumberUnscreened_fill(&bc->fac_out.u.DivertingLegInformation1.DivertedTo, &bc->redirecting.to);
print_facility(&bc->fac_out, bc);
misdn_lib_send_event(bc, EVENT_FACILITY);
}
bc->div_leg_3_tx_pending = 0;
/* Send DivertingLegInformation3 */
bc->fac_out.Function = Fac_DivertingLegInformation3;
bc->fac_out.u.DivertingLegInformation3.InvokeID = ++misdn_invoke_id;
bc->fac_out.u.DivertingLegInformation3.PresentationAllowedIndicator =
bc->redirecting.to.presentation == 0 ? 1 : 0;
print_facility(&bc->fac_out, bc);
misdn_lib_send_event(bc, EVENT_FACILITY);
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
}
}
/*****************************/
/*** AST Indications Start ***/
/*****************************/
static int misdn_call(struct ast_channel *ast, const char *dest, int timeout)
{
int port = 0;
int r;
int exceed;
int number_type;
struct chan_list *ch;
struct misdn_bchannel *newbc;
char *dest_cp;
Enhancements to connected line and redirecting work. From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
int append_msn = 0;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(intf); /* The interface token is discarded. */
AST_APP_ARG(ext); /* extension token */
AST_APP_ARG(opts); /* options token */
);
if (!ast) {
ast_log(LOG_WARNING, " --> ! misdn_call called on ast_channel *ast where ast == NULL\n");
return -1;
}
if (((ast_channel_state(ast) != AST_STATE_DOWN) && (ast_channel_state(ast) != AST_STATE_RESERVED)) || !dest) {
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
ast_log(LOG_WARNING, " --> ! misdn_call called on %s, neither down nor reserved (or dest==NULL)\n", ast_channel_name(ast));
ast_channel_hangupcause_set(ast, AST_CAUSE_NORMAL_TEMPORARY_FAILURE);
ast_setstate(ast, AST_STATE_DOWN);
return -1;
}
ch = MISDN_ASTERISK_TECH_PVT(ast);
if (!ch) {
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
ast_log(LOG_WARNING, " --> ! misdn_call called on %s, chan_list *ch==NULL\n", ast_channel_name(ast));
ast_channel_hangupcause_set(ast, AST_CAUSE_NORMAL_TEMPORARY_FAILURE);
ast_setstate(ast, AST_STATE_DOWN);
return -1;
}
newbc = ch->bc;
if (!newbc) {
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
ast_log(LOG_WARNING, " --> ! misdn_call called on %s, newbc==NULL\n", ast_channel_name(ast));
ast_channel_hangupcause_set(ast, AST_CAUSE_NORMAL_TEMPORARY_FAILURE);
ast_setstate(ast, AST_STATE_DOWN);
return -1;
}
port = newbc->port;
#if defined(AST_MISDN_ENHANCEMENTS)
if ((ch->peer = misdn_cc_caller_get(ast))) {
chan_misdn_log(3, port, " --> Found CC caller data, peer:%s\n",
ch->peer->chan ? "available" : "NULL");
}
if (ch->record_id != -1) {
struct misdn_cc_record *cc_record;
/* This is a call completion retry call */
AST_LIST_LOCK(&misdn_cc_records_db);
cc_record = misdn_cc_find_by_id(ch->record_id);
if (!cc_record) {
AST_LIST_UNLOCK(&misdn_cc_records_db);
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
ast_log(LOG_WARNING, " --> ! misdn_call called on %s, cc_record==NULL\n", ast_channel_name(ast));
ast_channel_hangupcause_set(ast, AST_CAUSE_NORMAL_TEMPORARY_FAILURE);
ast_setstate(ast, AST_STATE_DOWN);
return -1;
}
/* Setup calling parameters to retry the call. */
newbc->dialed = cc_record->redial.dialed;
newbc->caller = cc_record->redial.caller;
memset(&newbc->redirecting, 0, sizeof(newbc->redirecting));
newbc->capability = cc_record->redial.capability;
newbc->hdlc = cc_record->redial.hdlc;
newbc->sending_complete = 1;
if (cc_record->ptp) {
newbc->fac_out.Function = Fac_CCBS_T_Call;
newbc->fac_out.u.CCBS_T_Call.InvokeID = ++misdn_invoke_id;
} else {
newbc->fac_out.Function = Fac_CCBSCall;
newbc->fac_out.u.CCBSCall.InvokeID = ++misdn_invoke_id;
newbc->fac_out.u.CCBSCall.CCBSReference = cc_record->mode.ptmp.reference_id;
}
AST_LIST_UNLOCK(&misdn_cc_records_db);
ast_channel_exten_set(ast, newbc->dialed.number);
chan_misdn_log(1, port, "* Call completion to: %s\n", newbc->dialed.number);
chan_misdn_log(2, port, " --> * tech:%s context:%s\n", ast_channel_name(ast), ast_channel_context(ast));
} else
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
{
Add private representation of caller, connected and redirecting party ids. This patch adds the feature "Private representation of caller, connected and redirecting party ids", as previously discussed with us (DATUS) and Digium. 1. Feature motivation Until now it is quite difficult to modify a party number or name which can only be seen by exactly one particular instantiated technology channel subscriber. One example where a modified party number or name on one channel is spread over several channels are supplementary services like call transfer or pickup. To implement these features Asterisk internally copies caller and connected ids from one channel to another. Another example are extension subscriptions. The monitoring entities (watchers) are notified of state changes and - if desired - of party numbers or names which represent the involving call parties. One major feature where a private representation of party names is essentially needed, i.e. where a party name shall be exclusively signaled to only one particular user, is a private user-specific name resolution for party numbers. A lookup in a private destination-dependent telephone book shall provide party names which cannot be seen by any other user at any time. 2. Feature Description This feature comes along with the implementation of additional private party id elements for caller id, connected id and redirecting ids inside Asterisk channels. The private party id elements can be read or set by the user using Asterisk dialplan functions. When a technology channel is initiating a call, receives an internal connected-line update event, or receives an internal redirecting update event, it merges the corresponding public id with the private id to create an effective party id. The effective party id is then used for protocol signaling. The channel technologies which initially support the private id representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and PRI (chan_dahdi). Once a private name or number on a channel is set and (implicitly) made valid, it is generally used for any further protocol signaling until it is rewritten or invalidated. To simplify the invalidation of private ids all internally generated connected/redirecting update events and also all connected/redirecting update events which are generated by technology channels -- receiving regarding protocol information - automatically trigger the invalidation of private ids. If not using the private party id representation feature at all, i.e. if using only the 'regular' caller-id, connected and redirecting related functions, the current characteristic of Asterisk is not affected by the new extended functionality. 3. User interface Description To grant access to the private name and number representation from the Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan functions are extended by the following data types. The formats of these data types are equal to the corresponding regular 'non-private' already existing data types: CALLERID: priv-all priv-name priv-name-valid priv-name-charset priv-name-pres priv-num priv-num-valid priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE: priv-name priv-name-valid priv-name-pres priv-name-charset priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag REDIRECTING: priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd priv-orig-tag priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd priv-to-tag Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2030/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-10 19:54:55 +00:00
struct ast_party_id connected_id = ast_channel_connected_effective_id(ast);
/*
* dest is ---v
* Dial(mISDN/g:group_name[/extension[/options]])
* Dial(mISDN/port[:preselected_channel][/extension[/options]])
*
* The dial extension could be empty if you are using MISDN_KEYPAD
* to control ISDN provider features.
*/
dest_cp = ast_strdupa(dest);
AST_NONSTANDARD_APP_ARGS(args, dest_cp, '/');
if (!args.ext) {
args.ext = "";
}
chan_misdn_log(1, port, "* CALL: %s\n", dest);
chan_misdn_log(2, port, " --> * dialed:%s tech:%s context:%s\n", args.ext, ast_channel_name(ast), ast_channel_context(ast));
ast_channel_exten_set(ast, args.ext);
ast_copy_string(newbc->dialed.number, args.ext, sizeof(newbc->dialed.number));
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
if (ast_strlen_zero(newbc->caller.name)
Add private representation of caller, connected and redirecting party ids. This patch adds the feature "Private representation of caller, connected and redirecting party ids", as previously discussed with us (DATUS) and Digium. 1. Feature motivation Until now it is quite difficult to modify a party number or name which can only be seen by exactly one particular instantiated technology channel subscriber. One example where a modified party number or name on one channel is spread over several channels are supplementary services like call transfer or pickup. To implement these features Asterisk internally copies caller and connected ids from one channel to another. Another example are extension subscriptions. The monitoring entities (watchers) are notified of state changes and - if desired - of party numbers or names which represent the involving call parties. One major feature where a private representation of party names is essentially needed, i.e. where a party name shall be exclusively signaled to only one particular user, is a private user-specific name resolution for party numbers. A lookup in a private destination-dependent telephone book shall provide party names which cannot be seen by any other user at any time. 2. Feature Description This feature comes along with the implementation of additional private party id elements for caller id, connected id and redirecting ids inside Asterisk channels. The private party id elements can be read or set by the user using Asterisk dialplan functions. When a technology channel is initiating a call, receives an internal connected-line update event, or receives an internal redirecting update event, it merges the corresponding public id with the private id to create an effective party id. The effective party id is then used for protocol signaling. The channel technologies which initially support the private id representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and PRI (chan_dahdi). Once a private name or number on a channel is set and (implicitly) made valid, it is generally used for any further protocol signaling until it is rewritten or invalidated. To simplify the invalidation of private ids all internally generated connected/redirecting update events and also all connected/redirecting update events which are generated by technology channels -- receiving regarding protocol information - automatically trigger the invalidation of private ids. If not using the private party id representation feature at all, i.e. if using only the 'regular' caller-id, connected and redirecting related functions, the current characteristic of Asterisk is not affected by the new extended functionality. 3. User interface Description To grant access to the private name and number representation from the Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan functions are extended by the following data types. The formats of these data types are equal to the corresponding regular 'non-private' already existing data types: CALLERID: priv-all priv-name priv-name-valid priv-name-charset priv-name-pres priv-num priv-num-valid priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE: priv-name priv-name-valid priv-name-pres priv-name-charset priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag REDIRECTING: priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd priv-orig-tag priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd priv-to-tag Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2030/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-10 19:54:55 +00:00
&& connected_id.name.valid
&& !ast_strlen_zero(connected_id.name.str)) {
ast_copy_string(newbc->caller.name, connected_id.name.str, sizeof(newbc->caller.name));
chan_misdn_log(3, port, " --> * set caller:\"%s\" <%s>\n", newbc->caller.name, newbc->caller.number);
}
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
if (ast_strlen_zero(newbc->caller.number)
Add private representation of caller, connected and redirecting party ids. This patch adds the feature "Private representation of caller, connected and redirecting party ids", as previously discussed with us (DATUS) and Digium. 1. Feature motivation Until now it is quite difficult to modify a party number or name which can only be seen by exactly one particular instantiated technology channel subscriber. One example where a modified party number or name on one channel is spread over several channels are supplementary services like call transfer or pickup. To implement these features Asterisk internally copies caller and connected ids from one channel to another. Another example are extension subscriptions. The monitoring entities (watchers) are notified of state changes and - if desired - of party numbers or names which represent the involving call parties. One major feature where a private representation of party names is essentially needed, i.e. where a party name shall be exclusively signaled to only one particular user, is a private user-specific name resolution for party numbers. A lookup in a private destination-dependent telephone book shall provide party names which cannot be seen by any other user at any time. 2. Feature Description This feature comes along with the implementation of additional private party id elements for caller id, connected id and redirecting ids inside Asterisk channels. The private party id elements can be read or set by the user using Asterisk dialplan functions. When a technology channel is initiating a call, receives an internal connected-line update event, or receives an internal redirecting update event, it merges the corresponding public id with the private id to create an effective party id. The effective party id is then used for protocol signaling. The channel technologies which initially support the private id representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and PRI (chan_dahdi). Once a private name or number on a channel is set and (implicitly) made valid, it is generally used for any further protocol signaling until it is rewritten or invalidated. To simplify the invalidation of private ids all internally generated connected/redirecting update events and also all connected/redirecting update events which are generated by technology channels -- receiving regarding protocol information - automatically trigger the invalidation of private ids. If not using the private party id representation feature at all, i.e. if using only the 'regular' caller-id, connected and redirecting related functions, the current characteristic of Asterisk is not affected by the new extended functionality. 3. User interface Description To grant access to the private name and number representation from the Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan functions are extended by the following data types. The formats of these data types are equal to the corresponding regular 'non-private' already existing data types: CALLERID: priv-all priv-name priv-name-valid priv-name-charset priv-name-pres priv-num priv-num-valid priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE: priv-name priv-name-valid priv-name-pres priv-name-charset priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag REDIRECTING: priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd priv-orig-tag priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd priv-to-tag Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2030/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-10 19:54:55 +00:00
&& connected_id.number.valid
&& !ast_strlen_zero(connected_id.number.str)) {
ast_copy_string(newbc->caller.number, connected_id.number.str, sizeof(newbc->caller.number));
chan_misdn_log(3, port, " --> * set caller:\"%s\" <%s>\n", newbc->caller.name, newbc->caller.number);
}
Enhancements to connected line and redirecting work. From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
misdn_cfg_get(port, MISDN_CFG_APPEND_MSN_TO_CALLERID_TAG, &append_msn, sizeof(append_msn));
if (append_msn) {
strncat(newbc->incoming_cid_tag, "_", sizeof(newbc->incoming_cid_tag) - strlen(newbc->incoming_cid_tag) - 1);
strncat(newbc->incoming_cid_tag, newbc->caller.number, sizeof(newbc->incoming_cid_tag) - strlen(newbc->incoming_cid_tag) - 1);
}
ast_channel_caller(ast)->id.tag = ast_strdup(newbc->incoming_cid_tag);
Enhancements to connected line and redirecting work. From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
misdn_cfg_get(port, MISDN_CFG_LOCALDIALPLAN, &number_type, sizeof(number_type));
if (number_type < 0) {
Add private representation of caller, connected and redirecting party ids. This patch adds the feature "Private representation of caller, connected and redirecting party ids", as previously discussed with us (DATUS) and Digium. 1. Feature motivation Until now it is quite difficult to modify a party number or name which can only be seen by exactly one particular instantiated technology channel subscriber. One example where a modified party number or name on one channel is spread over several channels are supplementary services like call transfer or pickup. To implement these features Asterisk internally copies caller and connected ids from one channel to another. Another example are extension subscriptions. The monitoring entities (watchers) are notified of state changes and - if desired - of party numbers or names which represent the involving call parties. One major feature where a private representation of party names is essentially needed, i.e. where a party name shall be exclusively signaled to only one particular user, is a private user-specific name resolution for party numbers. A lookup in a private destination-dependent telephone book shall provide party names which cannot be seen by any other user at any time. 2. Feature Description This feature comes along with the implementation of additional private party id elements for caller id, connected id and redirecting ids inside Asterisk channels. The private party id elements can be read or set by the user using Asterisk dialplan functions. When a technology channel is initiating a call, receives an internal connected-line update event, or receives an internal redirecting update event, it merges the corresponding public id with the private id to create an effective party id. The effective party id is then used for protocol signaling. The channel technologies which initially support the private id representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and PRI (chan_dahdi). Once a private name or number on a channel is set and (implicitly) made valid, it is generally used for any further protocol signaling until it is rewritten or invalidated. To simplify the invalidation of private ids all internally generated connected/redirecting update events and also all connected/redirecting update events which are generated by technology channels -- receiving regarding protocol information - automatically trigger the invalidation of private ids. If not using the private party id representation feature at all, i.e. if using only the 'regular' caller-id, connected and redirecting related functions, the current characteristic of Asterisk is not affected by the new extended functionality. 3. User interface Description To grant access to the private name and number representation from the Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan functions are extended by the following data types. The formats of these data types are equal to the corresponding regular 'non-private' already existing data types: CALLERID: priv-all priv-name priv-name-valid priv-name-charset priv-name-pres priv-num priv-num-valid priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE: priv-name priv-name-valid priv-name-pres priv-name-charset priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag REDIRECTING: priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd priv-orig-tag priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd priv-to-tag Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2030/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-10 19:54:55 +00:00
if (connected_id.number.valid) {
newbc->caller.number_type = ast_to_misdn_ton(connected_id.number.plan);
newbc->caller.number_plan = ast_to_misdn_plan(connected_id.number.plan);
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
} else {
newbc->caller.number_type = NUMTYPE_UNKNOWN;
newbc->caller.number_plan = NUMPLAN_ISDN;
}
} else {
/* Force us to send in SETUP message */
newbc->caller.number_type = number_type;
newbc->caller.number_plan = NUMPLAN_ISDN;
}
debug_numtype(port, newbc->caller.number_type, "LTON");
newbc->capability = ast_channel_transfercapability(ast);
pbx_builtin_setvar_helper(ast, "TRANSFERCAPABILITY", ast_transfercapability2str(newbc->capability));
if (ast_channel_transfercapability(ast) == INFO_CAPABILITY_DIGITAL_UNRESTRICTED) {
chan_misdn_log(2, port, " --> * Call with flag Digital\n");
}
Merged revisions 49313 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r49313 | crichter | 2007-01-03 10:06:50 +0100 (Mi, 03 Jan 2007) | 41 lines Merged revisions 48319,48321,48467,48552,48576,49135,49303 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) | 1 line changed a few debugs to higher debug levels ........ r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) | 1 line added the export and import of the MISDN_ADDRESS_COMPLETE Variable to inidcate wether the extension is already completely dialed or if there might come additional digits by information elements. also added some docs for that. ........ r48467 | crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line removed FIXUP state. added check for channel allocation conflict when we create a setup while the other site creates a setup on the same channel, besides the check we resolve this conflict. ........ r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18 Dez 2006) | 1 line when our PTP Partner sends us a SETUP with a preselected channel we just accept it, even when we're NT. added some checks for segfaults. ........ r48576 | crichter | 2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line when we reject a channel, because it's in use already, we shouldn't process the setup anymore. made the channel allocation a bit easier and more understandable, removed a few unused lines ........ r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02 Jan 2007) | 1 line added check for channel ranges in the set/empty channel functions. set pmp_l1_check default to no. added misdn restart pid cli command. added cleaning of channel when we send a RELEASE_COMPLETE. ........ r49303 | crichter | 2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines * Added check for bridging in misdn_call to avoid setting echocancellation when 2 mISDN channels are involved and when bridging is set. That lead to a kernel panic before under different situations, because we switched about 2 times between hardware bridging and echocancelation * readded MISDN_URATE variable which got lost before, this should make app_v110 work again * fixed typo ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49321 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-03 11:15:02 +00:00
/* update caller screening and presentation */
update_config(ch);
/* fill in some ies from channel dialplan variables */
import_ch(ast, newbc, ch);
/* Finally The Options Override Everything */
if (!ast_strlen_zero(args.opts)) {
misdn_set_opt_exec(ast, args.opts);
} else {
chan_misdn_log(2, port, "NO OPTS GIVEN\n");
}
if (newbc->set_presentation) {
newbc->caller.presentation = newbc->presentation;
}
misdn_copy_redirecting_from_ast(newbc, ast);
switch (newbc->outgoing_colp) {
case 1:/* restricted */
case 2:/* blocked */
newbc->redirecting.from.presentation = 1;/* restricted */
break;
default:
break;
}
#if defined(AST_MISDN_ENHANCEMENTS)
if (newbc->redirecting.from.number[0] && misdn_lib_is_ptp(port)) {
if (newbc->redirecting.count < 1) {
newbc->redirecting.count = 1;
}
/* Create DivertingLegInformation2 facility */
newbc->fac_out.Function = Fac_DivertingLegInformation2;
newbc->fac_out.u.DivertingLegInformation2.InvokeID = ++misdn_invoke_id;
newbc->fac_out.u.DivertingLegInformation2.DivertingPresent = 1;
misdn_PresentedNumberUnscreened_fill(
&newbc->fac_out.u.DivertingLegInformation2.Diverting,
&newbc->redirecting.from);
switch (newbc->outgoing_colp) {
case 2:/* blocked */
/* Block the number going out */
newbc->fac_out.u.DivertingLegInformation2.Diverting.Type = 1;/* presentationRestricted */
/* Don't tell about any previous diversions or why for that matter. */
newbc->fac_out.u.DivertingLegInformation2.DiversionCounter = 1;
newbc->fac_out.u.DivertingLegInformation2.DiversionReason = 0;/* unknown */
break;
default:
newbc->fac_out.u.DivertingLegInformation2.DiversionCounter =
newbc->redirecting.count;
newbc->fac_out.u.DivertingLegInformation2.DiversionReason =
misdn_to_diversion_reason(newbc->redirecting.reason);
break;
}
newbc->fac_out.u.DivertingLegInformation2.OriginalCalledPresent = 0;
if (1 < newbc->fac_out.u.DivertingLegInformation2.DiversionCounter) {
newbc->fac_out.u.DivertingLegInformation2.OriginalCalledPresent = 1;
newbc->fac_out.u.DivertingLegInformation2.OriginalCalled.Type = 2;/* numberNotAvailableDueToInterworking */
}
/*
* Expect a DivertingLegInformation3 to update the COLR of the
* redirecting-to party we are attempting to call now.
*/
newbc->div_leg_3_rx_wanted = 1;
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
}
exceed = add_out_calls(port);
if (exceed != 0) {
char tmp[16];
snprintf(tmp, sizeof(tmp), "%d", exceed);
pbx_builtin_setvar_helper(ast, "MAX_OVERFLOW", tmp);
ast_channel_hangupcause_set(ast, AST_CAUSE_NORMAL_TEMPORARY_FAILURE);
ast_setstate(ast, AST_STATE_DOWN);
return -1;
}
#if defined(AST_MISDN_ENHANCEMENTS)
if (newbc->fac_out.Function != Fac_None) {
print_facility(&newbc->fac_out, newbc);
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
r = misdn_lib_send_event(newbc, EVENT_SETUP);
/** we should have l3id after sending setup **/
ch->l3id = newbc->l3_id;
if (r == -ENOCHAN) {
chan_misdn_log(0, port, " --> * Theres no Channel at the moment .. !\n");
chan_misdn_log(1, port, " --> * SEND: State Down pid:%d\n", newbc ? newbc->pid : -1);
ast_channel_hangupcause_set(ast, AST_CAUSE_NORMAL_CIRCUIT_CONGESTION);
ast_setstate(ast, AST_STATE_DOWN);
return -1;
}
chan_misdn_log(2, port, " --> * SEND: State Dialing pid:%d\n", newbc ? newbc->pid : 1);
ast_setstate(ast, AST_STATE_DIALING);
ast_channel_hangupcause_set(ast, AST_CAUSE_NORMAL_CLEARING);
if (newbc->nt) {
stop_bc_tones(ch);
}
ch->state = MISDN_CALLING;
return 0;
}
static int misdn_answer(struct ast_channel *ast)
{
struct chan_list *p;
const char *tmp;
if (!ast || !(p = MISDN_ASTERISK_TECH_PVT(ast))) {
return -1;
}
chan_misdn_log(1, p ? (p->bc ? p->bc->port : 0) : 0, "* ANSWER:\n");
if (!p) {
ast_log(LOG_WARNING, " --> Channel not connected ??\n");
ast_queue_hangup_with_cause(ast, AST_CAUSE_NETWORK_OUT_OF_ORDER);
}
if (!p->bc) {
chan_misdn_log(1, 0, " --> Got Answer, but there is no bc obj ??\n");
ast_queue_hangup_with_cause(ast, AST_CAUSE_PROTOCOL_ERROR);
}
ast_channel_lock(ast);
tmp = pbx_builtin_getvar_helper(ast, "CRYPT_KEY");
if (!ast_strlen_zero(tmp)) {
chan_misdn_log(1, p->bc->port, " --> Connection will be BF crypted\n");
ast_copy_string(p->bc->crypt_key, tmp, sizeof(p->bc->crypt_key));
} else {
chan_misdn_log(3, p->bc->port, " --> Connection is without BF encryption\n");
}
tmp = pbx_builtin_getvar_helper(ast, "MISDN_DIGITAL_TRANS");
if (!ast_strlen_zero(tmp) && ast_true(tmp)) {
chan_misdn_log(1, p->bc->port, " --> Connection is transparent digital\n");
p->bc->nodsp = 1;
p->bc->hdlc = 0;
p->bc->nojitter = 1;
}
ast_channel_unlock(ast);
p->state = MISDN_CONNECTED;
stop_indicate(p);
if (ast_strlen_zero(p->bc->connected.number)) {
chan_misdn_log(2,p->bc->port," --> empty connected number using dialed number\n");
ast_copy_string(p->bc->connected.number, p->bc->dialed.number, sizeof(p->bc->connected.number));
/*
* Use the misdn_set_opt() application to set the presentation
* before we answer or you can use the CONECTEDLINE() function
* to set everything before using the Answer() application.
*/
p->bc->connected.presentation = p->bc->presentation;
p->bc->connected.screening = 0; /* unscreened */
p->bc->connected.number_type = p->bc->dialed.number_type;
p->bc->connected.number_plan = p->bc->dialed.number_plan;
}
switch (p->bc->outgoing_colp) {
case 1:/* restricted */
case 2:/* blocked */
p->bc->connected.presentation = 1;/* restricted */
break;
default:
break;
}
#if defined(AST_MISDN_ENHANCEMENTS)
if (p->bc->div_leg_3_tx_pending) {
p->bc->div_leg_3_tx_pending = 0;
/* Send DivertingLegInformation3 */
p->bc->fac_out.Function = Fac_DivertingLegInformation3;
p->bc->fac_out.u.DivertingLegInformation3.InvokeID = ++misdn_invoke_id;
p->bc->fac_out.u.DivertingLegInformation3.PresentationAllowedIndicator =
(p->bc->connected.presentation == 0) ? 1 : 0;
print_facility(&p->bc->fac_out, p->bc);
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
misdn_lib_send_event(p->bc, EVENT_CONNECT);
start_bc_tones(p);
return 0;
}
static int misdn_digit_begin(struct ast_channel *chan, char digit)
{
/* XXX Modify this callback to support Asterisk controlling the length of DTMF */
return 0;
}
2007-01-19 18:06:03 +00:00
static int misdn_digit_end(struct ast_channel *ast, char digit, unsigned int duration)
{
struct chan_list *p;
struct misdn_bchannel *bc;
char buf[2] = { digit, 0 };
if (!ast || !(p = MISDN_ASTERISK_TECH_PVT(ast))) {
return -1;
}
bc = p->bc;
chan_misdn_log(1, bc ? bc->port : 0, "* IND : Digit %c\n", digit);
if (!bc) {
ast_log(LOG_WARNING, " --> !! Got Digit Event without having bchannel Object\n");
return -1;
}
switch (p->state) {
case MISDN_CALLING:
if (strlen(bc->infos_pending) < sizeof(bc->infos_pending) - 1) {
strncat(bc->infos_pending, buf, sizeof(bc->infos_pending) - strlen(bc->infos_pending) - 1);
}
break;
case MISDN_CALLING_ACKNOWLEDGE:
ast_copy_string(bc->info_dad, buf, sizeof(bc->info_dad));
if (strlen(bc->dialed.number) < sizeof(bc->dialed.number) - 1) {
strncat(bc->dialed.number, buf, sizeof(bc->dialed.number) - strlen(bc->dialed.number) - 1);
}
ast_channel_exten_set(p->ast, bc->dialed.number);
misdn_lib_send_event(bc, EVENT_INFORMATION);
break;
default:
if (bc->send_dtmf) {
send_digit_to_chan(p, digit);
}
break;
}
return 0;
}
static int misdn_fixup(struct ast_channel *oldast, struct ast_channel *ast)
{
struct chan_list *p;
if (!ast || !(p = MISDN_ASTERISK_TECH_PVT(ast))) {
return -1;
}
chan_misdn_log(1, p->bc ? p->bc->port : 0, "* IND: Got Fixup State:%s L3id:%x\n", misdn_get_ch_state(p), p->l3id);
p->ast = ast;
return 0;
}
static int misdn_indication(struct ast_channel *ast, int cond, const void *data, size_t datalen)
{
struct chan_list *p;
if (!ast || !(p = MISDN_ASTERISK_TECH_PVT(ast))) {
ast_log(LOG_WARNING, "Returned -1 in misdn_indication\n");
return -1;
}
if (!p->bc) {
if (p->hold.state == MISDN_HOLD_IDLE) {
chan_misdn_log(1, 0, "* IND : Indication [%d] ignored on %s\n", cond,
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
ast_channel_name(ast));
ast_log(LOG_WARNING, "Private Pointer but no bc ?\n");
} else {
chan_misdn_log(1, 0, "* IND : Indication [%d] ignored on hold %s\n",
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
cond, ast_channel_name(ast));
}
return -1;
}
Merged revisions 374515-374535 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ................ r374515 | rmudgett | 2012-10-04 17:52:36 -0500 (Thu, 04 Oct 2012) | 10 lines chan_misdn: Remove some deadcode * Made setup_bc() static. Patches: patch1_unused-code.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882 ................ r374516 | rmudgett | 2012-10-04 18:01:01 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused bchan states Patches: patch2_unused-states.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374517 | rmudgett | 2012-10-04 18:17:51 -0500 (Thu, 04 Oct 2012) | 16 lines chan_misdn: Remove unnecessary null pointer checks and checks for stack->nt * cleanup_bc() is always called with valid bc (or it would've crashed before). * Value of stack->nt is known in advance at some places. * Rename handle_event() to handle_event_te(), handle_frm() to handle_frm_te(). Patches: patch3_checks.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882 ................ r374518 | rmudgett | 2012-10-04 18:21:59 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Fix spelling in log messages Patches: patch4_spelling.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374519 | rmudgett | 2012-10-04 18:31:59 -0500 (Thu, 04 Oct 2012) | 15 lines chan_misdn: Don't cleanup a bc twice. In handle_frm_te() after calling misdn_lib_send_event(bc, EVENT_RELEASE_COMPLETE) bc is emptied, cleaned and set not in use, although misdn_lib_send_event() already did the same. This is bad. When it's not in use we are not allowed to touch it. * Moved log message in front of the resulting actions and fixed it to match the case. Patches: patch5_bccleanup.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374520 | rmudgett | 2012-10-04 18:43:56 -0500 (Thu, 04 Oct 2012) | 12 lines chan_misdn: Fix memory leaks, bc, chan not cleaned up etc., really bad stuff. * Fix return codes of cb_events() for EVENT_SETUP to use caller's cleanup mechanisms. * Move cl_queue_chan() call after bearer check. Patches: patch6_leaks.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374521 | rmudgett | 2012-10-04 18:48:38 -0500 (Thu, 04 Oct 2012) | 11 lines chan_misdn: We must initialize cause on sending a DISCONNECT. We must initialize cause on sending a DISCONNECT, so it is later correctly indicated to ast_channel in case the answer (RELEASE/RELEASE_COMPLETE) does not include one. Patches: patch7_hangupcause.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374522 | rmudgett | 2012-10-04 19:03:56 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused code for upqueue Patches: patch8_unused-upqueue.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374523 | rmudgett | 2012-10-04 19:11:50 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Improve debugging (port number, messages fixed, dups removed) Patches: patch9_debug.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374533 | rmudgett | 2012-10-05 12:17:18 -0500 (Fri, 05 Oct 2012) | 8 lines chan_misdn: Better debug: we can print_bc_info even if there's no ast leg. Patches: patch10_debug-bc-2.diff (license #6372) patch uploaded by Guenther Kelleter Modified. JIRA ABE-2882 ................ r374534 | rmudgett | 2012-10-05 12:34:10 -0500 (Fri, 05 Oct 2012) | 16 lines chan_misdn: setup_bc() is called too early for an incoming SETUP on TE. This prevents the B channel from being setup for HDLC mode when requested by the bearer capability and config option hdlc=yes. It violates ETS300102 Ch.5.2.3.2: "The user, in any case, must not connect to the channel until a CONNECT ACKNOWLEDGE message has been received." * Call setup_bc() on receipt of CONNECT_ACKNOWLEGDE for PTMP, and on first response to SETUP for PTP. Patches: abe-2881-2.diff (license #6372) patch uploaded by Guenther Kelleter Modified. JIRA ABE-2881 ................ r374535 | rmudgett | 2012-10-05 12:41:05 -0500 (Fri, 05 Oct 2012) | 2 lines chan_misdn: Remove some more deadcode. ................ ........ Merged revisions 374536 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374537 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374538 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374539 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-05 18:42:14 +00:00
chan_misdn_log(5, p->bc->port, "* IND : Indication [%d] on %s\n", cond, ast_channel_name(ast));
switch (cond) {
case AST_CONTROL_BUSY:
chan_misdn_log(1, p->bc->port, "* IND :\tbusy pid:%d\n", p->bc->pid);
ast_setstate(ast, AST_STATE_BUSY);
p->bc->out_cause = AST_CAUSE_USER_BUSY;
if (p->state != MISDN_CONNECTED) {
start_bc_tones(p);
misdn_lib_send_event(p->bc, EVENT_DISCONNECT);
}
return -1;
case AST_CONTROL_RING:
chan_misdn_log(1, p->bc->port, "* IND :\tring pid:%d\n", p->bc->pid);
return -1;
case AST_CONTROL_RINGING:
chan_misdn_log(1, p->bc->port, "* IND :\tringing pid:%d\n", p->bc->pid);
switch (p->state) {
case MISDN_ALERTING:
chan_misdn_log(2, p->bc->port, " --> * IND :\tringing pid:%d but I was Ringing before, so ignoring it\n", p->bc->pid);
break;
case MISDN_CONNECTED:
chan_misdn_log(2, p->bc->port, " --> * IND :\tringing pid:%d but Connected, so just send TONE_ALERTING without state changes \n", p->bc->pid);
return -1;
default:
p->state = MISDN_ALERTING;
chan_misdn_log(2, p->bc->port, " --> * IND :\tringing pid:%d\n", p->bc->pid);
misdn_lib_send_event(p->bc, EVENT_ALERTING);
chan_misdn_log(3, p->bc->port, " --> * SEND: State Ring pid:%d\n", p->bc->pid);
ast_setstate(ast, AST_STATE_RING);
if (!p->bc->nt && (p->originator == ORG_MISDN) && !p->incoming_early_audio) {
chan_misdn_log(2, p->bc->port, " --> incoming_early_audio off\n");
} else {
return -1;
}
}
break;
case AST_CONTROL_ANSWER:
chan_misdn_log(1, p->bc->port, " --> * IND :\tanswer pid:%d\n", p->bc->pid);
start_bc_tones(p);
break;
case AST_CONTROL_TAKEOFFHOOK:
chan_misdn_log(1, p->bc->port, " --> *\ttakeoffhook pid:%d\n", p->bc->pid);
return -1;
case AST_CONTROL_OFFHOOK:
chan_misdn_log(1, p->bc->port, " --> *\toffhook pid:%d\n", p->bc->pid);
return -1;
case AST_CONTROL_FLASH:
chan_misdn_log(1, p->bc->port, " --> *\tflash pid:%d\n", p->bc->pid);
break;
case AST_CONTROL_PROGRESS:
chan_misdn_log(1, p->bc->port, " --> * IND :\tprogress pid:%d\n", p->bc->pid);
misdn_lib_send_event(p->bc, EVENT_PROGRESS);
break;
case AST_CONTROL_PROCEEDING:
chan_misdn_log(1, p->bc->port, " --> * IND :\tproceeding pid:%d\n", p->bc->pid);
misdn_lib_send_event(p->bc, EVENT_PROCEEDING);
break;
Merged revisions 335078 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335078 | mjordan | 2011-09-09 11:27:01 -0500 (Fri, 09 Sep 2011) | 29 lines Merged revisions 335064 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335064 | mjordan | 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011) | 23 lines Updated SIP 484 handling; added Incomplete control frame When a SIP phone uses the dial application and receives a 484 Address Incomplete response, if overlapped dialing is enabled for SIP, then the 484 Address Incomplete is forwarded back to the SIP phone and the HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete application dialplan logic was automatically triggered; now, explicit dialplan usage of the application is required. Additionally, this patch adds a new AST_CONTOL_FRAME type called AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame, it is an indication that the dialplan expects more digits back from the device. If the device supports overlap dialing it should attempt to notify the device that the dialplan is waiting for more digits; otherwise, it can handle the frame in a manner appropriate to the channel driver. (closes issue ASTERISK-17288) Reported by: Mikael Carlsson Tested by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/1416/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335079 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-09 16:28:23 +00:00
case AST_CONTROL_INCOMPLETE:
chan_misdn_log(1, p->bc->port, " --> *\tincomplete pid:%d\n", p->bc->pid);
if (!p->overlap_dial) {
/* Overlapped dialing not enabled - send hangup */
p->bc->out_cause = AST_CAUSE_INVALID_NUMBER_FORMAT;
start_bc_tones(p);
misdn_lib_send_event(p->bc, EVENT_DISCONNECT);
if (p->bc->nt) {
hanguptone_indicate(p);
}
}
break;
case AST_CONTROL_CONGESTION:
chan_misdn_log(1, p->bc->port, " --> * IND :\tcongestion pid:%d\n", p->bc->pid);
p->bc->out_cause = AST_CAUSE_SWITCH_CONGESTION;
start_bc_tones(p);
misdn_lib_send_event(p->bc, EVENT_DISCONNECT);
if (p->bc->nt) {
hanguptone_indicate(p);
}
break;
case -1 :
chan_misdn_log(1, p->bc->port, " --> * IND :\t-1! (stop indication) pid:%d\n", p->bc->pid);
stop_indicate(p);
if (p->state == MISDN_CONNECTED) {
start_bc_tones(p);
}
break;
case AST_CONTROL_HOLD:
ast_moh_start(ast, data, p->mohinterpret);
chan_misdn_log(1, p->bc->port, " --> *\tHOLD pid:%d\n", p->bc->pid);
break;
case AST_CONTROL_UNHOLD:
ast_moh_stop(ast);
chan_misdn_log(1, p->bc->port, " --> *\tUNHOLD pid:%d\n", p->bc->pid);
break;
case AST_CONTROL_CONNECTED_LINE:
chan_misdn_log(1, p->bc->port, "* IND :\tconnected line update pid:%d\n", p->bc->pid);
misdn_update_connected_line(ast, p->bc, p->originator);
break;
case AST_CONTROL_REDIRECTING:
chan_misdn_log(1, p->bc->port, "* IND :\tredirecting info update pid:%d\n", p->bc->pid);
misdn_update_redirecting(ast, p->bc, p->originator);
break;
default:
chan_misdn_log(1, p->bc->port, " --> * Unknown Indication:%d pid:%d\n", cond, p->bc->pid);
/* fallthrough */
case AST_CONTROL_PVT_CAUSE_CODE:
return -1;
}
return 0;
}
static int misdn_hangup(struct ast_channel *ast)
{
struct chan_list *p;
struct misdn_bchannel *bc;
const char *var;
if (!ast) {
return -1;
}
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
ast_debug(1, "misdn_hangup(%s)\n", ast_channel_name(ast));
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
/* Take the ast_channel's tech_pvt reference. */
ast_mutex_lock(&release_lock);
p = MISDN_ASTERISK_TECH_PVT(ast);
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
if (!p) {
ast_mutex_unlock(&release_lock);
return -1;
}
MISDN_ASTERISK_TECH_PVT_SET(ast, NULL);
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
if (!misdn_chan_is_valid(p)) {
ast_mutex_unlock(&release_lock);
chan_list_unref(p, "Release ast_channel reference. Was not active?");
return 0;
}
if (p->hold.state == MISDN_HOLD_IDLE) {
bc = p->bc;
} else {
p->hold.state = MISDN_HOLD_DISCONNECT;
bc = misdn_lib_find_held_bc(p->hold.port, p->l3id);
if (!bc) {
chan_misdn_log(4, p->hold.port,
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
"misdn_hangup: Could not find held bc for (%s)\n", ast_channel_name(ast));
release_chan_early(p);
ast_mutex_unlock(&release_lock);
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
chan_list_unref(p, "Release ast_channel reference");
return 0;
}
}
if (ast_channel_state(ast) == AST_STATE_RESERVED || p->state == MISDN_NOTHING) {
/* between request and call */
ast_debug(1, "State Reserved (or nothing) => chanIsAvail\n");
release_chan_early(p);
if (bc) {
misdn_lib_release(bc);
}
ast_mutex_unlock(&release_lock);
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
chan_list_unref(p, "Release ast_channel reference");
return 0;
}
if (!bc) {
ast_log(LOG_WARNING, "Hangup with private but no bc ? state:%s l3id:%x\n",
misdn_get_ch_state(p), p->l3id);
release_chan_early(p);
ast_mutex_unlock(&release_lock);
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
chan_list_unref(p, "Release ast_channel reference");
return 0;
}
p->ast = NULL;
p->need_hangup = 0;
p->need_queue_hangup = 0;
p->need_busy = 0;
if (!bc->nt) {
stop_bc_tones(p);
}
bc->out_cause = ast_channel_hangupcause(ast) ? ast_channel_hangupcause(ast) : AST_CAUSE_NORMAL_CLEARING;
/* Channel lock is already held when we are called. */
//ast_channel_lock(ast);
var = pbx_builtin_getvar_helper(ast, "HANGUPCAUSE");
if (!var) {
var = pbx_builtin_getvar_helper(ast, "PRI_CAUSE");
}
if (var) {
int tmpcause;
tmpcause = atoi(var);
bc->out_cause = tmpcause ? tmpcause : AST_CAUSE_NORMAL_CLEARING;
}
var = pbx_builtin_getvar_helper(ast, "MISDN_USERUSER");
if (var) {
ast_log(LOG_NOTICE, "MISDN_USERUSER: %s\n", var);
ast_copy_string(bc->uu, var, sizeof(bc->uu));
bc->uulen = strlen(bc->uu);
}
//ast_channel_unlock(ast);
chan_misdn_log(1, bc->port,
"* IND : HANGUP\tpid:%d context:%s dialed:%s caller:\"%s\" <%s> State:%s\n",
bc->pid,
ast_channel_context(ast),
ast_channel_exten(ast),
(ast_channel_caller(ast)->id.name.valid && ast_channel_caller(ast)->id.name.str)
? ast_channel_caller(ast)->id.name.str : "",
(ast_channel_caller(ast)->id.number.valid && ast_channel_caller(ast)->id.number.str)
? ast_channel_caller(ast)->id.number.str : "",
misdn_get_ch_state(p));
chan_misdn_log(3, bc->port, " --> l3id:%x\n", p->l3id);
chan_misdn_log(3, bc->port, " --> cause:%d\n", bc->cause);
chan_misdn_log(2, bc->port, " --> out_cause:%d\n", bc->out_cause);
switch (p->state) {
case MISDN_INCOMING_SETUP:
/*
* This is the only place in misdn_hangup, where we
* can call release_chan, else it might create a lot of trouble.
*/
ast_log(LOG_NOTICE, "release channel, in INCOMING_SETUP state.. no other events happened\n");
release_chan(p, bc);
misdn_lib_send_event(bc, EVENT_RELEASE_COMPLETE);
ast_mutex_unlock(&release_lock);
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
chan_list_unref(p, "Release ast_channel reference");
return 0;
case MISDN_DIALING:
if (p->hold.state == MISDN_HOLD_IDLE) {
start_bc_tones(p);
hanguptone_indicate(p);
}
if (bc->need_disconnect) {
misdn_lib_send_event(bc, EVENT_DISCONNECT);
}
break;
case MISDN_CALLING_ACKNOWLEDGE:
if (p->hold.state == MISDN_HOLD_IDLE) {
start_bc_tones(p);
hanguptone_indicate(p);
}
if (bc->need_disconnect) {
misdn_lib_send_event(bc, EVENT_DISCONNECT);
}
break;
case MISDN_CALLING:
case MISDN_ALERTING:
case MISDN_PROGRESS:
case MISDN_PROCEEDING:
if (p->originator != ORG_AST && p->hold.state == MISDN_HOLD_IDLE) {
hanguptone_indicate(p);
}
if (bc->need_disconnect) {
misdn_lib_send_event(bc, EVENT_DISCONNECT);
}
break;
case MISDN_CONNECTED:
/* Alerting or Disconnect */
if (bc->nt && p->hold.state == MISDN_HOLD_IDLE) {
start_bc_tones(p);
hanguptone_indicate(p);
bc->progress_indicator = INFO_PI_INBAND_AVAILABLE;
}
if (bc->need_disconnect) {
misdn_lib_send_event(bc, EVENT_DISCONNECT);
}
break;
case MISDN_DISCONNECTED:
if (bc->need_release) {
misdn_lib_send_event(bc, EVENT_RELEASE);
}
break;
case MISDN_CLEANING:
ast_mutex_unlock(&release_lock);
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
chan_list_unref(p, "Release ast_channel reference");
return 0;
case MISDN_BUSY:
break;
default:
if (bc->nt) {
bc->out_cause = -1;
if (bc->need_release) {
misdn_lib_send_event(bc, EVENT_RELEASE);
}
} else {
if (bc->need_disconnect) {
misdn_lib_send_event(bc, EVENT_DISCONNECT);
}
}
break;
}
p->state = MISDN_CLEANING;
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
chan_misdn_log(3, bc->port, " --> Channel: %s hungup new state:%s\n", ast_channel_name(ast),
misdn_get_ch_state(p));
ast_mutex_unlock(&release_lock);
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
chan_list_unref(p, "Release ast_channel reference");
return 0;
}
static struct ast_frame *process_ast_dsp(struct chan_list *tmp, struct ast_frame *frame)
{
Merged revisions 247910 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r247910 | rmudgett | 2010-02-19 11:18:49 -0600 (Fri, 19 Feb 2010) | 55 lines Merged revision 247904 from https://origsvn.digium.com/svn/asterisk/be/branches/C.2-... .......... r247904 | rmudgett | 2010-02-19 10:49:44 -0600 (Fri, 19 Feb 2010) | 49 lines Make chan_misdn DTMF processing consistent with other channel technologies. The processing of DTMF tones on the receiving side of an ISDN channel is inconsistent with the way it is handled in other channels, especially DAHDI analog. This causes DTMF tones sent from an ISDN phone to be doubled at the connected party. We are using the following 2 options of misdn.conf 1) astdtmf=yes 2) senddtmf=yes Option one is necessary because the asterisk DSP DTMF detection is better than mISDN's internal DSP. Not as many false positives. Option two is necessary to transmit DTMF tones end to end when mISDN channels are connected to SIP channels with out of band DTMF for example. The symptom is that DTMF tones sent by an ISDN phone are doubled on the way through asterisk when two mISDN channels are connected with a Local channel in between or if it is bridged to an analog channel. The doubling of DTMF tones is because DTMF is passed inband to asterisk by the mISDN channel and passed out of band once again after the release of the DTMF tone. Passing it inband is wrong. Neither an analog channel nor SIP channel passes DTMF inband if configured to inband DTMF. Analog and SIP channels filter out the DTMF tones because they use the voice frames returned by ast_dsp_process. But chan_misdn passes the unfiltered input voice frames instead. To overcome one aspect of the problem, the doubling of DTMF tones when two mISDN channels are directly bridged, someone made an 'optimization', where in that case the DTMF tone passed out-of-band to the peer channel is not translated to an inband tone at the transmit side. This optimization is bad because it does not work in general. For example, analog channels or mISDN channels when bridged through an intermediary local channel will generate DTMF tones from out-of-band information. Also, of course, it must not be done when there is no inband DTMF available. This patch fixes the issue. Now chan_misdn will filter the received inband DTMF signal the same as other channel types. Another change included: No need to build an extra translation path because ast_process_dsp does it if required. Patches: misdn-dtmf.patch JIRA ABE-2080 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247914 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-19 17:33:33 +00:00
struct ast_frame *f;
if (tmp->dsp) {
f = ast_dsp_process(tmp->ast, tmp->dsp, frame);
} else {
Merged revisions 247910 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r247910 | rmudgett | 2010-02-19 11:18:49 -0600 (Fri, 19 Feb 2010) | 55 lines Merged revision 247904 from https://origsvn.digium.com/svn/asterisk/be/branches/C.2-... .......... r247904 | rmudgett | 2010-02-19 10:49:44 -0600 (Fri, 19 Feb 2010) | 49 lines Make chan_misdn DTMF processing consistent with other channel technologies. The processing of DTMF tones on the receiving side of an ISDN channel is inconsistent with the way it is handled in other channels, especially DAHDI analog. This causes DTMF tones sent from an ISDN phone to be doubled at the connected party. We are using the following 2 options of misdn.conf 1) astdtmf=yes 2) senddtmf=yes Option one is necessary because the asterisk DSP DTMF detection is better than mISDN's internal DSP. Not as many false positives. Option two is necessary to transmit DTMF tones end to end when mISDN channels are connected to SIP channels with out of band DTMF for example. The symptom is that DTMF tones sent by an ISDN phone are doubled on the way through asterisk when two mISDN channels are connected with a Local channel in between or if it is bridged to an analog channel. The doubling of DTMF tones is because DTMF is passed inband to asterisk by the mISDN channel and passed out of band once again after the release of the DTMF tone. Passing it inband is wrong. Neither an analog channel nor SIP channel passes DTMF inband if configured to inband DTMF. Analog and SIP channels filter out the DTMF tones because they use the voice frames returned by ast_dsp_process. But chan_misdn passes the unfiltered input voice frames instead. To overcome one aspect of the problem, the doubling of DTMF tones when two mISDN channels are directly bridged, someone made an 'optimization', where in that case the DTMF tone passed out-of-band to the peer channel is not translated to an inband tone at the transmit side. This optimization is bad because it does not work in general. For example, analog channels or mISDN channels when bridged through an intermediary local channel will generate DTMF tones from out-of-band information. Also, of course, it must not be done when there is no inband DTMF available. This patch fixes the issue. Now chan_misdn will filter the received inband DTMF signal the same as other channel types. Another change included: No need to build an extra translation path because ast_process_dsp does it if required. Patches: misdn-dtmf.patch JIRA ABE-2080 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247914 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-19 17:33:33 +00:00
chan_misdn_log(0, tmp->bc->port, "No DSP-Path found\n");
return NULL;
}
if (!f || (f->frametype != AST_FRAME_DTMF)) {
Merged revisions 247910 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r247910 | rmudgett | 2010-02-19 11:18:49 -0600 (Fri, 19 Feb 2010) | 55 lines Merged revision 247904 from https://origsvn.digium.com/svn/asterisk/be/branches/C.2-... .......... r247904 | rmudgett | 2010-02-19 10:49:44 -0600 (Fri, 19 Feb 2010) | 49 lines Make chan_misdn DTMF processing consistent with other channel technologies. The processing of DTMF tones on the receiving side of an ISDN channel is inconsistent with the way it is handled in other channels, especially DAHDI analog. This causes DTMF tones sent from an ISDN phone to be doubled at the connected party. We are using the following 2 options of misdn.conf 1) astdtmf=yes 2) senddtmf=yes Option one is necessary because the asterisk DSP DTMF detection is better than mISDN's internal DSP. Not as many false positives. Option two is necessary to transmit DTMF tones end to end when mISDN channels are connected to SIP channels with out of band DTMF for example. The symptom is that DTMF tones sent by an ISDN phone are doubled on the way through asterisk when two mISDN channels are connected with a Local channel in between or if it is bridged to an analog channel. The doubling of DTMF tones is because DTMF is passed inband to asterisk by the mISDN channel and passed out of band once again after the release of the DTMF tone. Passing it inband is wrong. Neither an analog channel nor SIP channel passes DTMF inband if configured to inband DTMF. Analog and SIP channels filter out the DTMF tones because they use the voice frames returned by ast_dsp_process. But chan_misdn passes the unfiltered input voice frames instead. To overcome one aspect of the problem, the doubling of DTMF tones when two mISDN channels are directly bridged, someone made an 'optimization', where in that case the DTMF tone passed out-of-band to the peer channel is not translated to an inband tone at the transmit side. This optimization is bad because it does not work in general. For example, analog channels or mISDN channels when bridged through an intermediary local channel will generate DTMF tones from out-of-band information. Also, of course, it must not be done when there is no inband DTMF available. This patch fixes the issue. Now chan_misdn will filter the received inband DTMF signal the same as other channel types. Another change included: No need to build an extra translation path because ast_process_dsp does it if required. Patches: misdn-dtmf.patch JIRA ABE-2080 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247914 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-19 17:33:33 +00:00
return f;
}
ast_debug(1, "Detected inband DTMF digit: %c\n", f->subclass.integer);
if (tmp->faxdetect && (f->subclass.integer == 'f')) {
/* Fax tone -- Handle and return NULL */
if (!tmp->faxhandled) {
struct ast_channel *ast = tmp->ast;
tmp->faxhandled++;
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
chan_misdn_log(0, tmp->bc->port, "Fax detected, preparing %s for fax transfer.\n", ast_channel_name(ast));
tmp->bc->rxgain = 0;
isdn_lib_update_rxgain(tmp->bc);
tmp->bc->txgain = 0;
isdn_lib_update_txgain(tmp->bc);
#ifdef MISDN_1_2
*tmp->bc->pipeline = 0;
#else
tmp->bc->ec_enable = 0;
#endif
isdn_lib_update_ec(tmp->bc);
isdn_lib_stop_dtmf(tmp->bc);
switch (tmp->faxdetect) {
case 1:
if (strcmp(ast_channel_exten(ast), "fax")) {
const char *context;
char context_tmp[BUFFERSIZE];
misdn_cfg_get(tmp->bc->port, MISDN_CFG_FAXDETECT_CONTEXT, &context_tmp, sizeof(context_tmp));
context = S_OR(context_tmp, S_OR(ast_channel_macrocontext(ast), ast_channel_context(ast)));
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
if (ast_exists_extension(ast, context, "fax", 1,
S_COR(ast_channel_caller(ast)->id.number.valid, ast_channel_caller(ast)->id.number.str, NULL))) {
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
ast_verb(3, "Redirecting %s to fax extension (context:%s)\n", ast_channel_name(ast), context);
/* Save the DID/DNIS when we transfer the fax call to a "fax" extension */
pbx_builtin_setvar_helper(ast,"FAXEXTEN",ast_channel_exten(ast));
if (ast_async_goto(ast, context, "fax", 1)) {
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
ast_log(LOG_WARNING, "Failed to async goto '%s' into fax of '%s'\n", ast_channel_name(ast), context);
}
} else {
ast_log(LOG_NOTICE, "Fax detected but no fax extension, context:%s exten:%s\n", context, ast_channel_exten(ast));
}
} else {
ast_debug(1, "Already in a fax extension, not redirecting\n");
}
break;
case 2:
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
ast_verb(3, "Not redirecting %s to fax extension, nojump is set.\n", ast_channel_name(ast));
break;
default:
break;
}
} else {
ast_debug(1, "Fax already handled\n");
}
}
if (tmp->ast_dsp && (f->subclass.integer != 'f')) {
chan_misdn_log(2, tmp->bc->port, " --> * SEND: DTMF (AST_DSP) :%c\n", f->subclass.integer);
}
return f;
}
static struct ast_frame *misdn_read(struct ast_channel *ast)
{
struct chan_list *tmp;
int len, t;
struct pollfd pfd = { .fd = -1, .events = POLLIN };
if (!ast) {
chan_misdn_log(1, 0, "misdn_read called without ast\n");
return NULL;
}
if (!(tmp = MISDN_ASTERISK_TECH_PVT(ast))) {
chan_misdn_log(1, 0, "misdn_read called without ast->pvt\n");
return NULL;
}
if (!tmp->bc && tmp->hold.state == MISDN_HOLD_IDLE) {
chan_misdn_log(1, 0, "misdn_read called without bc\n");
return NULL;
}
pfd.fd = tmp->pipe[0];
t = ast_poll(&pfd, 1, 20);
if (t < 0) {
chan_misdn_log(-1, tmp->bc->port, "poll() error (err=%s)\n", strerror(errno));
return NULL;
}
if (!t) {
chan_misdn_log(3, tmp->bc->port, "poll() timed out\n");
len = 160;
} else if (pfd.revents & POLLIN) {
len = read(tmp->pipe[0], tmp->ast_rd_buf, sizeof(tmp->ast_rd_buf));
if (len <= 0) {
/* we hangup here, since our pipe is closed */
chan_misdn_log(2, tmp->bc->port, "misdn_read: Pipe closed, hanging up\n");
return NULL;
}
} else {
return NULL;
}
tmp->frame.frametype = AST_FRAME_VOICE;
ast_format_set(&tmp->frame.subclass.format, AST_FORMAT_ALAW, 0);
tmp->frame.datalen = len;
tmp->frame.samples = len;
tmp->frame.mallocd = 0;
tmp->frame.offset = 0;
tmp->frame.delivery = ast_tv(0, 0);
tmp->frame.src = NULL;
tmp->frame.data.ptr = tmp->ast_rd_buf;
if (tmp->faxdetect && !tmp->faxhandled) {
if (tmp->faxdetect_timeout) {
if (ast_tvzero(tmp->faxdetect_tv)) {
tmp->faxdetect_tv = ast_tvnow();
chan_misdn_log(2, tmp->bc->port, "faxdetect: starting detection with timeout: %ds ...\n", tmp->faxdetect_timeout);
return process_ast_dsp(tmp, &tmp->frame);
} else {
struct timeval tv_now = ast_tvnow();
int diff = ast_tvdiff_ms(tv_now, tmp->faxdetect_tv);
if (diff <= (tmp->faxdetect_timeout * 1000)) {
chan_misdn_log(5, tmp->bc->port, "faxdetect: detecting ...\n");
return process_ast_dsp(tmp, &tmp->frame);
} else {
chan_misdn_log(2, tmp->bc->port, "faxdetect: stopping detection (time ran out) ...\n");
tmp->faxdetect = 0;
return &tmp->frame;
}
}
} else {
chan_misdn_log(5, tmp->bc->port, "faxdetect: detecting ... (no timeout)\n");
return process_ast_dsp(tmp, &tmp->frame);
}
} else {
if (tmp->ast_dsp) {
return process_ast_dsp(tmp, &tmp->frame);
} else {
return &tmp->frame;
}
}
}
static int misdn_write(struct ast_channel *ast, struct ast_frame *frame)
{
struct chan_list *ch;
if (!ast || !(ch = MISDN_ASTERISK_TECH_PVT(ast))) {
return -1;
}
if (ch->hold.state != MISDN_HOLD_IDLE) {
chan_misdn_log(7, 0, "misdn_write: Returning because hold active\n");
return 0;
}
if (!ch->bc) {
ast_log(LOG_WARNING, "private but no bc\n");
return -1;
}
if (ch->notxtone) {
chan_misdn_log(7, ch->bc->port, "misdn_write: Returning because notxtone\n");
return 0;
}
if (!frame->subclass.format.id) {
chan_misdn_log(4, ch->bc->port, "misdn_write: * prods us\n");
return 0;
}
if (ast_format_cmp(&frame->subclass.format, &prefformat) == AST_FORMAT_CMP_NOT_EQUAL) {
chan_misdn_log(-1, ch->bc->port, "Got Unsupported Frame with Format:%s\n", ast_getformatname(&frame->subclass.format));
return 0;
}
if (!frame->samples) {
chan_misdn_log(4, ch->bc->port, "misdn_write: zero write\n");
if (!strcmp(frame->src,"ast_prod")) {
chan_misdn_log(1, ch->bc->port, "misdn_write: state (%s) prodded.\n", misdn_get_ch_state(ch));
if (ch->ts) {
chan_misdn_log(4, ch->bc->port, "Starting Playtones\n");
misdn_lib_tone_generator_start(ch->bc);
}
return 0;
}
return -1;
}
if (!ch->bc->addr) {
chan_misdn_log(8, ch->bc->port, "misdn_write: no addr for bc dropping:%d\n", frame->samples);
return 0;
}
#ifdef MISDN_DEBUG
{
int i;
int max = 5 > frame->samples ? frame->samples : 5;
ast_debug(1, "write2mISDN %p %d bytes: ", p, frame->samples);
for (i = 0; i < max; i++) {
ast_debug(1, "%2.2x ", ((char *) frame->data.ptr)[i]);
}
}
#endif
switch (ch->bc->bc_state) {
case BCHAN_ACTIVATED:
case BCHAN_BRIDGED:
break;
default:
if (!ch->dropped_frame_cnt) {
chan_misdn_log(5, ch->bc->port,
"BC not active (nor bridged) dropping: %d frames addr:%x exten:%s cid:%s ch->state:%s bc_state:%d l3id:%x\n",
frame->samples, ch->bc->addr, ast_channel_exten(ast),
S_COR(ast_channel_caller(ast)->id.number.valid, ast_channel_caller(ast)->id.number.str, ""),
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
misdn_get_ch_state(ch), ch->bc->bc_state, ch->bc->l3_id);
}
if (++ch->dropped_frame_cnt > 100) {
ch->dropped_frame_cnt = 0;
chan_misdn_log(5, ch->bc->port, "BC not active (nor bridged) dropping: %d frames addr:%x dropped > 100 frames!\n", frame->samples, ch->bc->addr);
}
return 0;
}
chan_misdn_log(9, ch->bc->port, "Sending :%d bytes to MISDN\n", frame->samples);
if (!ch->bc->nojitter && misdn_cap_is_speech(ch->bc->capability)) {
/* Buffered Transmit (triggered by read from isdn side)*/
if (misdn_jb_fill(ch->jb, frame->data.ptr, frame->samples) < 0) {
if (ch->bc->active) {
cb_log(0, ch->bc->port, "Misdn Jitterbuffer Overflow.\n");
}
}
} else {
/* transmit without jitterbuffer */
misdn_lib_tx2misdn_frm(ch->bc, frame->data.ptr, frame->samples);
}
return 0;
}
#if defined(mISDN_NATIVE_BRIDGING)
static enum ast_bridge_result misdn_bridge(struct ast_channel *c0,
struct ast_channel *c1, int flags,
struct ast_frame **fo,
struct ast_channel **rc,
int timeoutms)
{
struct chan_list *ch1, *ch2;
struct ast_channel *carr[2], *who;
int to = -1;
struct ast_frame *f;
int p1_b, p2_b;
int bridging;
misdn_cfg_get(0, MISDN_GEN_BRIDGING, &bridging, sizeof(bridging));
if (!bridging) {
/* Native mISDN bridging globally disabled. */
return AST_BRIDGE_FAILED_NOWARN;
}
ch1 = get_chan_by_ast(c0);
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
if (!ch1) {
return AST_BRIDGE_FAILED;
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
}
ch2 = get_chan_by_ast(c1);
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
if (!ch2) {
chan_list_unref(ch1, "Failed to find ch2");
return AST_BRIDGE_FAILED;
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
}
carr[0] = c0;
carr[1] = c1;
misdn_cfg_get(ch1->bc->port, MISDN_CFG_BRIDGING, &p1_b, sizeof(p1_b));
misdn_cfg_get(ch2->bc->port, MISDN_CFG_BRIDGING, &p2_b, sizeof(p2_b));
if (!p1_b || !p2_b) {
ast_log(LOG_NOTICE, "Falling back to Asterisk bridging\n");
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
chan_list_unref(ch1, "Bridge fallback ch1");
chan_list_unref(ch2, "Bridge fallback ch2");
return AST_BRIDGE_FAILED_NOWARN;
}
/* make a mISDN_dsp conference */
chan_misdn_log(1, ch1->bc->port, "I SEND: Making conference with Number:%d\n", ch1->bc->pid + 1);
misdn_lib_bridge(ch1->bc, ch2->bc);
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
ast_verb(3, "Native bridging %s and %s\n", ast_channel_name(c0), ast_channel_name(c1));
chan_misdn_log(1, ch1->bc->port, "* Making Native Bridge between \"%s\" <%s> and \"%s\" <%s>\n",
ch1->bc->caller.name,
ch1->bc->caller.number,
ch2->bc->caller.name,
ch2->bc->caller.number);
if (!(flags & AST_BRIDGE_DTMF_CHANNEL_0)) {
ch1->ignore_dtmf = 1;
}
if (!(flags & AST_BRIDGE_DTMF_CHANNEL_1)) {
ch2->ignore_dtmf = 1;
}
for (;/*ever*/;) {
to = -1;
who = ast_waitfor_n(carr, 2, &to);
if (!who) {
ast_log(LOG_NOTICE, "misdn_bridge: empty read, breaking out\n");
break;
}
f = ast_read(who);
if (!f || (f->frametype == AST_FRAME_CONTROL && f->subclass.integer != AST_CONTROL_PVT_CAUSE_CODE)) {
/* got hangup .. */
if (!f) {
chan_misdn_log(4, ch1->bc->port, "Read Null Frame\n");
} else {
chan_misdn_log(4, ch1->bc->port, "Read Frame Control class:%d\n", f->subclass.integer);
}
*fo = f;
*rc = who;
break;
}
if (f->frametype == AST_FRAME_DTMF) {
chan_misdn_log(1, 0, "Read DTMF %d from %s\n", f->subclass.integer, ast_channel_exten(who));
*fo = f;
*rc = who;
break;
}
#if 0
if (f->frametype == AST_FRAME_VOICE) {
chan_misdn_log(1, ch1->bc->port, "I SEND: Splitting conference with Number:%d\n", ch1->bc->pid +1);
continue;
}
#endif
if (f->frametype == AST_FRAME_CONTROL && f->subclass.integer == AST_CONTROL_PVT_CAUSE_CODE) {
ast_channel_hangupcause_hash_set((who == c0) ? c1 : c0, f->data.ptr, f->datalen);
} else {
ast_write((who == c0) ? c1 : c0, f);
}
}
chan_misdn_log(1, ch1->bc->port, "I SEND: Splitting conference with Number:%d\n", ch1->bc->pid + 1);
misdn_lib_split_bridge(ch1->bc, ch2->bc);
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
chan_list_unref(ch1, "Bridge complete ch1");
chan_list_unref(ch2, "Bridge complete ch2");
return AST_BRIDGE_COMPLETE;
}
#endif /* defined(mISDN_NATIVE_BRIDGING) */
/** AST INDICATIONS END **/
static int dialtone_indicate(struct chan_list *cl)
{
struct ast_channel *ast = cl->ast;
int nd = 0;
Merged revisions 49313 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r49313 | crichter | 2007-01-03 10:06:50 +0100 (Mi, 03 Jan 2007) | 41 lines Merged revisions 48319,48321,48467,48552,48576,49135,49303 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) | 1 line changed a few debugs to higher debug levels ........ r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) | 1 line added the export and import of the MISDN_ADDRESS_COMPLETE Variable to inidcate wether the extension is already completely dialed or if there might come additional digits by information elements. also added some docs for that. ........ r48467 | crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line removed FIXUP state. added check for channel allocation conflict when we create a setup while the other site creates a setup on the same channel, besides the check we resolve this conflict. ........ r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18 Dez 2006) | 1 line when our PTP Partner sends us a SETUP with a preselected channel we just accept it, even when we're NT. added some checks for segfaults. ........ r48576 | crichter | 2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line when we reject a channel, because it's in use already, we shouldn't process the setup anymore. made the channel allocation a bit easier and more understandable, removed a few unused lines ........ r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02 Jan 2007) | 1 line added check for channel ranges in the set/empty channel functions. set pmp_l1_check default to no. added misdn restart pid cli command. added cleaning of channel when we send a RELEASE_COMPLETE. ........ r49303 | crichter | 2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines * Added check for bridging in misdn_call to avoid setting echocancellation when 2 mISDN channels are involved and when bridging is set. That lead to a kernel panic before under different situations, because we switched about 2 times between hardware bridging and echocancelation * readded MISDN_URATE variable which got lost before, this should make app_v110 work again * fixed typo ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49321 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-03 11:15:02 +00:00
if (!ast) {
chan_misdn_log(0, cl->bc->port, "No Ast in dialtone_indicate\n");
Merged revisions 49313 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r49313 | crichter | 2007-01-03 10:06:50 +0100 (Mi, 03 Jan 2007) | 41 lines Merged revisions 48319,48321,48467,48552,48576,49135,49303 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) | 1 line changed a few debugs to higher debug levels ........ r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) | 1 line added the export and import of the MISDN_ADDRESS_COMPLETE Variable to inidcate wether the extension is already completely dialed or if there might come additional digits by information elements. also added some docs for that. ........ r48467 | crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line removed FIXUP state. added check for channel allocation conflict when we create a setup while the other site creates a setup on the same channel, besides the check we resolve this conflict. ........ r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18 Dez 2006) | 1 line when our PTP Partner sends us a SETUP with a preselected channel we just accept it, even when we're NT. added some checks for segfaults. ........ r48576 | crichter | 2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line when we reject a channel, because it's in use already, we shouldn't process the setup anymore. made the channel allocation a bit easier and more understandable, removed a few unused lines ........ r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02 Jan 2007) | 1 line added check for channel ranges in the set/empty channel functions. set pmp_l1_check default to no. added misdn restart pid cli command. added cleaning of channel when we send a RELEASE_COMPLETE. ........ r49303 | crichter | 2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines * Added check for bridging in misdn_call to avoid setting echocancellation when 2 mISDN channels are involved and when bridging is set. That lead to a kernel panic before under different situations, because we switched about 2 times between hardware bridging and echocancelation * readded MISDN_URATE variable which got lost before, this should make app_v110 work again * fixed typo ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49321 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-03 11:15:02 +00:00
return -1;
}
misdn_cfg_get(cl->bc->port, MISDN_CFG_NODIALTONE, &nd, sizeof(nd));
if (nd) {
chan_misdn_log(1, cl->bc->port, "Not sending Dialtone, because config wants it\n");
return 0;
}
chan_misdn_log(3, cl->bc->port, " --> Dial\n");
Merge a large set of updates to the Asterisk indications API. This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17 20:41:24 +00:00
cl->ts = ast_get_indication_tone(ast_channel_zone(ast), "dial");
Merge a large set of updates to the Asterisk indications API. This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17 20:41:24 +00:00
if (cl->ts) {
cl->notxtone = 0;
cl->norxtone = 0;
/* This prods us in misdn_write */
Merge a large set of updates to the Asterisk indications API. This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17 20:41:24 +00:00
ast_playtones_start(ast, 0, cl->ts->data, 0);
}
return 0;
}
static void hanguptone_indicate(struct chan_list *cl)
{
misdn_lib_send_tone(cl->bc, TONE_HANGUP);
}
static int stop_indicate(struct chan_list *cl)
{
struct ast_channel *ast = cl->ast;
Merged revisions 49313 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r49313 | crichter | 2007-01-03 10:06:50 +0100 (Mi, 03 Jan 2007) | 41 lines Merged revisions 48319,48321,48467,48552,48576,49135,49303 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) | 1 line changed a few debugs to higher debug levels ........ r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) | 1 line added the export and import of the MISDN_ADDRESS_COMPLETE Variable to inidcate wether the extension is already completely dialed or if there might come additional digits by information elements. also added some docs for that. ........ r48467 | crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line removed FIXUP state. added check for channel allocation conflict when we create a setup while the other site creates a setup on the same channel, besides the check we resolve this conflict. ........ r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18 Dez 2006) | 1 line when our PTP Partner sends us a SETUP with a preselected channel we just accept it, even when we're NT. added some checks for segfaults. ........ r48576 | crichter | 2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line when we reject a channel, because it's in use already, we shouldn't process the setup anymore. made the channel allocation a bit easier and more understandable, removed a few unused lines ........ r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02 Jan 2007) | 1 line added check for channel ranges in the set/empty channel functions. set pmp_l1_check default to no. added misdn restart pid cli command. added cleaning of channel when we send a RELEASE_COMPLETE. ........ r49303 | crichter | 2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines * Added check for bridging in misdn_call to avoid setting echocancellation when 2 mISDN channels are involved and when bridging is set. That lead to a kernel panic before under different situations, because we switched about 2 times between hardware bridging and echocancelation * readded MISDN_URATE variable which got lost before, this should make app_v110 work again * fixed typo ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49321 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-03 11:15:02 +00:00
if (!ast) {
chan_misdn_log(0, cl->bc->port, "No Ast in stop_indicate\n");
Merged revisions 49313 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r49313 | crichter | 2007-01-03 10:06:50 +0100 (Mi, 03 Jan 2007) | 41 lines Merged revisions 48319,48321,48467,48552,48576,49135,49303 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) | 1 line changed a few debugs to higher debug levels ........ r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) | 1 line added the export and import of the MISDN_ADDRESS_COMPLETE Variable to inidcate wether the extension is already completely dialed or if there might come additional digits by information elements. also added some docs for that. ........ r48467 | crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line removed FIXUP state. added check for channel allocation conflict when we create a setup while the other site creates a setup on the same channel, besides the check we resolve this conflict. ........ r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18 Dez 2006) | 1 line when our PTP Partner sends us a SETUP with a preselected channel we just accept it, even when we're NT. added some checks for segfaults. ........ r48576 | crichter | 2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line when we reject a channel, because it's in use already, we shouldn't process the setup anymore. made the channel allocation a bit easier and more understandable, removed a few unused lines ........ r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02 Jan 2007) | 1 line added check for channel ranges in the set/empty channel functions. set pmp_l1_check default to no. added misdn restart pid cli command. added cleaning of channel when we send a RELEASE_COMPLETE. ........ r49303 | crichter | 2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines * Added check for bridging in misdn_call to avoid setting echocancellation when 2 mISDN channels are involved and when bridging is set. That lead to a kernel panic before under different situations, because we switched about 2 times between hardware bridging and echocancelation * readded MISDN_URATE variable which got lost before, this should make app_v110 work again * fixed typo ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49321 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-03 11:15:02 +00:00
return -1;
}
chan_misdn_log(3, cl->bc->port, " --> None\n");
misdn_lib_tone_generator_stop(cl->bc);
ast_playtones_stop(ast);
Merge a large set of updates to the Asterisk indications API. This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17 20:41:24 +00:00
if (cl->ts) {
cl->ts = ast_tone_zone_sound_unref(cl->ts);
}
return 0;
}
static int start_bc_tones(struct chan_list* cl)
{
misdn_lib_tone_generator_stop(cl->bc);
cl->notxtone = 0;
cl->norxtone = 0;
return 0;
}
static int stop_bc_tones(struct chan_list *cl)
{
if (!cl) {
return -1;
}
cl->notxtone = 1;
cl->norxtone = 1;
return 0;
}
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
/*!
* \internal
* \brief Destroy the chan_list object.
*
* \param obj chan_list object to destroy.
*
* \return Nothing
*/
static void chan_list_destructor(void *obj)
{
struct chan_list *ch = obj;
#if defined(AST_MISDN_ENHANCEMENTS)
if (ch->peer) {
ao2_ref(ch->peer, -1);
ch->peer = NULL;
}
#endif /* AST_MISDN_ENHANCEMENTS */
if (ch->dsp) {
ast_dsp_free(ch->dsp);
ch->dsp = NULL;
}
/* releasing jitterbuffer */
if (ch->jb) {
misdn_jb_destroy(ch->jb);
ch->jb = NULL;
}
if (ch->overlap_dial) {
if (ch->overlap_dial_task != -1) {
misdn_tasks_remove(ch->overlap_dial_task);
ch->overlap_dial_task = -1;
}
ast_mutex_destroy(&ch->overlap_tv_lock);
}
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
if (-1 < ch->pipe[0]) {
close(ch->pipe[0]);
}
if (-1 < ch->pipe[1]) {
close(ch->pipe[1]);
}
}
/*! Returns a reference to the new chan_list. */
static struct chan_list *chan_list_init(int orig)
{
struct chan_list *cl;
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
cl = ao2_alloc(sizeof(*cl), chan_list_destructor);
if (!cl) {
chan_misdn_log(-1, 0, "misdn_request: malloc failed!");
return NULL;
}
cl->originator = orig;
cl->need_queue_hangup = 1;
cl->need_hangup = 1;
cl->need_busy = 1;
cl->overlap_dial_task = -1;
#if defined(AST_MISDN_ENHANCEMENTS)
cl->record_id = -1;
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
cl->pipe[0] = -1;
cl->pipe[1] = -1;
return cl;
}
static struct ast_channel *misdn_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause)
{
struct ast_channel *ast;
char group[BUFFERSIZE + 1] = "";
char dial_str[128];
char *dest_cp;
char *p = NULL;
int channel = 0;
int port = 0;
struct misdn_bchannel *newbc = NULL;
int dec = 0;
#if defined(AST_MISDN_ENHANCEMENTS)
int cc_retry_call = 0; /* TRUE if this is a call completion retry call */
long record_id = -1;
struct misdn_cc_record *cc_record;
const char *err_msg;
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
struct chan_list *cl;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(intf); /* interface token */
AST_APP_ARG(ext); /* extension token */
AST_APP_ARG(opts); /* options token */
);
snprintf(dial_str, sizeof(dial_str), "%s/%s", misdn_type, data);
/*
* data is ---v
* Dial(mISDN/g:group_name[/extension[/options]])
* Dial(mISDN/port[:preselected_channel][/extension[/options]])
* Dial(mISDN/cc/cc-record-id)
*
* The dial extension could be empty if you are using MISDN_KEYPAD
* to control ISDN provider features.
*/
dest_cp = ast_strdupa(data);
AST_NONSTANDARD_APP_ARGS(args, dest_cp, '/');
if (!args.ext) {
args.ext = "";
}
if (!ast_strlen_zero(args.intf)) {
if (args.intf[0] == 'g' && args.intf[1] == ':') {
/* We make a group call lets checkout which ports are in my group */
args.intf += 2;
ast_copy_string(group, args.intf, sizeof(group));
chan_misdn_log(2, 0, " --> Group Call group: %s\n", group);
#if defined(AST_MISDN_ENHANCEMENTS)
} else if (strcmp(args.intf, "cc") == 0) {
cc_retry_call = 1;
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
} else if ((p = strchr(args.intf, ':'))) {
/* we have a preselected channel */
*p++ = 0;
channel = atoi(p);
port = atoi(args.intf);
chan_misdn_log(2, port, " --> Call on preselected Channel (%d).\n", channel);
} else {
port = atoi(args.intf);
}
} else {
ast_log(LOG_WARNING, " --> ! IND : Dial(%s) WITHOUT Port or Group, check extensions.conf\n", dial_str);
return NULL;
}
#if defined(AST_MISDN_ENHANCEMENTS)
if (cc_retry_call) {
if (ast_strlen_zero(args.ext)) {
ast_log(LOG_WARNING, " --> ! IND : Dial(%s) WITHOUT cc-record-id, check extensions.conf\n", dial_str);
return NULL;
}
if (!isdigit(*args.ext)) {
ast_log(LOG_WARNING, " --> ! IND : Dial(%s) cc-record-id must be a number.\n", dial_str);
return NULL;
}
record_id = atol(args.ext);
AST_LIST_LOCK(&misdn_cc_records_db);
cc_record = misdn_cc_find_by_id(record_id);
if (!cc_record) {
AST_LIST_UNLOCK(&misdn_cc_records_db);
err_msg = misdn_cc_record_not_found;
ast_log(LOG_WARNING, " --> ! IND : Dial(%s) %s.\n", dial_str, err_msg);
return NULL;
}
if (!cc_record->activated) {
AST_LIST_UNLOCK(&misdn_cc_records_db);
err_msg = "Call completion has not been activated";
ast_log(LOG_WARNING, " --> ! IND : Dial(%s) %s.\n", dial_str, err_msg);
return NULL;
}
port = cc_record->port;
AST_LIST_UNLOCK(&misdn_cc_records_db);
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
if (misdn_cfg_is_group_method(group, METHOD_STANDARD_DEC)) {
chan_misdn_log(4, port, " --> STARTING STANDARD DEC...\n");
dec = 1;
}
if (!ast_strlen_zero(group)) {
char cfg_group[BUFFERSIZE + 1];
struct robin_list *rr = NULL;
/* Group dial */
if (misdn_cfg_is_group_method(group, METHOD_ROUND_ROBIN)) {
chan_misdn_log(4, port, " --> STARTING ROUND ROBIN...\n");
rr = get_robin_position(group);
}
if (rr) {
int port_start;
int bchan_start;
int port_up;
int check;
int maxbchans;
int wraped = 0;
if (!rr->port) {
rr->port = misdn_cfg_get_next_port_spin(0);
}
if (!rr->channel) {
rr->channel = 1;
}
bchan_start = rr->channel;
port_start = rr->port;
do {
misdn_cfg_get(rr->port, MISDN_CFG_GROUPNAME, cfg_group, sizeof(cfg_group));
if (strcasecmp(cfg_group, group)) {
wraped = 1;
rr->port = misdn_cfg_get_next_port_spin(rr->port);
rr->channel = 1;
continue;
}
misdn_cfg_get(rr->port, MISDN_CFG_PMP_L1_CHECK, &check, sizeof(check));
port_up = misdn_lib_port_up(rr->port, check);
if (!port_up) {
chan_misdn_log(1, rr->port, "L1 is not Up on this Port\n");
rr->port = misdn_cfg_get_next_port_spin(rr->port);
rr->channel = 1;
} else if (port_up < 0) {
ast_log(LOG_WARNING, "This port (%d) is blocked\n", rr->port);
rr->port = misdn_cfg_get_next_port_spin(rr->port);
rr->channel = 1;
} else {
chan_misdn_log(4, rr->port, "portup\n");
maxbchans = misdn_lib_get_maxchans(rr->port);
for (;rr->channel <= maxbchans;rr->channel++) {
/* ive come full circle and can stop now */
if (wraped && (rr->port == port_start) && (rr->channel == bchan_start)) {
break;
}
chan_misdn_log(4, rr->port, "Checking channel %d\n", rr->channel);
if ((newbc = misdn_lib_get_free_bc(rr->port, rr->channel, 0, 0))) {
chan_misdn_log(4, rr->port, " Success! Found port:%d channel:%d\n", newbc->port, newbc->channel);
rr->channel++;
break;
}
}
if (wraped && (rr->port == port_start) && (rr->channel <= bchan_start)) {
break;
} else if (!newbc || (rr->channel == maxbchans)) {
rr->port = misdn_cfg_get_next_port_spin(rr->port);
rr->channel = 1;
}
}
wraped = 1;
} while (!newbc && (rr->port > 0));
} else {
for (port = misdn_cfg_get_next_port(0); port > 0;
port = misdn_cfg_get_next_port(port)) {
misdn_cfg_get(port, MISDN_CFG_GROUPNAME, cfg_group, sizeof(cfg_group));
chan_misdn_log(3, port, "Group [%s] Port [%d]\n", group, port);
if (!strcasecmp(cfg_group, group)) {
int port_up;
int check;
misdn_cfg_get(port, MISDN_CFG_PMP_L1_CHECK, &check, sizeof(check));
port_up = misdn_lib_port_up(port, check);
chan_misdn_log(4, port, "portup:%d\n", port_up);
if (port_up > 0) {
newbc = misdn_lib_get_free_bc(port, 0, 0, dec);
if (newbc) {
break;
}
}
}
}
}
/* Group dial failed ?*/
if (!newbc) {
ast_log(LOG_WARNING,
"Could not Dial out on group '%s'.\n"
"\tEither the L2 and L1 on all of these ports where DOWN (see 'show application misdn_check_l2l1')\n"
"\tOr there was no free channel on none of the ports\n\n",
group);
return NULL;
}
} else {
/* 'Normal' Port dial * Port dial */
if (channel) {
chan_misdn_log(1, port, " --> preselected_channel: %d\n", channel);
}
newbc = misdn_lib_get_free_bc(port, channel, 0, dec);
if (!newbc) {
ast_log(LOG_WARNING, "Could not create channel on port:%d for Dial(%s)\n", port, dial_str);
return NULL;
}
}
/* create ast_channel and link all the objects together */
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
cl = chan_list_init(ORG_AST);
if (!cl) {
misdn_lib_release(newbc);
ast_log(LOG_ERROR, "Could not create call record for Dial(%s)\n", dial_str);
return NULL;
}
cl->bc = newbc;
ast = misdn_new(cl, AST_STATE_RESERVED, args.ext, NULL, cap, assignedids, requestor, port, channel);
if (!ast) {
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
chan_list_unref(cl, "Failed to create a new channel");
misdn_lib_release(newbc);
ast_log(LOG_ERROR, "Could not create Asterisk channel for Dial(%s)\n", dial_str);
return NULL;
}
#if defined(AST_MISDN_ENHANCEMENTS)
cl->record_id = record_id;
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
/* register chan in local list */
cl_queue_chan(cl);
/* fill in the config into the objects */
read_config(cl);
/* important */
cl->need_hangup = 0;
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
chan_list_unref(cl, "Successful misdn_request()");
return ast;
}
static int misdn_send_text(struct ast_channel *chan, const char *text)
{
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
struct chan_list *tmp = MISDN_ASTERISK_TECH_PVT(chan);
if (tmp && tmp->bc) {
ast_copy_string(tmp->bc->display, text, sizeof(tmp->bc->display));
misdn_lib_send_event(tmp->bc, EVENT_INFORMATION);
} else {
ast_log(LOG_WARNING, "No chan_list but send_text request?\n");
return -1;
}
return 0;
}
static struct ast_channel_tech misdn_tech = {
.type = misdn_type,
.description = "Channel driver for mISDN Support (Bri/Pri)",
.requester = misdn_request,
.send_digit_begin = misdn_digit_begin,
.send_digit_end = misdn_digit_end,
.call = misdn_call,
.hangup = misdn_hangup,
.answer = misdn_answer,
.read = misdn_read,
.write = misdn_write,
.indicate = misdn_indication,
.fixup = misdn_fixup,
.send_text = misdn_send_text,
.properties = 0,
};
static int glob_channel = 0;
static void update_name(struct ast_channel *tmp, int port, int c)
{
int chan_offset = 0;
int tmp_port = misdn_cfg_get_next_port(0);
char newname[255];
for (; tmp_port > 0; tmp_port = misdn_cfg_get_next_port(tmp_port)) {
if (tmp_port == port) {
break;
}
chan_offset += misdn_lib_port_is_pri(tmp_port) ? 30 : 2;
}
if (c < 0) {
c = 0;
}
snprintf(newname, sizeof(newname), "%s/%d-", misdn_type, chan_offset + c);
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
if (strncmp(ast_channel_name(tmp), newname, strlen(newname))) {
snprintf(newname, sizeof(newname), "%s/%d-u%d", misdn_type, chan_offset + c, glob_channel++);
ast_change_name(tmp, newname);
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
chan_misdn_log(3, port, " --> updating channel name to [%s]\n", ast_channel_name(tmp));
}
}
static struct ast_channel *misdn_new(struct chan_list *chlist, int state, char *exten, char *callerid, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, int port, int c)
{
struct ast_channel *tmp;
char *cid_name = NULL;
char *cid_num = NULL;
int chan_offset = 0;
int tmp_port = misdn_cfg_get_next_port(0);
struct ast_format tmpfmt;
for (; tmp_port > 0; tmp_port = misdn_cfg_get_next_port(tmp_port)) {
if (tmp_port == port) {
break;
}
chan_offset += misdn_lib_port_is_pri(tmp_port) ? 30 : 2;
}
if (c < 0) {
c = 0;
}
if (callerid) {
ast_callerid_parse(callerid, &cid_name, &cid_num);
}
tmp = ast_channel_alloc(1, state, cid_num, cid_name, "", exten, "", assignedids, requestor, 0, "%s/%s%d-u%d", misdn_type, c ? "" : "tmp", chan_offset + c, glob_channel++);
if (tmp) {
Merged revisions 374515-374535 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ................ r374515 | rmudgett | 2012-10-04 17:52:36 -0500 (Thu, 04 Oct 2012) | 10 lines chan_misdn: Remove some deadcode * Made setup_bc() static. Patches: patch1_unused-code.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882 ................ r374516 | rmudgett | 2012-10-04 18:01:01 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused bchan states Patches: patch2_unused-states.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374517 | rmudgett | 2012-10-04 18:17:51 -0500 (Thu, 04 Oct 2012) | 16 lines chan_misdn: Remove unnecessary null pointer checks and checks for stack->nt * cleanup_bc() is always called with valid bc (or it would've crashed before). * Value of stack->nt is known in advance at some places. * Rename handle_event() to handle_event_te(), handle_frm() to handle_frm_te(). Patches: patch3_checks.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882 ................ r374518 | rmudgett | 2012-10-04 18:21:59 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Fix spelling in log messages Patches: patch4_spelling.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374519 | rmudgett | 2012-10-04 18:31:59 -0500 (Thu, 04 Oct 2012) | 15 lines chan_misdn: Don't cleanup a bc twice. In handle_frm_te() after calling misdn_lib_send_event(bc, EVENT_RELEASE_COMPLETE) bc is emptied, cleaned and set not in use, although misdn_lib_send_event() already did the same. This is bad. When it's not in use we are not allowed to touch it. * Moved log message in front of the resulting actions and fixed it to match the case. Patches: patch5_bccleanup.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374520 | rmudgett | 2012-10-04 18:43:56 -0500 (Thu, 04 Oct 2012) | 12 lines chan_misdn: Fix memory leaks, bc, chan not cleaned up etc., really bad stuff. * Fix return codes of cb_events() for EVENT_SETUP to use caller's cleanup mechanisms. * Move cl_queue_chan() call after bearer check. Patches: patch6_leaks.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374521 | rmudgett | 2012-10-04 18:48:38 -0500 (Thu, 04 Oct 2012) | 11 lines chan_misdn: We must initialize cause on sending a DISCONNECT. We must initialize cause on sending a DISCONNECT, so it is later correctly indicated to ast_channel in case the answer (RELEASE/RELEASE_COMPLETE) does not include one. Patches: patch7_hangupcause.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374522 | rmudgett | 2012-10-04 19:03:56 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused code for upqueue Patches: patch8_unused-upqueue.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374523 | rmudgett | 2012-10-04 19:11:50 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Improve debugging (port number, messages fixed, dups removed) Patches: patch9_debug.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374533 | rmudgett | 2012-10-05 12:17:18 -0500 (Fri, 05 Oct 2012) | 8 lines chan_misdn: Better debug: we can print_bc_info even if there's no ast leg. Patches: patch10_debug-bc-2.diff (license #6372) patch uploaded by Guenther Kelleter Modified. JIRA ABE-2882 ................ r374534 | rmudgett | 2012-10-05 12:34:10 -0500 (Fri, 05 Oct 2012) | 16 lines chan_misdn: setup_bc() is called too early for an incoming SETUP on TE. This prevents the B channel from being setup for HDLC mode when requested by the bearer capability and config option hdlc=yes. It violates ETS300102 Ch.5.2.3.2: "The user, in any case, must not connect to the channel until a CONNECT ACKNOWLEDGE message has been received." * Call setup_bc() on receipt of CONNECT_ACKNOWLEGDE for PTMP, and on first response to SETUP for PTP. Patches: abe-2881-2.diff (license #6372) patch uploaded by Guenther Kelleter Modified. JIRA ABE-2881 ................ r374535 | rmudgett | 2012-10-05 12:41:05 -0500 (Fri, 05 Oct 2012) | 2 lines chan_misdn: Remove some more deadcode. ................ ........ Merged revisions 374536 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374537 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374538 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374539 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-05 18:42:14 +00:00
chan_misdn_log(2, port, " --> * NEW CHANNEL dialed:%s caller:%s\n", exten, callerid);
ast_best_codec(cap, &tmpfmt);
ast_format_cap_add(ast_channel_nativeformats(tmp), &prefformat);
ast_format_copy(ast_channel_writeformat(tmp), &tmpfmt);
ast_format_copy(ast_channel_rawwriteformat(tmp), &tmpfmt);
ast_format_copy(ast_channel_readformat(tmp), &tmpfmt);
ast_format_copy(ast_channel_rawreadformat(tmp), &tmpfmt);
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
/* Link the channel and private together */
chan_list_ref(chlist, "Give a reference to ast_channel");
MISDN_ASTERISK_TECH_PVT_SET(tmp, chlist);
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
chlist->ast = tmp;
ast_channel_tech_set(tmp, &misdn_tech);
ast_channel_priority_set(tmp, 1);
if (exten) {
ast_channel_exten_set(tmp, exten);
} else {
chan_misdn_log(1, 0, "misdn_new: no exten given.\n");
}
if (!ast_strlen_zero(cid_num)) {
/* Don't use ast_set_callerid() here because it will
* generate a needless NewCallerID event */
ast_channel_caller(tmp)->ani.number.valid = 1;
ast_channel_caller(tmp)->ani.number.str = ast_strdup(cid_num);
}
if (pipe(chlist->pipe) < 0) {
ast_log(LOG_ERROR, "Pipe failed\n");
}
ast_channel_set_fd(tmp, 0, chlist->pipe[0]);
ast_channel_rings_set(tmp, (state == AST_STATE_RING) ? 1 : 0);
ast_jb_configure(tmp, misdn_get_global_jbconf());
ast_channel_unlock(tmp);
} else {
chan_misdn_log(-1, 0, "Unable to allocate channel structure\n");
}
return tmp;
}
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
/*! Returns a reference to the found chan_list. */
static struct chan_list *find_chan_by_bc(struct misdn_bchannel *bc)
{
struct chan_list *help;
ast_mutex_lock(&cl_te_lock);
for (help = cl_te; help; help = help->next) {
if (help->bc == bc) {
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
chan_list_ref(help, "Found chan_list by bc");
ast_mutex_unlock(&cl_te_lock);
return help;
}
}
ast_mutex_unlock(&cl_te_lock);
chan_misdn_log(6, bc->port,
"$$$ find_chan_by_bc: No channel found for dialed:%s caller:\"%s\" <%s>\n",
bc->dialed.number,
bc->caller.name,
bc->caller.number);
return NULL;
}
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
/*! Returns a reference to the found chan_list. */
static struct chan_list *find_hold_call(struct misdn_bchannel *bc)
{
struct chan_list *help;
if (bc->pri) {
return NULL;
}
chan_misdn_log(6, bc->port, "$$$ find_hold_call: channel:%d dialed:%s caller:\"%s\" <%s>\n",
bc->channel,
bc->dialed.number,
bc->caller.name,
bc->caller.number);
ast_mutex_lock(&cl_te_lock);
for (help = cl_te; help; help = help->next) {
chan_misdn_log(4, bc->port, "$$$ find_hold_call: --> hold:%d channel:%d\n", help->hold.state, help->hold.channel);
if (help->hold.state == MISDN_HOLD_ACTIVE && help->hold.port == bc->port) {
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
chan_list_ref(help, "Found chan_list hold call");
ast_mutex_unlock(&cl_te_lock);
return help;
}
}
ast_mutex_unlock(&cl_te_lock);
chan_misdn_log(6, bc->port,
"$$$ find_hold_call: No channel found for dialed:%s caller:\"%s\" <%s>\n",
bc->dialed.number,
bc->caller.name,
bc->caller.number);
return NULL;
}
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
/*! Returns a reference to the found chan_list. */
static struct chan_list *find_hold_call_l3(unsigned long l3_id)
{
struct chan_list *help;
ast_mutex_lock(&cl_te_lock);
for (help = cl_te; help; help = help->next) {
if (help->hold.state != MISDN_HOLD_IDLE && help->l3id == l3_id) {
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
chan_list_ref(help, "Found chan_list hold call l3");
ast_mutex_unlock(&cl_te_lock);
return help;
}
}
ast_mutex_unlock(&cl_te_lock);
return NULL;
}
#define TRANSFER_ON_HELD_CALL_HANGUP 1
#if defined(TRANSFER_ON_HELD_CALL_HANGUP)
/*!
* \internal
* \brief Find a suitable active call to go with a held call so we could try a transfer.
*
* \param bc B channel record.
*
* \return Found call record or NULL.
*
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
* \note Returns a reference to the found chan_list.
*
* \note There could be a possibility where we find the wrong active call to transfer.
* This concern is mitigated by the fact that there could be at most one other call
* on a PTMP BRI link to another device. Maybe the l3_id could help in locating an
* active call on the same TEI?
*/
static struct chan_list *find_hold_active_call(struct misdn_bchannel *bc)
{
struct chan_list *list;
ast_mutex_lock(&cl_te_lock);
for (list = cl_te; list; list = list->next) {
if (list->hold.state == MISDN_HOLD_IDLE && list->bc && list->bc->port == bc->port
&& list->ast) {
switch (list->state) {
case MISDN_PROCEEDING:
case MISDN_PROGRESS:
case MISDN_ALERTING:
case MISDN_CONNECTED:
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
chan_list_ref(list, "Found chan_list hold active call");
ast_mutex_unlock(&cl_te_lock);
return list;
default:
break;
}
}
}
ast_mutex_unlock(&cl_te_lock);
return NULL;
}
#endif /* defined(TRANSFER_ON_HELD_CALL_HANGUP) */
static void cl_queue_chan(struct chan_list *chan)
{
chan_misdn_log(4, chan->bc ? chan->bc->port : 0, "* Queuing chan %p\n", chan);
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
chan_list_ref(chan, "Adding chan_list to list");
ast_mutex_lock(&cl_te_lock);
chan->next = NULL;
if (!cl_te) {
/* List is empty, make head of list. */
cl_te = chan;
} else {
struct chan_list *help;
/* Put at end of list. */
for (help = cl_te; help->next; help = help->next) {
}
help->next = chan;
}
ast_mutex_unlock(&cl_te_lock);
}
static int cl_dequeue_chan(struct chan_list *chan)
{
int found_it;
struct chan_list *help;
ast_mutex_lock(&cl_te_lock);
if (!cl_te) {
/* List is empty. */
ast_mutex_unlock(&cl_te_lock);
return 0;
}
if (cl_te == chan) {
/* What we want is the head of the list. */
cl_te = cl_te->next;
ast_mutex_unlock(&cl_te_lock);
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
chan_list_unref(chan, "Removed chan_list from list head");
return 1;
}
found_it = 0;
for (help = cl_te; help->next; help = help->next) {
if (help->next == chan) {
/* Found it in the list. */
help->next = help->next->next;
found_it = 1;
break;
}
}
ast_mutex_unlock(&cl_te_lock);
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
if (found_it) {
chan_list_unref(chan, "Removed chan_list from list");
}
return found_it;
}
/** Channel Queue End **/
static int pbx_start_chan(struct chan_list *ch)
{
int ret = ast_pbx_start(ch->ast);
ch->need_hangup = (ret >= 0) ? 0 : 1;
return ret;
}
static void hangup_chan(struct chan_list *ch, struct misdn_bchannel *bc)
{
Merged revisions 374515-374535 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ................ r374515 | rmudgett | 2012-10-04 17:52:36 -0500 (Thu, 04 Oct 2012) | 10 lines chan_misdn: Remove some deadcode * Made setup_bc() static. Patches: patch1_unused-code.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882 ................ r374516 | rmudgett | 2012-10-04 18:01:01 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused bchan states Patches: patch2_unused-states.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374517 | rmudgett | 2012-10-04 18:17:51 -0500 (Thu, 04 Oct 2012) | 16 lines chan_misdn: Remove unnecessary null pointer checks and checks for stack->nt * cleanup_bc() is always called with valid bc (or it would've crashed before). * Value of stack->nt is known in advance at some places. * Rename handle_event() to handle_event_te(), handle_frm() to handle_frm_te(). Patches: patch3_checks.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882 ................ r374518 | rmudgett | 2012-10-04 18:21:59 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Fix spelling in log messages Patches: patch4_spelling.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374519 | rmudgett | 2012-10-04 18:31:59 -0500 (Thu, 04 Oct 2012) | 15 lines chan_misdn: Don't cleanup a bc twice. In handle_frm_te() after calling misdn_lib_send_event(bc, EVENT_RELEASE_COMPLETE) bc is emptied, cleaned and set not in use, although misdn_lib_send_event() already did the same. This is bad. When it's not in use we are not allowed to touch it. * Moved log message in front of the resulting actions and fixed it to match the case. Patches: patch5_bccleanup.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374520 | rmudgett | 2012-10-04 18:43:56 -0500 (Thu, 04 Oct 2012) | 12 lines chan_misdn: Fix memory leaks, bc, chan not cleaned up etc., really bad stuff. * Fix return codes of cb_events() for EVENT_SETUP to use caller's cleanup mechanisms. * Move cl_queue_chan() call after bearer check. Patches: patch6_leaks.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374521 | rmudgett | 2012-10-04 18:48:38 -0500 (Thu, 04 Oct 2012) | 11 lines chan_misdn: We must initialize cause on sending a DISCONNECT. We must initialize cause on sending a DISCONNECT, so it is later correctly indicated to ast_channel in case the answer (RELEASE/RELEASE_COMPLETE) does not include one. Patches: patch7_hangupcause.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374522 | rmudgett | 2012-10-04 19:03:56 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused code for upqueue Patches: patch8_unused-upqueue.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374523 | rmudgett | 2012-10-04 19:11:50 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Improve debugging (port number, messages fixed, dups removed) Patches: patch9_debug.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374533 | rmudgett | 2012-10-05 12:17:18 -0500 (Fri, 05 Oct 2012) | 8 lines chan_misdn: Better debug: we can print_bc_info even if there's no ast leg. Patches: patch10_debug-bc-2.diff (license #6372) patch uploaded by Guenther Kelleter Modified. JIRA ABE-2882 ................ r374534 | rmudgett | 2012-10-05 12:34:10 -0500 (Fri, 05 Oct 2012) | 16 lines chan_misdn: setup_bc() is called too early for an incoming SETUP on TE. This prevents the B channel from being setup for HDLC mode when requested by the bearer capability and config option hdlc=yes. It violates ETS300102 Ch.5.2.3.2: "The user, in any case, must not connect to the channel until a CONNECT ACKNOWLEDGE message has been received." * Call setup_bc() on receipt of CONNECT_ACKNOWLEGDE for PTMP, and on first response to SETUP for PTP. Patches: abe-2881-2.diff (license #6372) patch uploaded by Guenther Kelleter Modified. JIRA ABE-2881 ................ r374535 | rmudgett | 2012-10-05 12:41:05 -0500 (Fri, 05 Oct 2012) | 2 lines chan_misdn: Remove some more deadcode. ................ ........ Merged revisions 374536 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374537 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374538 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374539 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-05 18:42:14 +00:00
int port = bc->port;
if (!ch) {
Merged revisions 374515-374535 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ................ r374515 | rmudgett | 2012-10-04 17:52:36 -0500 (Thu, 04 Oct 2012) | 10 lines chan_misdn: Remove some deadcode * Made setup_bc() static. Patches: patch1_unused-code.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882 ................ r374516 | rmudgett | 2012-10-04 18:01:01 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused bchan states Patches: patch2_unused-states.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374517 | rmudgett | 2012-10-04 18:17:51 -0500 (Thu, 04 Oct 2012) | 16 lines chan_misdn: Remove unnecessary null pointer checks and checks for stack->nt * cleanup_bc() is always called with valid bc (or it would've crashed before). * Value of stack->nt is known in advance at some places. * Rename handle_event() to handle_event_te(), handle_frm() to handle_frm_te(). Patches: patch3_checks.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882 ................ r374518 | rmudgett | 2012-10-04 18:21:59 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Fix spelling in log messages Patches: patch4_spelling.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374519 | rmudgett | 2012-10-04 18:31:59 -0500 (Thu, 04 Oct 2012) | 15 lines chan_misdn: Don't cleanup a bc twice. In handle_frm_te() after calling misdn_lib_send_event(bc, EVENT_RELEASE_COMPLETE) bc is emptied, cleaned and set not in use, although misdn_lib_send_event() already did the same. This is bad. When it's not in use we are not allowed to touch it. * Moved log message in front of the resulting actions and fixed it to match the case. Patches: patch5_bccleanup.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374520 | rmudgett | 2012-10-04 18:43:56 -0500 (Thu, 04 Oct 2012) | 12 lines chan_misdn: Fix memory leaks, bc, chan not cleaned up etc., really bad stuff. * Fix return codes of cb_events() for EVENT_SETUP to use caller's cleanup mechanisms. * Move cl_queue_chan() call after bearer check. Patches: patch6_leaks.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374521 | rmudgett | 2012-10-04 18:48:38 -0500 (Thu, 04 Oct 2012) | 11 lines chan_misdn: We must initialize cause on sending a DISCONNECT. We must initialize cause on sending a DISCONNECT, so it is later correctly indicated to ast_channel in case the answer (RELEASE/RELEASE_COMPLETE) does not include one. Patches: patch7_hangupcause.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374522 | rmudgett | 2012-10-04 19:03:56 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused code for upqueue Patches: patch8_unused-upqueue.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374523 | rmudgett | 2012-10-04 19:11:50 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Improve debugging (port number, messages fixed, dups removed) Patches: patch9_debug.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374533 | rmudgett | 2012-10-05 12:17:18 -0500 (Fri, 05 Oct 2012) | 8 lines chan_misdn: Better debug: we can print_bc_info even if there's no ast leg. Patches: patch10_debug-bc-2.diff (license #6372) patch uploaded by Guenther Kelleter Modified. JIRA ABE-2882 ................ r374534 | rmudgett | 2012-10-05 12:34:10 -0500 (Fri, 05 Oct 2012) | 16 lines chan_misdn: setup_bc() is called too early for an incoming SETUP on TE. This prevents the B channel from being setup for HDLC mode when requested by the bearer capability and config option hdlc=yes. It violates ETS300102 Ch.5.2.3.2: "The user, in any case, must not connect to the channel until a CONNECT ACKNOWLEDGE message has been received." * Call setup_bc() on receipt of CONNECT_ACKNOWLEGDE for PTMP, and on first response to SETUP for PTP. Patches: abe-2881-2.diff (license #6372) patch uploaded by Guenther Kelleter Modified. JIRA ABE-2881 ................ r374535 | rmudgett | 2012-10-05 12:41:05 -0500 (Fri, 05 Oct 2012) | 2 lines chan_misdn: Remove some more deadcode. ................ ........ Merged revisions 374536 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374537 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374538 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374539 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-05 18:42:14 +00:00
cb_log(1, port, "Cannot hangup chan, no ch\n");
return;
}
cb_log(5, port, "hangup_chan called\n");
if (ch->need_hangup) {
cb_log(2, port, " --> hangup\n");
ch->need_hangup = 0;
ch->need_queue_hangup = 0;
if (ch->ast && send_cause2ast(ch->ast, bc, ch)) {
ast_hangup(ch->ast);
}
return;
}
if (!ch->need_queue_hangup) {
cb_log(2, port, " --> No need to queue hangup\n");
return;
}
ch->need_queue_hangup = 0;
if (ch->ast) {
if (send_cause2ast(ch->ast, bc, ch)) {
ast_queue_hangup_with_cause(ch->ast, bc->cause);
cb_log(2, port, " --> queue_hangup\n");
}
} else {
cb_log(1, port, "Cannot hangup chan, no ast\n");
}
}
/*!
* \internal
* \brief ISDN asked us to release channel, pendant to misdn_hangup.
*
* \param ch Call channel record to release.
* \param bc Current B channel record associated with ch.
*
* \return Nothing
*
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
* \note The only valid thing to do with ch after calling is to chan_list_unref(ch, "").
*/
static void release_chan(struct chan_list *ch, struct misdn_bchannel *bc)
{
struct ast_channel *ast;
chan_misdn_log(5, bc->port, "release_chan: bc with pid:%d l3id: %x\n", bc->pid, bc->l3_id);
ast_mutex_lock(&release_lock);
for (;;) {
ast = ch->ast;
if (!ast || !ast_channel_trylock(ast)) {
break;
}
DEADLOCK_AVOIDANCE(&release_lock);
}
if (!cl_dequeue_chan(ch)) {
/* Someone already released it. */
if (ast) {
ast_channel_unlock(ast);
}
ast_mutex_unlock(&release_lock);
return;
}
ch->state = MISDN_CLEANING;
ch->ast = NULL;
if (ast) {
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
struct chan_list *ast_ch;
ast_ch = MISDN_ASTERISK_TECH_PVT(ast);
MISDN_ASTERISK_TECH_PVT_SET(ast, NULL);
chan_misdn_log(1, bc->port,
"* RELEASING CHANNEL pid:%d context:%s dialed:%s caller:\"%s\" <%s>\n",
bc->pid,
ast_channel_context(ast),
ast_channel_exten(ast),
S_COR(ast_channel_caller(ast)->id.name.valid, ast_channel_caller(ast)->id.name.str, ""),
S_COR(ast_channel_caller(ast)->id.number.valid, ast_channel_caller(ast)->id.number.str, ""));
if (ast_channel_state(ast) != AST_STATE_RESERVED) {
chan_misdn_log(3, bc->port, " --> Setting AST State to down\n");
ast_setstate(ast, AST_STATE_DOWN);
}
ast_channel_unlock(ast);
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
if (ast_ch) {
chan_list_unref(ast_ch, "Release ast_channel reference.");
}
}
if (ch->originator == ORG_AST) {
--misdn_out_calls[bc->port];
} else {
--misdn_in_calls[bc->port];
}
ast_mutex_unlock(&release_lock);
}
/*!
* \internal
* \brief Do everything in release_chan() that makes sense without a bc.
*
* \param ch Call channel record to release.
*
* \return Nothing
*
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
* \note The only valid thing to do with ch after calling is to chan_list_unref(ch, "").
*/
static void release_chan_early(struct chan_list *ch)
{
struct ast_channel *ast;
ast_mutex_lock(&release_lock);
for (;;) {
ast = ch->ast;
if (!ast || !ast_channel_trylock(ast)) {
break;
}
DEADLOCK_AVOIDANCE(&release_lock);
}
if (!cl_dequeue_chan(ch)) {
/* Someone already released it. */
if (ast) {
ast_channel_unlock(ast);
}
ast_mutex_unlock(&release_lock);
return;
}
ch->state = MISDN_CLEANING;
ch->ast = NULL;
if (ast) {
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
struct chan_list *ast_ch;
ast_ch = MISDN_ASTERISK_TECH_PVT(ast);
MISDN_ASTERISK_TECH_PVT_SET(ast, NULL);
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
if (ast_channel_state(ast) != AST_STATE_RESERVED) {
ast_setstate(ast, AST_STATE_DOWN);
}
ast_channel_unlock(ast);
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
if (ast_ch) {
chan_list_unref(ast_ch, "Release ast_channel reference.");
}
}
if (ch->hold.state != MISDN_HOLD_IDLE) {
if (ch->originator == ORG_AST) {
--misdn_out_calls[ch->hold.port];
} else {
--misdn_in_calls[ch->hold.port];
}
}
ast_mutex_unlock(&release_lock);
}
/*!
* \internal
* \brief Attempt to transfer the active channel party to the held channel party.
*
* \param active_ch Channel currently connected.
* \param held_ch Channel currently on hold.
*
* \retval 0 on success.
* \retval -1 on error.
*/
static int misdn_attempt_transfer(struct chan_list *active_ch, struct chan_list *held_ch)
{
int retval;
enum ast_transfer_result xfer_res;
struct ast_channel *to_target;
struct ast_channel *to_transferee;
switch (active_ch->state) {
case MISDN_PROCEEDING:
case MISDN_PROGRESS:
case MISDN_ALERTING:
case MISDN_CONNECTED:
break;
default:
return -1;
}
ast_channel_lock_both(held_ch->ast, active_ch->ast);
to_target = active_ch->ast;
to_transferee = held_ch->ast;
chan_misdn_log(1, held_ch->hold.port, "TRANSFERRING %s to %s\n",
ast_channel_name(to_transferee), ast_channel_name(to_target));
held_ch->hold.state = MISDN_HOLD_TRANSFER;
ast_channel_ref(to_target);
ast_channel_ref(to_transferee);
ast_channel_unlock(to_target);
ast_channel_unlock(to_transferee);
retval = 0;
xfer_res = ast_bridge_transfer_attended(to_transferee, to_target);
if (xfer_res != AST_BRIDGE_TRANSFER_SUCCESS) {
retval = -1;
}
ast_channel_unref(to_target);
ast_channel_unref(to_transferee);
return retval;
}
static void do_immediate_setup(struct misdn_bchannel *bc, struct chan_list *ch, struct ast_channel *ast)
{
char *predial;
struct ast_frame fr;
predial = ast_strdupa(ast_channel_exten(ast));
ch->state = MISDN_DIALING;
if (!ch->noautorespond_on_setup) {
if (bc->nt) {
misdn_lib_send_event(bc, EVENT_SETUP_ACKNOWLEDGE);
} else {
if (misdn_lib_is_ptp(bc->port)) {
misdn_lib_send_event(bc, EVENT_SETUP_ACKNOWLEDGE);
} else {
misdn_lib_send_event(bc, EVENT_PROCEEDING);
}
}
} else {
ch->state = MISDN_INCOMING_SETUP;
}
chan_misdn_log(1, bc->port,
"* Starting Ast context:%s dialed:%s caller:\"%s\" <%s> with 's' extension\n",
ast_channel_context(ast),
ast_channel_exten(ast),
(ast_channel_caller(ast)->id.name.valid && ast_channel_caller(ast)->id.name.str)
? ast_channel_caller(ast)->id.name.str : "",
(ast_channel_caller(ast)->id.number.valid && ast_channel_caller(ast)->id.number.str)
? ast_channel_caller(ast)->id.number.str : "");
ast_channel_exten_set(ast, "s");
if (!ast_canmatch_extension(ast, ast_channel_context(ast), ast_channel_exten(ast), 1, bc->caller.number) || pbx_start_chan(ch) < 0) {
ast = NULL;
bc->out_cause = AST_CAUSE_UNALLOCATED;
hangup_chan(ch, bc);
hanguptone_indicate(ch);
misdn_lib_send_event(bc, bc->nt ? EVENT_RELEASE_COMPLETE : EVENT_DISCONNECT);
}
while (!ast_strlen_zero(predial)) {
fr.frametype = AST_FRAME_DTMF;
fr.subclass.integer = *predial;
fr.src = NULL;
fr.data.ptr = NULL;
fr.datalen = 0;
fr.samples = 0;
fr.mallocd = 0;
fr.offset = 0;
fr.delivery = ast_tv(0,0);
if (ch->ast && MISDN_ASTERISK_TECH_PVT(ch->ast)) {
ast_queue_frame(ch->ast, &fr);
}
predial++;
}
}
/*!
* \retval -1 if can hangup after calling.
* \retval 0 if cannot hangup after calling.
*/
static int send_cause2ast(struct ast_channel *ast, struct misdn_bchannel *bc, struct chan_list *ch)
{
int can_hangup;
if (!ast) {
chan_misdn_log(1, 0, "send_cause2ast: No Ast\n");
return 0;
}
if (!bc) {
chan_misdn_log(1, 0, "send_cause2ast: No BC\n");
return 0;
}
if (!ch) {
chan_misdn_log(1, 0, "send_cause2ast: No Ch\n");
return 0;
}
ast_channel_hangupcause_set(ast, bc->cause);
can_hangup = -1;
switch (bc->cause) {
case AST_CAUSE_UNALLOCATED:
case AST_CAUSE_NO_ROUTE_TRANSIT_NET:
case AST_CAUSE_NO_ROUTE_DESTINATION:
case 4: /* Send special information tone */
case AST_CAUSE_NUMBER_CHANGED:
case AST_CAUSE_DESTINATION_OUT_OF_ORDER:
/* Congestion Cases */
/*
* Not Queueing the Congestion anymore, since we want to hear
* the inband message
*
chan_misdn_log(1, bc ? bc->port : 0, " --> * SEND: Queue Congestion pid:%d\n", bc ? bc->pid : -1);
ch->state = MISDN_BUSY;
ast_queue_control(ast, AST_CONTROL_CONGESTION);
*/
break;
case AST_CAUSE_CALL_REJECTED:
case AST_CAUSE_USER_BUSY:
ch->state = MISDN_BUSY;
if (!ch->need_busy) {
chan_misdn_log(1, bc ? bc->port : 0, "Queued busy already\n");
break;
}
ch->need_busy = 0;
chan_misdn_log(1, bc ? bc->port : 0, " --> * SEND: Queue Busy pid:%d\n", bc ? bc->pid : -1);
ast_queue_control(ast, AST_CONTROL_BUSY);
/* The BUSY is likely to cause a hangup or the user needs to hear it. */
can_hangup = 0;
break;
}
return can_hangup;
}
/*! \brief Import parameters from the dialplan environment variables */
void import_ch(struct ast_channel *chan, struct misdn_bchannel *bc, struct chan_list *ch)
{
const char *tmp;
ast_channel_lock(chan);
tmp = pbx_builtin_getvar_helper(chan, "MISDN_ADDRESS_COMPLETE");
Merged revisions 49313 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r49313 | crichter | 2007-01-03 10:06:50 +0100 (Mi, 03 Jan 2007) | 41 lines Merged revisions 48319,48321,48467,48552,48576,49135,49303 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) | 1 line changed a few debugs to higher debug levels ........ r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) | 1 line added the export and import of the MISDN_ADDRESS_COMPLETE Variable to inidcate wether the extension is already completely dialed or if there might come additional digits by information elements. also added some docs for that. ........ r48467 | crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line removed FIXUP state. added check for channel allocation conflict when we create a setup while the other site creates a setup on the same channel, besides the check we resolve this conflict. ........ r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18 Dez 2006) | 1 line when our PTP Partner sends us a SETUP with a preselected channel we just accept it, even when we're NT. added some checks for segfaults. ........ r48576 | crichter | 2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line when we reject a channel, because it's in use already, we shouldn't process the setup anymore. made the channel allocation a bit easier and more understandable, removed a few unused lines ........ r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02 Jan 2007) | 1 line added check for channel ranges in the set/empty channel functions. set pmp_l1_check default to no. added misdn restart pid cli command. added cleaning of channel when we send a RELEASE_COMPLETE. ........ r49303 | crichter | 2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines * Added check for bridging in misdn_call to avoid setting echocancellation when 2 mISDN channels are involved and when bridging is set. That lead to a kernel panic before under different situations, because we switched about 2 times between hardware bridging and echocancelation * readded MISDN_URATE variable which got lost before, this should make app_v110 work again * fixed typo ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49321 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-03 11:15:02 +00:00
if (tmp && (atoi(tmp) == 1)) {
bc->sending_complete = 1;
Merged revisions 49313 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r49313 | crichter | 2007-01-03 10:06:50 +0100 (Mi, 03 Jan 2007) | 41 lines Merged revisions 48319,48321,48467,48552,48576,49135,49303 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) | 1 line changed a few debugs to higher debug levels ........ r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) | 1 line added the export and import of the MISDN_ADDRESS_COMPLETE Variable to inidcate wether the extension is already completely dialed or if there might come additional digits by information elements. also added some docs for that. ........ r48467 | crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line removed FIXUP state. added check for channel allocation conflict when we create a setup while the other site creates a setup on the same channel, besides the check we resolve this conflict. ........ r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18 Dez 2006) | 1 line when our PTP Partner sends us a SETUP with a preselected channel we just accept it, even when we're NT. added some checks for segfaults. ........ r48576 | crichter | 2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line when we reject a channel, because it's in use already, we shouldn't process the setup anymore. made the channel allocation a bit easier and more understandable, removed a few unused lines ........ r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02 Jan 2007) | 1 line added check for channel ranges in the set/empty channel functions. set pmp_l1_check default to no. added misdn restart pid cli command. added cleaning of channel when we send a RELEASE_COMPLETE. ........ r49303 | crichter | 2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines * Added check for bridging in misdn_call to avoid setting echocancellation when 2 mISDN channels are involved and when bridging is set. That lead to a kernel panic before under different situations, because we switched about 2 times between hardware bridging and echocancelation * readded MISDN_URATE variable which got lost before, this should make app_v110 work again * fixed typo ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49321 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-03 11:15:02 +00:00
}
tmp = pbx_builtin_getvar_helper(chan, "MISDN_USERUSER");
if (tmp) {
ast_log(LOG_NOTICE, "MISDN_USERUSER: %s\n", tmp);
ast_copy_string(bc->uu, tmp, sizeof(bc->uu));
bc->uulen = strlen(bc->uu);
}
tmp = pbx_builtin_getvar_helper(chan, "MISDN_KEYPAD");
if (tmp) {
ast_copy_string(bc->keypad, tmp, sizeof(bc->keypad));
}
ast_channel_unlock(chan);
}
/*! \brief Export parameters to the dialplan environment variables */
void export_ch(struct ast_channel *chan, struct misdn_bchannel *bc, struct chan_list *ch)
{
char tmp[32];
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
/*
* The only use for MISDN_PID is if there is a problem and you
* have to use the "misdn restart pid" CLI command. Otherwise,
* the pid is not used by anyone. The internal use of MISDN_PID
* has been deleted.
*/
chan_misdn_log(3, bc->port, " --> EXPORT_PID: pid:%d\n", bc->pid);
snprintf(tmp, sizeof(tmp), "%d", bc->pid);
pbx_builtin_setvar_helper(chan, "_MISDN_PID", tmp);
if (bc->sending_complete) {
snprintf(tmp, sizeof(tmp), "%d", bc->sending_complete);
pbx_builtin_setvar_helper(chan, "MISDN_ADDRESS_COMPLETE", tmp);
}
if (bc->urate) {
snprintf(tmp, sizeof(tmp), "%d", bc->urate);
pbx_builtin_setvar_helper(chan, "MISDN_URATE", tmp);
}
if (bc->uulen) {
pbx_builtin_setvar_helper(chan, "MISDN_USERUSER", bc->uu);
}
if (!ast_strlen_zero(bc->keypad)) {
pbx_builtin_setvar_helper(chan, "MISDN_KEYPAD", bc->keypad);
}
}
int add_in_calls(int port)
{
int max_in_calls;
misdn_cfg_get(port, MISDN_CFG_MAX_IN, &max_in_calls, sizeof(max_in_calls));
misdn_in_calls[port]++;
if (max_in_calls >= 0 && max_in_calls < misdn_in_calls[port]) {
ast_log(LOG_NOTICE, "Marking Incoming Call on port[%d]\n", port);
return misdn_in_calls[port] - max_in_calls;
}
return 0;
}
int add_out_calls(int port)
{
int max_out_calls;
misdn_cfg_get(port, MISDN_CFG_MAX_OUT, &max_out_calls, sizeof(max_out_calls));
if (max_out_calls >= 0 && max_out_calls <= misdn_out_calls[port]) {
ast_log(LOG_NOTICE, "Rejecting Outgoing Call on port[%d]\n", port);
return (misdn_out_calls[port] + 1) - max_out_calls;
}
misdn_out_calls[port]++;
return 0;
}
static void start_pbx(struct chan_list *ch, struct misdn_bchannel *bc, struct ast_channel *chan)
{
if (pbx_start_chan(ch) < 0) {
hangup_chan(ch, bc);
chan_misdn_log(-1, bc->port, "ast_pbx_start returned <0 in SETUP\n");
if (bc->nt) {
hanguptone_indicate(ch);
misdn_lib_send_event(bc, EVENT_RELEASE_COMPLETE);
} else {
misdn_lib_send_event(bc, EVENT_RELEASE);
}
}
}
static void wait_for_digits(struct chan_list *ch, struct misdn_bchannel *bc, struct ast_channel *chan)
{
ch->state = MISDN_WAITING4DIGS;
misdn_lib_send_event(bc, EVENT_SETUP_ACKNOWLEDGE);
if (bc->nt && !bc->dialed.number[0]) {
dialtone_indicate(ch);
}
}
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Handle the FACILITY CCBSStatusRequest message.
*
* \param port Logical port number.
* \param facility Facility ie contents.
*
* \return Nothing
*/
static void misdn_cc_handle_ccbs_status_request(int port, const struct FacParm *facility)
{
struct misdn_cc_record *cc_record;
struct misdn_bchannel dummy;
switch (facility->u.CCBSStatusRequest.ComponentType) {
case FacComponent_Invoke:
/* Build message */
misdn_make_dummy(&dummy, port, 0, misdn_lib_port_is_nt(port), 0);
dummy.fac_out.Function = Fac_CCBSStatusRequest;
dummy.fac_out.u.CCBSStatusRequest.InvokeID = facility->u.CCBSStatusRequest.InvokeID;
dummy.fac_out.u.CCBSStatusRequest.ComponentType = FacComponent_Result;
/* Answer User-A free question */
AST_LIST_LOCK(&misdn_cc_records_db);
cc_record = misdn_cc_find_by_reference(port, facility->u.CCBSStatusRequest.Component.Invoke.CCBSReference);
if (cc_record) {
dummy.fac_out.u.CCBSStatusRequest.Component.Result.Free = cc_record->party_a_free;
} else {
/* No record so say User-A is free */
dummy.fac_out.u.CCBSStatusRequest.Component.Result.Free = 1;
}
AST_LIST_UNLOCK(&misdn_cc_records_db);
/* Send message */
print_facility(&dummy.fac_out, &dummy);
misdn_lib_send_event(&dummy, EVENT_FACILITY);
break;
default:
chan_misdn_log(0, port, " --> not yet handled: facility type:0x%04X\n", facility->Function);
break;
}
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Start a PBX to notify that User-B is available.
*
* \param record_id Call completion record ID
* \param notify Dialplan location to start processing.
*
* \return Nothing
*/
static void misdn_cc_pbx_notify(long record_id, const struct misdn_cc_notify *notify)
{
struct ast_channel *chan;
char id_str[32];
static unsigned short sequence = 0;
/* Create a channel to notify with */
snprintf(id_str, sizeof(id_str), "%ld", record_id);
chan = ast_channel_alloc(0, AST_STATE_DOWN, id_str, NULL, NULL,
notify->exten, notify->context, NULL, 0,
"mISDN-CC/%ld-%X", record_id, (unsigned) ++sequence);
if (!chan) {
ast_log(LOG_ERROR, "Unable to allocate channel!\n");
return;
}
ast_channel_priority_set(chan, notify->priority);
ast_free(ast_channel_dialed(chan)->number.str);
ast_channel_dialed(chan)->number.str = ast_strdup(notify->exten);
ast_channel_unlock(chan);
if (ast_pbx_start(chan)) {
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
ast_log(LOG_WARNING, "Unable to start pbx channel %s!\n", ast_channel_name(chan));
ast_channel_release(chan);
} else {
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
ast_verb(1, "Started pbx for call completion notify channel %s\n", ast_channel_name(chan));
}
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Handle the FACILITY CCBS_T_RemoteUserFree message.
*
* \param bc B channel control structure message came in on
*
* \return Nothing
*/
static void misdn_cc_handle_T_remote_user_free(struct misdn_bchannel *bc)
{
struct misdn_cc_record *cc_record;
struct misdn_cc_notify notify;
long record_id;
AST_LIST_LOCK(&misdn_cc_records_db);
cc_record = misdn_cc_find_by_bc(bc);
if (cc_record) {
if (cc_record->party_a_free) {
notify = cc_record->remote_user_free;
} else {
/* Send CCBS_T_Suspend message */
bc->fac_out.Function = Fac_CCBS_T_Suspend;
bc->fac_out.u.CCBS_T_Suspend.InvokeID = ++misdn_invoke_id;
print_facility(&bc->fac_out, bc);
misdn_lib_send_event(bc, EVENT_FACILITY);
notify = cc_record->b_free;
}
record_id = cc_record->record_id;
AST_LIST_UNLOCK(&misdn_cc_records_db);
if (notify.context[0]) {
/* Party A is free or B-Free notify has been setup. */
misdn_cc_pbx_notify(record_id, &notify);
}
} else {
AST_LIST_UNLOCK(&misdn_cc_records_db);
}
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Handle the FACILITY CCBSRemoteUserFree message.
*
* \param port Logical port number.
* \param facility Facility ie contents.
*
* \return Nothing
*/
static void misdn_cc_handle_remote_user_free(int port, const struct FacParm *facility)
{
struct misdn_cc_record *cc_record;
struct misdn_cc_notify notify;
long record_id;
AST_LIST_LOCK(&misdn_cc_records_db);
cc_record = misdn_cc_find_by_reference(port, facility->u.CCBSRemoteUserFree.CCBSReference);
if (cc_record) {
notify = cc_record->remote_user_free;
record_id = cc_record->record_id;
AST_LIST_UNLOCK(&misdn_cc_records_db);
misdn_cc_pbx_notify(record_id, &notify);
} else {
AST_LIST_UNLOCK(&misdn_cc_records_db);
}
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Handle the FACILITY CCBSBFree message.
*
* \param port Logical port number.
* \param facility Facility ie contents.
*
* \return Nothing
*/
static void misdn_cc_handle_b_free(int port, const struct FacParm *facility)
{
struct misdn_cc_record *cc_record;
struct misdn_cc_notify notify;
long record_id;
AST_LIST_LOCK(&misdn_cc_records_db);
cc_record = misdn_cc_find_by_reference(port, facility->u.CCBSBFree.CCBSReference);
if (cc_record && cc_record->b_free.context[0]) {
/* B-Free notify has been setup. */
notify = cc_record->b_free;
record_id = cc_record->record_id;
AST_LIST_UNLOCK(&misdn_cc_records_db);
misdn_cc_pbx_notify(record_id, &notify);
} else {
AST_LIST_UNLOCK(&misdn_cc_records_db);
}
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
/*!
* \internal
* \brief Handle the incoming facility ie contents
*
* \param event Message type facility ie came in on
* \param bc B channel control structure message came in on
* \param ch Associated channel call record
*
* \return Nothing
*/
static void misdn_facility_ie_handler(enum event_e event, struct misdn_bchannel *bc, struct chan_list *ch)
{
#if defined(AST_MISDN_ENHANCEMENTS)
const char *diagnostic_msg;
struct misdn_cc_record *cc_record;
char buf[32];
struct misdn_party_id party_id;
long new_record_id;
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
print_facility(&bc->fac_in, bc);
switch (bc->fac_in.Function) {
#if defined(AST_MISDN_ENHANCEMENTS)
case Fac_ActivationDiversion:
switch (bc->fac_in.u.ActivationDiversion.ComponentType) {
case FacComponent_Result:
/* Positive ACK to activation */
/* We don't handle this yet */
break;
default:
chan_misdn_log(0, bc->port," --> not yet handled: facility type:0x%04X\n",
bc->fac_in.Function);
break;
}
break;
case Fac_DeactivationDiversion:
switch (bc->fac_in.u.DeactivationDiversion.ComponentType) {
case FacComponent_Result:
/* Positive ACK to deactivation */
/* We don't handle this yet */
break;
default:
chan_misdn_log(0, bc->port," --> not yet handled: facility type:0x%04X\n",
bc->fac_in.Function);
break;
}
break;
case Fac_ActivationStatusNotificationDiv:
/* Sent to other MSN numbers on the line when a user activates call forwarding. */
/* Sent in the first call control message of an outgoing call from the served user. */
/* We do not have anything to do for this message. */
break;
case Fac_DeactivationStatusNotificationDiv:
/* Sent to other MSN numbers on the line when a user deactivates call forwarding. */
/* We do not have anything to do for this message. */
break;
#if 0 /* We don't handle this yet */
case Fac_InterrogationDiversion:
/* We don't handle this yet */
break;
case Fac_InterrogateServedUserNumbers:
/* We don't handle this yet */
break;
#endif /* We don't handle this yet */
case Fac_DiversionInformation:
/* Sent to the served user when a call is forwarded. */
/* We do not have anything to do for this message. */
break;
case Fac_CallDeflection:
if (ch && ch->ast) {
switch (bc->fac_in.u.CallDeflection.ComponentType) {
case FacComponent_Invoke:
ast_copy_string(bc->redirecting.from.number, bc->dialed.number,
sizeof(bc->redirecting.from.number));
bc->redirecting.from.name[0] = 0;
bc->redirecting.from.number_plan = bc->dialed.number_plan;
bc->redirecting.from.number_type = bc->dialed.number_type;
bc->redirecting.from.screening = 0;/* Unscreened */
if (bc->fac_in.u.CallDeflection.Component.Invoke.PresentationAllowedToDivertedToUserPresent) {
bc->redirecting.from.presentation =
bc->fac_in.u.CallDeflection.Component.Invoke.PresentationAllowedToDivertedToUser
? 0 /* Allowed */ : 1 /* Restricted */;
} else {
bc->redirecting.from.presentation = 0;/* Allowed */
}
/* Add configured prefix to the call deflection number */
memset(&party_id, 0, sizeof(party_id));
misdn_PartyNumber_extract(&party_id,
&bc->fac_in.u.CallDeflection.Component.Invoke.Deflection.Party);
misdn_add_number_prefix(bc->port, party_id.number_type,
party_id.number, sizeof(party_id.number));
//party_id.presentation = 0;/* Allowed */
//party_id.screening = 0;/* Unscreened */
bc->redirecting.to = party_id;
++bc->redirecting.count;
bc->redirecting.reason = mISDN_REDIRECTING_REASON_DEFLECTION;
Enhancements to connected line and redirecting work. From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
misdn_copy_redirecting_to_ast(ch->ast, &bc->redirecting, bc->incoming_cid_tag);
ast_channel_call_forward_set(ch->ast, bc->redirecting.to.number);
/* Send back positive ACK */
#if 1
/*
* Since there are no return result arguments it must be a
* generic result message. ETSI 300-196
*/
bc->fac_out.Function = Fac_RESULT;
bc->fac_out.u.RESULT.InvokeID = bc->fac_in.u.CallDeflection.InvokeID;
#else
bc->fac_out.Function = Fac_CallDeflection;
bc->fac_out.u.CallDeflection.InvokeID = bc->fac_in.u.CallDeflection.InvokeID;
bc->fac_out.u.CallDeflection.ComponentType = FacComponent_Result;
#endif
print_facility(&bc->fac_out, bc);
misdn_lib_send_event(bc, EVENT_DISCONNECT);
/* This line is BUSY to further attempts by this dialing attempt. */
ast_queue_control(ch->ast, AST_CONTROL_BUSY);
break;
case FacComponent_Result:
/* Positive ACK to call deflection */
/*
* Sent in DISCONNECT or FACILITY message depending upon network option.
* It is in the FACILITY message if the call is still offered to the user
* while trying to alert the deflected to party.
*/
/* Ignore the ACK */
break;
default:
break;
}
}
break;
#if 0 /* We don't handle this yet */
case Fac_CallRerouteing:
/* Private-Public ISDN interworking message */
/* We don't handle this yet */
break;
#endif /* We don't handle this yet */
case Fac_DivertingLegInformation1:
/* Private-Public ISDN interworking message */
bc->div_leg_3_rx_wanted = 0;
if (ch && ch->ast) {
bc->redirecting.reason =
diversion_reason_to_misdn(bc->fac_in.u.DivertingLegInformation1.DiversionReason);
if (bc->fac_in.u.DivertingLegInformation1.DivertedToPresent) {
misdn_PresentedNumberUnscreened_extract(&bc->redirecting.to,
&bc->fac_in.u.DivertingLegInformation1.DivertedTo);
Merged revisions 49313 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r49313 | crichter | 2007-01-03 10:06:50 +0100 (Mi, 03 Jan 2007) | 41 lines Merged revisions 48319,48321,48467,48552,48576,49135,49303 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) | 1 line changed a few debugs to higher debug levels ........ r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) | 1 line added the export and import of the MISDN_ADDRESS_COMPLETE Variable to inidcate wether the extension is already completely dialed or if there might come additional digits by information elements. also added some docs for that. ........ r48467 | crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line removed FIXUP state. added check for channel allocation conflict when we create a setup while the other site creates a setup on the same channel, besides the check we resolve this conflict. ........ r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18 Dez 2006) | 1 line when our PTP Partner sends us a SETUP with a preselected channel we just accept it, even when we're NT. added some checks for segfaults. ........ r48576 | crichter | 2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line when we reject a channel, because it's in use already, we shouldn't process the setup anymore. made the channel allocation a bit easier and more understandable, removed a few unused lines ........ r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02 Jan 2007) | 1 line added check for channel ranges in the set/empty channel functions. set pmp_l1_check default to no. added misdn restart pid cli command. added cleaning of channel when we send a RELEASE_COMPLETE. ........ r49303 | crichter | 2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines * Added check for bridging in misdn_call to avoid setting echocancellation when 2 mISDN channels are involved and when bridging is set. That lead to a kernel panic before under different situations, because we switched about 2 times between hardware bridging and echocancelation * readded MISDN_URATE variable which got lost before, this should make app_v110 work again * fixed typo ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49321 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-03 11:15:02 +00:00
/* Add configured prefix to redirecting.to.number */
misdn_add_number_prefix(bc->port, bc->redirecting.to.number_type,
bc->redirecting.to.number, sizeof(bc->redirecting.to.number));
} else {
bc->redirecting.to.number[0] = '\0';
bc->redirecting.to.number_plan = NUMPLAN_ISDN;
bc->redirecting.to.number_type = NUMTYPE_UNKNOWN;
bc->redirecting.to.presentation = 1;/* restricted */
bc->redirecting.to.screening = 0;/* unscreened */
}
Enhancements to connected line and redirecting work. From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
misdn_copy_redirecting_to_ast(ch->ast, &bc->redirecting, bc->incoming_cid_tag);
bc->div_leg_3_rx_wanted = 1;
}
break;
case Fac_DivertingLegInformation2:
/* Private-Public ISDN interworking message */
switch (event) {
case EVENT_SETUP:
/* Comes in on a SETUP with redirecting.from information */
bc->div_leg_3_tx_pending = 1;
if (ch && ch->ast) {
/*
* Setup the redirecting.to informtion so we can identify
* if the user wants to manually supply the COLR for this
* redirected to number if further redirects could happen.
*
* All the user needs to do is set the REDIRECTING(to-pres)
* to the COLR and REDIRECTING(to-num) = ${EXTEN} to be safe
* after determining that the incoming call was redirected by
* checking if there is a REDIRECTING(from-num).
*/
ast_copy_string(bc->redirecting.to.number, bc->dialed.number,
sizeof(bc->redirecting.to.number));
bc->redirecting.to.number_plan = bc->dialed.number_plan;
bc->redirecting.to.number_type = bc->dialed.number_type;
bc->redirecting.to.presentation = 1;/* restricted */
bc->redirecting.to.screening = 0;/* unscreened */
bc->redirecting.reason =
diversion_reason_to_misdn(bc->fac_in.u.DivertingLegInformation2.DiversionReason);
bc->redirecting.count = bc->fac_in.u.DivertingLegInformation2.DiversionCounter;
if (bc->fac_in.u.DivertingLegInformation2.DivertingPresent) {
/* This information is redundant if there was a redirecting ie in the SETUP. */
misdn_PresentedNumberUnscreened_extract(&bc->redirecting.from,
&bc->fac_in.u.DivertingLegInformation2.Diverting);
/* Add configured prefix to redirecting.from.number */
misdn_add_number_prefix(bc->port, bc->redirecting.from.number_type,
bc->redirecting.from.number, sizeof(bc->redirecting.from.number));
}
#if 0
if (bc->fac_in.u.DivertingLegInformation2.OriginalCalledPresent) {
/* We have no place to put the OriginalCalled number */
}
#endif
Enhancements to connected line and redirecting work. From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
misdn_copy_redirecting_to_ast(ch->ast, &bc->redirecting, bc->incoming_cid_tag);
}
break;
default:
chan_misdn_log(0, bc->port," --> Expected in a SETUP message: facility type:0x%04X\n",
bc->fac_in.Function);
break;
}
break;
case Fac_DivertingLegInformation3:
/* Private-Public ISDN interworking message */
if (bc->div_leg_3_rx_wanted) {
bc->div_leg_3_rx_wanted = 0;
if (ch && ch->ast) {
Add private representation of caller, connected and redirecting party ids. This patch adds the feature "Private representation of caller, connected and redirecting party ids", as previously discussed with us (DATUS) and Digium. 1. Feature motivation Until now it is quite difficult to modify a party number or name which can only be seen by exactly one particular instantiated technology channel subscriber. One example where a modified party number or name on one channel is spread over several channels are supplementary services like call transfer or pickup. To implement these features Asterisk internally copies caller and connected ids from one channel to another. Another example are extension subscriptions. The monitoring entities (watchers) are notified of state changes and - if desired - of party numbers or names which represent the involving call parties. One major feature where a private representation of party names is essentially needed, i.e. where a party name shall be exclusively signaled to only one particular user, is a private user-specific name resolution for party numbers. A lookup in a private destination-dependent telephone book shall provide party names which cannot be seen by any other user at any time. 2. Feature Description This feature comes along with the implementation of additional private party id elements for caller id, connected id and redirecting ids inside Asterisk channels. The private party id elements can be read or set by the user using Asterisk dialplan functions. When a technology channel is initiating a call, receives an internal connected-line update event, or receives an internal redirecting update event, it merges the corresponding public id with the private id to create an effective party id. The effective party id is then used for protocol signaling. The channel technologies which initially support the private id representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and PRI (chan_dahdi). Once a private name or number on a channel is set and (implicitly) made valid, it is generally used for any further protocol signaling until it is rewritten or invalidated. To simplify the invalidation of private ids all internally generated connected/redirecting update events and also all connected/redirecting update events which are generated by technology channels -- receiving regarding protocol information - automatically trigger the invalidation of private ids. If not using the private party id representation feature at all, i.e. if using only the 'regular' caller-id, connected and redirecting related functions, the current characteristic of Asterisk is not affected by the new extended functionality. 3. User interface Description To grant access to the private name and number representation from the Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan functions are extended by the following data types. The formats of these data types are equal to the corresponding regular 'non-private' already existing data types: CALLERID: priv-all priv-name priv-name-valid priv-name-charset priv-name-pres priv-num priv-num-valid priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE: priv-name priv-name-valid priv-name-pres priv-name-charset priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag REDIRECTING: priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd priv-orig-tag priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd priv-to-tag Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2030/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-10 19:54:55 +00:00
struct ast_party_redirecting redirecting;
ast_channel_redirecting(ch->ast)->to.number.presentation =
bc->fac_in.u.DivertingLegInformation3.PresentationAllowedIndicator
? AST_PRES_ALLOWED | AST_PRES_USER_NUMBER_UNSCREENED
: AST_PRES_RESTRICTED | AST_PRES_USER_NUMBER_UNSCREENED;
Add private representation of caller, connected and redirecting party ids. This patch adds the feature "Private representation of caller, connected and redirecting party ids", as previously discussed with us (DATUS) and Digium. 1. Feature motivation Until now it is quite difficult to modify a party number or name which can only be seen by exactly one particular instantiated technology channel subscriber. One example where a modified party number or name on one channel is spread over several channels are supplementary services like call transfer or pickup. To implement these features Asterisk internally copies caller and connected ids from one channel to another. Another example are extension subscriptions. The monitoring entities (watchers) are notified of state changes and - if desired - of party numbers or names which represent the involving call parties. One major feature where a private representation of party names is essentially needed, i.e. where a party name shall be exclusively signaled to only one particular user, is a private user-specific name resolution for party numbers. A lookup in a private destination-dependent telephone book shall provide party names which cannot be seen by any other user at any time. 2. Feature Description This feature comes along with the implementation of additional private party id elements for caller id, connected id and redirecting ids inside Asterisk channels. The private party id elements can be read or set by the user using Asterisk dialplan functions. When a technology channel is initiating a call, receives an internal connected-line update event, or receives an internal redirecting update event, it merges the corresponding public id with the private id to create an effective party id. The effective party id is then used for protocol signaling. The channel technologies which initially support the private id representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and PRI (chan_dahdi). Once a private name or number on a channel is set and (implicitly) made valid, it is generally used for any further protocol signaling until it is rewritten or invalidated. To simplify the invalidation of private ids all internally generated connected/redirecting update events and also all connected/redirecting update events which are generated by technology channels -- receiving regarding protocol information - automatically trigger the invalidation of private ids. If not using the private party id representation feature at all, i.e. if using only the 'regular' caller-id, connected and redirecting related functions, the current characteristic of Asterisk is not affected by the new extended functionality. 3. User interface Description To grant access to the private name and number representation from the Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan functions are extended by the following data types. The formats of these data types are equal to the corresponding regular 'non-private' already existing data types: CALLERID: priv-all priv-name priv-name-valid priv-name-charset priv-name-pres priv-num priv-num-valid priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE: priv-name priv-name-valid priv-name-pres priv-name-charset priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag REDIRECTING: priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd priv-orig-tag priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd priv-to-tag Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2030/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-10 19:54:55 +00:00
ast_party_redirecting_init(&redirecting);
ast_party_redirecting_copy(&redirecting, ast_channel_redirecting(ch->ast));
/*
* Reset any earlier private redirecting id representations and
* make sure that it is invalidated at the remote end.
*/
ast_party_id_reset(&redirecting.priv_orig);
ast_party_id_reset(&redirecting.priv_from);
ast_party_id_reset(&redirecting.priv_to);
ast_channel_queue_redirecting_update(ch->ast, &redirecting, NULL);
ast_party_redirecting_free(&redirecting);
}
}
break;
#else /* !defined(AST_MISDN_ENHANCEMENTS) */
case Fac_CD:
if (ch && ch->ast) {
ast_copy_string(bc->redirecting.from.number, bc->dialed.number,
sizeof(bc->redirecting.from.number));
bc->redirecting.from.name[0] = 0;
bc->redirecting.from.number_plan = bc->dialed.number_plan;
bc->redirecting.from.number_type = bc->dialed.number_type;
bc->redirecting.from.screening = 0;/* Unscreened */
bc->redirecting.from.presentation =
bc->fac_in.u.CDeflection.PresentationAllowed
? 0 /* Allowed */ : 1 /* Restricted */;
ast_copy_string(bc->redirecting.to.number,
(char *) bc->fac_in.u.CDeflection.DeflectedToNumber,
sizeof(bc->redirecting.to.number));
bc->redirecting.to.name[0] = 0;
bc->redirecting.to.number_plan = NUMPLAN_UNKNOWN;
bc->redirecting.to.number_type = NUMTYPE_UNKNOWN;
bc->redirecting.to.presentation = 0;/* Allowed */
bc->redirecting.to.screening = 0;/* Unscreened */
++bc->redirecting.count;
bc->redirecting.reason = mISDN_REDIRECTING_REASON_DEFLECTION;
Enhancements to connected line and redirecting work. From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
misdn_copy_redirecting_to_ast(ch->ast, &bc->redirecting, bc->incoming_cid_tag);
ast_channel_call_forward_set(ch->ast, bc->redirecting.to.number);
misdn_lib_send_event(bc, EVENT_DISCONNECT);
/* This line is BUSY to further attempts by this dialing attempt. */
ast_queue_control(ch->ast, AST_CONTROL_BUSY);
}
break;
#endif /* !defined(AST_MISDN_ENHANCEMENTS) */
case Fac_AOCDCurrency:
if (ch && ch->ast) {
bc->AOCDtype = Fac_AOCDCurrency;
memcpy(&bc->AOCD.currency, &bc->fac_in.u.AOCDcur, sizeof(bc->AOCD.currency));
bc->AOCD_need_export = 1;
export_aoc_vars(ch->originator, ch->ast, bc);
}
break;
case Fac_AOCDChargingUnit:
if (ch && ch->ast) {
bc->AOCDtype = Fac_AOCDChargingUnit;
memcpy(&bc->AOCD.chargingUnit, &bc->fac_in.u.AOCDchu, sizeof(bc->AOCD.chargingUnit));
bc->AOCD_need_export = 1;
export_aoc_vars(ch->originator, ch->ast, bc);
}
break;
#if defined(AST_MISDN_ENHANCEMENTS)
case Fac_ERROR:
diagnostic_msg = misdn_to_str_error_code(bc->fac_in.u.ERROR.errorValue);
chan_misdn_log(1, bc->port, " --> Facility error code: %s\n", diagnostic_msg);
switch (event) {
case EVENT_DISCONNECT:
case EVENT_RELEASE:
case EVENT_RELEASE_COMPLETE:
/* Possible call failure as a result of Fac_CCBSCall/Fac_CCBS_T_Call */
if (ch && ch->peer) {
misdn_cc_set_peer_var(ch->peer, MISDN_ERROR_MSG, diagnostic_msg);
}
break;
default:
break;
}
AST_LIST_LOCK(&misdn_cc_records_db);
cc_record = misdn_cc_find_by_invoke(bc->port, bc->fac_in.u.ERROR.invokeId);
if (cc_record) {
cc_record->outstanding_message = 0;
cc_record->error_code = bc->fac_in.u.ERROR.errorValue;
}
AST_LIST_UNLOCK(&misdn_cc_records_db);
break;
case Fac_REJECT:
diagnostic_msg = misdn_to_str_reject_code(bc->fac_in.u.REJECT.Code);
chan_misdn_log(1, bc->port, " --> Facility reject code: %s\n", diagnostic_msg);
switch (event) {
case EVENT_DISCONNECT:
case EVENT_RELEASE:
case EVENT_RELEASE_COMPLETE:
/* Possible call failure as a result of Fac_CCBSCall/Fac_CCBS_T_Call */
if (ch && ch->peer) {
misdn_cc_set_peer_var(ch->peer, MISDN_ERROR_MSG, diagnostic_msg);
}
break;
default:
break;
}
if (bc->fac_in.u.REJECT.InvokeIDPresent) {
AST_LIST_LOCK(&misdn_cc_records_db);
cc_record = misdn_cc_find_by_invoke(bc->port, bc->fac_in.u.REJECT.InvokeID);
if (cc_record) {
cc_record->outstanding_message = 0;
cc_record->reject_code = bc->fac_in.u.REJECT.Code;
}
AST_LIST_UNLOCK(&misdn_cc_records_db);
}
break;
case Fac_RESULT:
AST_LIST_LOCK(&misdn_cc_records_db);
cc_record = misdn_cc_find_by_invoke(bc->port, bc->fac_in.u.RESULT.InvokeID);
if (cc_record) {
cc_record->outstanding_message = 0;
}
AST_LIST_UNLOCK(&misdn_cc_records_db);
break;
#if 0 /* We don't handle this yet */
case Fac_EctExecute:
/* We don't handle this yet */
break;
case Fac_ExplicitEctExecute:
/* We don't handle this yet */
break;
case Fac_EctLinkIdRequest:
/* We don't handle this yet */
break;
#endif /* We don't handle this yet */
case Fac_SubaddressTransfer:
/* We do not have anything to do for this message since we do not handle subaddreses. */
break;
case Fac_RequestSubaddress:
Merged revisions 287017 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r287017 | rmudgett | 2010-09-15 15:53:38 -0500 (Wed, 15 Sep 2010) | 65 lines Merged revision 287014 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r287014 | rmudgett | 2010-09-15 15:32:24 -0500 (Wed, 15 Sep 2010) | 58 lines The handling of call transfer signaling for mISDN PTMP is not fully implemented. The handling of call transfer signaling for mISDN PTMP is not fully implemented. The signaling of number updates with ISDN/DSS1 ECT supplementary services (ETS 300 369-1) comes along with a notification indicator IE and redirection number IE for PTMP. The implementation in the current Asterisk mISDN channel unfortunately can handle these information elements only in a NOTIFY message. These information elements are also signaled in a FACILTY message with a RequestSubaddress facility, when the subscriber is already in the active state (see 9.2.4 and 9.2.5 of ETS 300 369-1). ********** abe_2526_ast.patch * Added support to handle the notification indicator IE and redirection number IE with the RequestSubaddress facility. * Made misdn_update_connected_line() send a NOTIFY message if Asterisk originated the call and it is not connected yet. * Made misdn_update_connected_line() send a FACILITY message if the call is already connected. This patch requires the presence of the associated mISDN patches to compile. I had to enhance mISDN to allow the notification indicator IE and the redirection number IE to be used with a FACILITY message. Earlier versions of the Digium enhanced mISDN are no longer going to work. ********** abe_2526_misdn.patch * Made an incoming FACILITY message allow the presence of the notification indicator IE and the redirection number IE. ********** abe_2526_misdnuser_v3.patch * Added support to send and receive a FACILITY message with the notification indicator IE and the redirection number IE. * Added the ability to send a NOTIFY message in PTMP/NT mode to all responding subcalls in Q.931 states 6, 7, 8, 9, and 25. ********** Patches: abe_2526_ast.patch uploaded by rmudgett (license 664) abe_2526_misdn.patch uploaded by rmudgett (license 664) abe_2526_misdnuser_v3.patch uploaded by rmudgett (license 664) Tested by: rmudgett and reporter JIRA SWP-2146 JIRA ABE-2526 .......... ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287018 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-15 20:56:21 +00:00
/*
* We do not have anything to do for this message since we do not handle subaddreses.
* However, we do care about some other ie's that should be present.
*/
if (bc->redirecting.to_changed) {
/* Add configured prefix to redirecting.to.number */
misdn_add_number_prefix(bc->port, bc->redirecting.to.number_type,
bc->redirecting.to.number, sizeof(bc->redirecting.to.number));
}
switch (bc->notify_description_code) {
case mISDN_NOTIFY_CODE_INVALID:
/* Notify ie was not present. */
bc->redirecting.to_changed = 0;
break;
case mISDN_NOTIFY_CODE_CALL_TRANSFER_ALERTING:
/*
* It would be preferable to update the connected line information
* only when the message callStatus is active. However, the
* optional redirection number may not be present in the active
* message if an alerting message were received earlier.
*
* The consequences if we wind up sending two updates is benign.
* The other end will think that it got transferred twice.
*/
if (!bc->redirecting.to_changed) {
break;
}
bc->redirecting.to_changed = 0;
if (!ch || !ch->ast) {
break;
}
misdn_update_remote_party(ch->ast, &bc->redirecting.to,
AST_CONNECTED_LINE_UPDATE_SOURCE_TRANSFER_ALERTING,
bc->incoming_cid_tag);
break;
case mISDN_NOTIFY_CODE_CALL_TRANSFER_ACTIVE:
if (!bc->redirecting.to_changed) {
break;
}
bc->redirecting.to_changed = 0;
if (!ch || !ch->ast) {
break;
}
misdn_update_remote_party(ch->ast, &bc->redirecting.to,
AST_CONNECTED_LINE_UPDATE_SOURCE_TRANSFER, bc->incoming_cid_tag);
break;
default:
bc->redirecting.to_changed = 0;
chan_misdn_log(0, bc->port," --> not yet handled: notify code:0x%02X\n",
bc->notify_description_code);
break;
}
bc->notify_description_code = mISDN_NOTIFY_CODE_INVALID;
break;
case Fac_EctInform:
/* Private-Public ISDN interworking message */
if (ch && ch->ast && bc->fac_in.u.EctInform.RedirectionPresent) {
/* Add configured prefix to the redirection number */
memset(&party_id, 0, sizeof(party_id));
misdn_PresentedNumberUnscreened_extract(&party_id,
&bc->fac_in.u.EctInform.Redirection);
misdn_add_number_prefix(bc->port, party_id.number_type,
party_id.number, sizeof(party_id.number));
/*
* It would be preferable to update the connected line information
* only when the message callStatus is active. However, the
* optional redirection number may not be present in the active
* message if an alerting message were received earlier.
*
* The consequences if we wind up sending two updates is benign.
* The other end will think that it got transferred twice.
*/
misdn_update_remote_party(ch->ast, &party_id,
(bc->fac_in.u.EctInform.Status == 0 /* alerting */)
? AST_CONNECTED_LINE_UPDATE_SOURCE_TRANSFER_ALERTING
Enhancements to connected line and redirecting work. From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
: AST_CONNECTED_LINE_UPDATE_SOURCE_TRANSFER,
bc->incoming_cid_tag);
}
break;
#if 0 /* We don't handle this yet */
case Fac_EctLoopTest:
/* The use of this message is unclear on how it works to detect loops. */
/* We don't handle this yet */
break;
#endif /* We don't handle this yet */
case Fac_CallInfoRetain:
switch (event) {
case EVENT_ALERTING:
case EVENT_DISCONNECT:
/* CCBS/CCNR is available */
if (ch && ch->peer) {
AST_LIST_LOCK(&misdn_cc_records_db);
if (ch->record_id == -1) {
cc_record = misdn_cc_new();
} else {
/*
* We are doing a call-completion attempt
* or the switch is sending us extra call-completion
* availability indications (erroneously?).
*
* Assume that the network request retention option
* is not on and that the current call-completion
* request is disabled.
*/
cc_record = misdn_cc_find_by_id(ch->record_id);
if (cc_record) {
if (cc_record->ptp && cc_record->mode.ptp.bc) {
/*
* What? We are getting mixed messages from the
* switch. We are currently setup for
* point-to-point. Now we are switching to
* point-to-multipoint.
*
* Close the call-completion signaling link
*/
cc_record->mode.ptp.bc->fac_out.Function = Fac_None;
cc_record->mode.ptp.bc->out_cause = AST_CAUSE_NORMAL_CLEARING;
misdn_lib_send_event(cc_record->mode.ptp.bc, EVENT_RELEASE_COMPLETE);
}
/*
* Resetup the existing record for a possible new
* call-completion request.
*/
new_record_id = misdn_cc_record_id_new();
if (new_record_id < 0) {
/* Looks like we must keep the old id anyway. */
} else {
cc_record->record_id = new_record_id;
ch->record_id = new_record_id;
}
cc_record->ptp = 0;
cc_record->port = bc->port;
memset(&cc_record->mode, 0, sizeof(cc_record->mode));
cc_record->mode.ptmp.linkage_id = bc->fac_in.u.CallInfoRetain.CallLinkageID;
cc_record->invoke_id = ++misdn_invoke_id;
cc_record->activated = 0;
cc_record->outstanding_message = 0;
cc_record->activation_requested = 0;
cc_record->error_code = FacError_None;
cc_record->reject_code = FacReject_None;
memset(&cc_record->remote_user_free, 0, sizeof(cc_record->remote_user_free));
memset(&cc_record->b_free, 0, sizeof(cc_record->b_free));
cc_record->time_created = time(NULL);
cc_record = NULL;
} else {
/*
* Where did the record go? We will have to recapture
* the call setup information. Unfortunately, some
* setup information may have been changed.
*/
ch->record_id = -1;
cc_record = misdn_cc_new();
}
}
if (cc_record) {
ch->record_id = cc_record->record_id;
cc_record->ptp = 0;
cc_record->port = bc->port;
cc_record->mode.ptmp.linkage_id = bc->fac_in.u.CallInfoRetain.CallLinkageID;
/* Record call information for possible call-completion attempt. */
cc_record->redial.caller = bc->caller;
cc_record->redial.dialed = bc->dialed;
cc_record->redial.setup_bc_hlc_llc = bc->setup_bc_hlc_llc;
cc_record->redial.capability = bc->capability;
cc_record->redial.hdlc = bc->hdlc;
}
AST_LIST_UNLOCK(&misdn_cc_records_db);
/* Set MISDN_CC_RECORD_ID in original channel */
if (ch->record_id != -1) {
snprintf(buf, sizeof(buf), "%ld", ch->record_id);
} else {
buf[0] = 0;
}
misdn_cc_set_peer_var(ch->peer, MISDN_CC_RECORD_ID, buf);
}
break;
default:
chan_misdn_log(0, bc->port,
" --> Expected in a DISCONNECT or ALERTING message: facility type:0x%04X\n",
bc->fac_in.Function);
break;
}
break;
case Fac_CCBS_T_Call:
case Fac_CCBSCall:
switch (event) {
case EVENT_SETUP:
/*
* This is a call completion retry call.
* If we had anything to do we would do it here.
*/
break;
default:
chan_misdn_log(0, bc->port, " --> Expected in a SETUP message: facility type:0x%04X\n",
bc->fac_in.Function);
break;
}
break;
case Fac_CCBSDeactivate:
switch (bc->fac_in.u.CCBSDeactivate.ComponentType) {
case FacComponent_Result:
AST_LIST_LOCK(&misdn_cc_records_db);
cc_record = misdn_cc_find_by_invoke(bc->port, bc->fac_in.u.CCBSDeactivate.InvokeID);
if (cc_record) {
cc_record->outstanding_message = 0;
}
AST_LIST_UNLOCK(&misdn_cc_records_db);
break;
default:
chan_misdn_log(0, bc->port, " --> not yet handled: facility type:0x%04X\n",
bc->fac_in.Function);
break;
}
break;
case Fac_CCBSErase:
AST_LIST_LOCK(&misdn_cc_records_db);
cc_record = misdn_cc_find_by_reference(bc->port, bc->fac_in.u.CCBSErase.CCBSReference);
if (cc_record) {
misdn_cc_delete(cc_record);
}
AST_LIST_UNLOCK(&misdn_cc_records_db);
break;
case Fac_CCBSRemoteUserFree:
misdn_cc_handle_remote_user_free(bc->port, &bc->fac_in);
break;
case Fac_CCBSBFree:
misdn_cc_handle_b_free(bc->port, &bc->fac_in);
break;
case Fac_CCBSStatusRequest:
misdn_cc_handle_ccbs_status_request(bc->port, &bc->fac_in);
break;
case Fac_EraseCallLinkageID:
AST_LIST_LOCK(&misdn_cc_records_db);
cc_record = misdn_cc_find_by_linkage(bc->port,
bc->fac_in.u.EraseCallLinkageID.CallLinkageID);
if (cc_record && !cc_record->activation_requested) {
/*
* The T-RETENTION timer expired before we requested
* call completion activation. Call completion is no
* longer available.
*/
misdn_cc_delete(cc_record);
}
AST_LIST_UNLOCK(&misdn_cc_records_db);
break;
case Fac_CCBSStopAlerting:
/* We do not have anything to do for this message. */
break;
case Fac_CCBSRequest:
case Fac_CCNRRequest:
switch (bc->fac_in.u.CCBSRequest.ComponentType) {
case FacComponent_Result:
AST_LIST_LOCK(&misdn_cc_records_db);
cc_record = misdn_cc_find_by_invoke(bc->port, bc->fac_in.u.CCBSRequest.InvokeID);
if (cc_record && !cc_record->ptp) {
cc_record->outstanding_message = 0;
cc_record->activated = 1;
cc_record->mode.ptmp.recall_mode = bc->fac_in.u.CCBSRequest.Component.Result.RecallMode;
cc_record->mode.ptmp.reference_id = bc->fac_in.u.CCBSRequest.Component.Result.CCBSReference;
}
AST_LIST_UNLOCK(&misdn_cc_records_db);
break;
default:
chan_misdn_log(0, bc->port, " --> not yet handled: facility type:0x%04X\n",
bc->fac_in.Function);
break;
}
break;
#if 0 /* We don't handle this yet */
case Fac_CCBSInterrogate:
case Fac_CCNRInterrogate:
/* We don't handle this yet */
break;
case Fac_StatusRequest:
/* We don't handle this yet */
break;
#endif /* We don't handle this yet */
#if 0 /* We don't handle this yet */
case Fac_CCBS_T_Suspend:
case Fac_CCBS_T_Resume:
/* We don't handle this yet */
break;
#endif /* We don't handle this yet */
case Fac_CCBS_T_RemoteUserFree:
misdn_cc_handle_T_remote_user_free(bc);
break;
case Fac_CCBS_T_Available:
switch (event) {
case EVENT_ALERTING:
case EVENT_DISCONNECT:
/* CCBS-T/CCNR-T is available */
if (ch && ch->peer) {
int set_id = 1;
AST_LIST_LOCK(&misdn_cc_records_db);
if (ch->record_id == -1) {
cc_record = misdn_cc_new();
} else {
/*
* We are doing a call-completion attempt
* or the switch is sending us extra call-completion
* availability indications (erroneously?).
*/
cc_record = misdn_cc_find_by_id(ch->record_id);
if (cc_record) {
if (cc_record->ptp && cc_record->mode.ptp.retention_enabled) {
/*
* Call-completion is still activated.
* The user does not have to request it again.
*/
chan_misdn_log(1, bc->port, " --> Call-completion request retention option is enabled\n");
set_id = 0;
} else {
if (cc_record->ptp && cc_record->mode.ptp.bc) {
/*
* The network request retention option
* is not on and the current call-completion
* request is to be disabled.
*
* We should get here only if EVENT_DISCONNECT
*
* Close the call-completion signaling link
*/
cc_record->mode.ptp.bc->fac_out.Function = Fac_None;
cc_record->mode.ptp.bc->out_cause = AST_CAUSE_NORMAL_CLEARING;
misdn_lib_send_event(cc_record->mode.ptp.bc, EVENT_RELEASE_COMPLETE);
}
/*
* Resetup the existing record for a possible new
* call-completion request.
*/
new_record_id = misdn_cc_record_id_new();
if (new_record_id < 0) {
/* Looks like we must keep the old id anyway. */
} else {
cc_record->record_id = new_record_id;
ch->record_id = new_record_id;
}
cc_record->ptp = 1;
cc_record->port = bc->port;
memset(&cc_record->mode, 0, sizeof(cc_record->mode));
cc_record->invoke_id = ++misdn_invoke_id;
cc_record->activated = 0;
cc_record->outstanding_message = 0;
cc_record->activation_requested = 0;
cc_record->error_code = FacError_None;
cc_record->reject_code = FacReject_None;
memset(&cc_record->remote_user_free, 0, sizeof(cc_record->remote_user_free));
memset(&cc_record->b_free, 0, sizeof(cc_record->b_free));
cc_record->time_created = time(NULL);
}
cc_record = NULL;
} else {
/*
* Where did the record go? We will have to recapture
* the call setup information. Unfortunately, some
* setup information may have been changed.
*/
ch->record_id = -1;
cc_record = misdn_cc_new();
}
}
if (cc_record) {
ch->record_id = cc_record->record_id;
cc_record->ptp = 1;
cc_record->port = bc->port;
/* Record call information for possible call-completion attempt. */
cc_record->redial.caller = bc->caller;
cc_record->redial.dialed = bc->dialed;
cc_record->redial.setup_bc_hlc_llc = bc->setup_bc_hlc_llc;
cc_record->redial.capability = bc->capability;
cc_record->redial.hdlc = bc->hdlc;
}
AST_LIST_UNLOCK(&misdn_cc_records_db);
/* Set MISDN_CC_RECORD_ID in original channel */
if (ch->record_id != -1 && set_id) {
snprintf(buf, sizeof(buf), "%ld", ch->record_id);
} else {
buf[0] = 0;
}
misdn_cc_set_peer_var(ch->peer, MISDN_CC_RECORD_ID, buf);
}
break;
default:
chan_misdn_log(0, bc->port,
" --> Expected in a DISCONNECT or ALERTING message: facility type:0x%04X\n",
bc->fac_in.Function);
break;
}
break;
case Fac_CCBS_T_Request:
case Fac_CCNR_T_Request:
switch (bc->fac_in.u.CCBS_T_Request.ComponentType) {
case FacComponent_Result:
AST_LIST_LOCK(&misdn_cc_records_db);
cc_record = misdn_cc_find_by_invoke(bc->port, bc->fac_in.u.CCBS_T_Request.InvokeID);
if (cc_record && cc_record->ptp) {
cc_record->outstanding_message = 0;
cc_record->activated = 1;
cc_record->mode.ptp.retention_enabled =
cc_record->mode.ptp.requested_retention
? bc->fac_in.u.CCBS_T_Request.Component.Result.RetentionSupported
? 1 : 0
: 0;
}
AST_LIST_UNLOCK(&misdn_cc_records_db);
break;
case FacComponent_Invoke:
/* We cannot be User-B in ptp mode. */
default:
chan_misdn_log(0, bc->port, " --> not yet handled: facility type:0x%04X\n",
bc->fac_in.Function);
break;
}
break;
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
case Fac_None:
break;
default:
chan_misdn_log(0, bc->port, " --> not yet handled: facility type:0x%04X\n",
bc->fac_in.Function);
break;
}
}
/*!
* \internal
* \brief Determine if the given dialed party matches our MSN.
* \since 1.8
*
* \param port ISDN port
* \param dialed Dialed party information of incoming call.
*
* \retval non-zero if MSN is valid.
* \retval 0 if MSN invalid.
*/
static int misdn_is_msn_valid(int port, const struct misdn_party_dialing *dialed)
{
char number[sizeof(dialed->number)];
ast_copy_string(number, dialed->number, sizeof(number));
misdn_add_number_prefix(port, dialed->number_type, number, sizeof(number));
return misdn_cfg_is_msn_valid(port, number);
}
/************************************************************/
/* Receive Events from isdn_lib here */
/************************************************************/
static enum event_response_e
cb_events(enum event_e event, struct misdn_bchannel *bc, void *user_data)
{
#if defined(AST_MISDN_ENHANCEMENTS)
struct misdn_cc_record *cc_record;
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
struct chan_list *held_ch;
struct chan_list *ch = find_chan_by_bc(bc);
if (event != EVENT_BCHAN_DATA && event != EVENT_TONE_GENERATE) {
int debuglevel = 1;
/* Debug Only Non-Bchan */
if (event == EVENT_CLEANUP && !user_data) {
debuglevel = 5;
}
chan_misdn_log(debuglevel, bc->port,
"I IND :%s caller:\"%s\" <%s> dialed:%s pid:%d state:%s\n",
manager_isdn_get_info(event),
bc->caller.name,
bc->caller.number,
bc->dialed.number,
bc->pid,
ch ? misdn_get_ch_state(ch) : "none");
if (debuglevel == 1) {
misdn_lib_log_ies(bc);
chan_misdn_log(4, bc->port, " --> bc_state:%s\n", bc_state2str(bc->bc_state));
}
}
if (!ch) {
switch(event) {
case EVENT_SETUP:
case EVENT_DISCONNECT:
case EVENT_RELEASE:
case EVENT_RELEASE_COMPLETE:
case EVENT_PORT_ALARM:
case EVENT_RETRIEVE:
case EVENT_NEW_BC:
case EVENT_FACILITY:
case EVENT_REGISTER:
break;
case EVENT_CLEANUP:
case EVENT_TONE_GENERATE:
case EVENT_BCHAN_DATA:
return -1;
default:
chan_misdn_log(1, bc->port, "Chan not existing at the moment bc->l3id:%x bc:%p event:%s port:%d channel:%d\n", bc->l3_id, bc, manager_isdn_get_info(event), bc->port, bc->channel);
return -1;
}
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
} else {
switch (event) {
case EVENT_TONE_GENERATE:
break;
case EVENT_DISCONNECT:
case EVENT_RELEASE:
case EVENT_RELEASE_COMPLETE:
case EVENT_CLEANUP:
case EVENT_TIMEOUT:
if (!ch->ast) {
chan_misdn_log(3, bc->port, "ast_hangup already called, so we have no ast ptr anymore in event(%s)\n", manager_isdn_get_info(event));
}
break;
default:
if (!ch->ast || !MISDN_ASTERISK_TECH_PVT(ch->ast)) {
if (event != EVENT_BCHAN_DATA) {
ast_log(LOG_NOTICE, "No Ast or No private Pointer in Event (%d:%s)\n", event, manager_isdn_get_info(event));
}
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
chan_list_unref(ch, "No Ast or Ast private pointer");
return -1;
}
break;
}
}
switch (event) {
case EVENT_PORT_ALARM:
{
int boa = 0;
misdn_cfg_get(bc->port, MISDN_CFG_ALARM_BLOCK, &boa, sizeof(boa));
if (boa) {
cb_log(1, bc->port, " --> blocking\n");
misdn_lib_port_block(bc->port);
}
}
break;
case EVENT_BCHAN_ACTIVATED:
break;
case EVENT_NEW_CHANNEL:
update_name(ch->ast,bc->port,bc->channel);
break;
case EVENT_NEW_L3ID:
ch->l3id=bc->l3_id;
ch->addr=bc->addr;
break;
case EVENT_NEW_BC:
if (!ch) {
ch = find_hold_call(bc);
}
if (!ch) {
ast_log(LOG_WARNING, "NEW_BC without chan_list?\n");
break;
}
if (bc) {
ch->bc = (struct misdn_bchannel *) user_data;
}
break;
case EVENT_DTMF_TONE:
{
/* sending INFOS as DTMF-Frames :) */
struct ast_frame fr;
memset(&fr, 0, sizeof(fr));
fr.frametype = AST_FRAME_DTMF;
fr.subclass.integer = bc->dtmf ;
fr.src = NULL;
fr.data.ptr = NULL;
fr.datalen = 0;
fr.samples = 0;
fr.mallocd = 0;
fr.offset = 0;
fr.delivery = ast_tv(0,0);
if (!ch->ignore_dtmf) {
chan_misdn_log(2, bc->port, " --> DTMF:%c\n", bc->dtmf);
ast_queue_frame(ch->ast, &fr);
} else {
chan_misdn_log(2, bc->port, " --> Ignoring DTMF:%c due to bridge flags\n", bc->dtmf);
}
break;
}
case EVENT_STATUS:
break;
case EVENT_INFORMATION:
if (ch->state != MISDN_CONNECTED) {
stop_indicate(ch);
}
if (!ch->ast) {
break;
}
if (ch->state == MISDN_WAITING4DIGS) {
RAII_VAR(struct ast_features_pickup_config *, pickup_cfg, NULL, ao2_cleanup);
const char *pickupexten;
/* Ok, incomplete Setup, waiting till extension exists */
if (ast_strlen_zero(bc->info_dad) && ! ast_strlen_zero(bc->keypad)) {
chan_misdn_log(1, bc->port, " --> using keypad as info\n");
ast_copy_string(bc->info_dad, bc->keypad, sizeof(bc->info_dad));
}
strncat(bc->dialed.number, bc->info_dad, sizeof(bc->dialed.number) - strlen(bc->dialed.number) - 1);
ast_channel_exten_set(ch->ast, bc->dialed.number);
ast_channel_lock(ch->ast);
pickup_cfg = ast_get_chan_features_pickup_config(ch->ast);
if (!pickup_cfg) {
ast_log(LOG_ERROR, "Unable to retrieve pickup configuration options. Unable to detect call pickup extension\n");
pickupexten = "";
} else {
pickupexten = ast_strdupa(pickup_cfg->pickupexten);
}
ast_channel_unlock(ch->ast);
/* Check for Pickup Request first */
if (!strcmp(ast_channel_exten(ch->ast), pickupexten)) {
if (ast_pickup_call(ch->ast)) {
hangup_chan(ch, bc);
} else {
ch->state = MISDN_CALLING_ACKNOWLEDGE;
hangup_chan(ch, bc);
ch->ast = NULL;
break;
}
}
if (!ast_canmatch_extension(ch->ast, ch->context, bc->dialed.number, 1, bc->caller.number)) {
if (ast_exists_extension(ch->ast, ch->context, "i", 1, bc->caller.number)) {
ast_log(LOG_WARNING,
"Extension '%s@%s' can never match. Jumping to 'i' extension. port:%d\n",
bc->dialed.number, ch->context, bc->port);
pbx_builtin_setvar_helper(ch->ast, "INVALID_EXTEN", bc->dialed.number);
ast_channel_exten_set(ch->ast, "i");
ch->state = MISDN_DIALING;
start_pbx(ch, bc, ch->ast);
break;
}
ast_log(LOG_WARNING,
"Extension '%s@%s' can never match. Disconnecting. port:%d\n"
"\tMaybe you want to add an 'i' extension to catch this case.\n",
bc->dialed.number, ch->context, bc->port);
if (bc->nt) {
hanguptone_indicate(ch);
}
ch->state = MISDN_EXTCANTMATCH;
bc->out_cause = AST_CAUSE_UNALLOCATED;
misdn_lib_send_event(bc, EVENT_DISCONNECT);
break;
}
if (ch->overlap_dial) {
ast_mutex_lock(&ch->overlap_tv_lock);
ch->overlap_tv = ast_tvnow();
ast_mutex_unlock(&ch->overlap_tv_lock);
if (ch->overlap_dial_task == -1) {
ch->overlap_dial_task =
misdn_tasks_add_variable(ch->overlap_dial, misdn_overlap_dial_task, ch);
}
break;
}
if (ast_exists_extension(ch->ast, ch->context, bc->dialed.number, 1, bc->caller.number)) {
ch->state = MISDN_DIALING;
start_pbx(ch, bc, ch->ast);
}
} else {
/* sending INFOS as DTMF-Frames :) */
struct ast_frame fr;
int digits;
memset(&fr, 0, sizeof(fr));
fr.frametype = AST_FRAME_DTMF;
fr.subclass.integer = bc->info_dad[0] ;
fr.src = NULL;
fr.data.ptr = NULL;
fr.datalen = 0;
fr.samples = 0;
fr.mallocd = 0;
fr.offset = 0;
fr.delivery = ast_tv(0,0);
misdn_cfg_get(0, MISDN_GEN_APPEND_DIGITS2EXTEN, &digits, sizeof(digits));
if (ch->state != MISDN_CONNECTED) {
if (digits) {
strncat(bc->dialed.number, bc->info_dad, sizeof(bc->dialed.number) - strlen(bc->dialed.number) - 1);
ast_channel_exten_set(ch->ast, bc->dialed.number);
}
ast_queue_frame(ch->ast, &fr);
}
}
break;
case EVENT_SETUP:
{
struct ast_channel *chan;
int exceed;
int ai;
int im;
Enhancements to connected line and redirecting work. From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
int append_msn = 0;
RAII_VAR(struct ast_features_pickup_config *, pickup_cfg, NULL, ao2_cleanup);
const char *pickupexten;
if (ch) {
switch (ch->state) {
case MISDN_NOTHING:
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
chan_list_unref(ch, "Ignore found ch. Is it for an outgoing call?");
ch = NULL;
break;
default:
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
chan_list_unref(ch, "Already have a call.");
chan_misdn_log(1, bc->port, " --> Ignoring Call we have already one\n");
return RESPONSE_IGNORE_SETUP_WITHOUT_CLOSE; /* Ignore MSNs which are not in our List */
}
}
if (!bc->nt && !misdn_is_msn_valid(bc->port, &bc->dialed)) {
chan_misdn_log(1, bc->port, " --> Ignoring Call, its not in our MSN List\n");
return RESPONSE_IGNORE_SETUP; /* Ignore MSNs which are not in our List */
}
if (bc->cw) {
int cause;
chan_misdn_log(0, bc->port, " --> Call Waiting on PMP sending RELEASE_COMPLETE\n");
misdn_cfg_get(bc->port, MISDN_CFG_REJECT_CAUSE, &cause, sizeof(cause));
bc->out_cause = cause ? cause : AST_CAUSE_NORMAL_CLEARING;
return RESPONSE_RELEASE_SETUP;
}
print_bearer(bc);
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
ch = chan_list_init(ORG_MISDN);
if (!ch) {
chan_misdn_log(-1, bc->port, "cb_events: malloc for chan_list failed!\n");
Merged revisions 374515-374535 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ................ r374515 | rmudgett | 2012-10-04 17:52:36 -0500 (Thu, 04 Oct 2012) | 10 lines chan_misdn: Remove some deadcode * Made setup_bc() static. Patches: patch1_unused-code.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882 ................ r374516 | rmudgett | 2012-10-04 18:01:01 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused bchan states Patches: patch2_unused-states.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374517 | rmudgett | 2012-10-04 18:17:51 -0500 (Thu, 04 Oct 2012) | 16 lines chan_misdn: Remove unnecessary null pointer checks and checks for stack->nt * cleanup_bc() is always called with valid bc (or it would've crashed before). * Value of stack->nt is known in advance at some places. * Rename handle_event() to handle_event_te(), handle_frm() to handle_frm_te(). Patches: patch3_checks.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882 ................ r374518 | rmudgett | 2012-10-04 18:21:59 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Fix spelling in log messages Patches: patch4_spelling.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374519 | rmudgett | 2012-10-04 18:31:59 -0500 (Thu, 04 Oct 2012) | 15 lines chan_misdn: Don't cleanup a bc twice. In handle_frm_te() after calling misdn_lib_send_event(bc, EVENT_RELEASE_COMPLETE) bc is emptied, cleaned and set not in use, although misdn_lib_send_event() already did the same. This is bad. When it's not in use we are not allowed to touch it. * Moved log message in front of the resulting actions and fixed it to match the case. Patches: patch5_bccleanup.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374520 | rmudgett | 2012-10-04 18:43:56 -0500 (Thu, 04 Oct 2012) | 12 lines chan_misdn: Fix memory leaks, bc, chan not cleaned up etc., really bad stuff. * Fix return codes of cb_events() for EVENT_SETUP to use caller's cleanup mechanisms. * Move cl_queue_chan() call after bearer check. Patches: patch6_leaks.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374521 | rmudgett | 2012-10-04 18:48:38 -0500 (Thu, 04 Oct 2012) | 11 lines chan_misdn: We must initialize cause on sending a DISCONNECT. We must initialize cause on sending a DISCONNECT, so it is later correctly indicated to ast_channel in case the answer (RELEASE/RELEASE_COMPLETE) does not include one. Patches: patch7_hangupcause.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374522 | rmudgett | 2012-10-04 19:03:56 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused code for upqueue Patches: patch8_unused-upqueue.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374523 | rmudgett | 2012-10-04 19:11:50 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Improve debugging (port number, messages fixed, dups removed) Patches: patch9_debug.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374533 | rmudgett | 2012-10-05 12:17:18 -0500 (Fri, 05 Oct 2012) | 8 lines chan_misdn: Better debug: we can print_bc_info even if there's no ast leg. Patches: patch10_debug-bc-2.diff (license #6372) patch uploaded by Guenther Kelleter Modified. JIRA ABE-2882 ................ r374534 | rmudgett | 2012-10-05 12:34:10 -0500 (Fri, 05 Oct 2012) | 16 lines chan_misdn: setup_bc() is called too early for an incoming SETUP on TE. This prevents the B channel from being setup for HDLC mode when requested by the bearer capability and config option hdlc=yes. It violates ETS300102 Ch.5.2.3.2: "The user, in any case, must not connect to the channel until a CONNECT ACKNOWLEDGE message has been received." * Call setup_bc() on receipt of CONNECT_ACKNOWLEGDE for PTMP, and on first response to SETUP for PTP. Patches: abe-2881-2.diff (license #6372) patch uploaded by Guenther Kelleter Modified. JIRA ABE-2881 ................ r374535 | rmudgett | 2012-10-05 12:41:05 -0500 (Fri, 05 Oct 2012) | 2 lines chan_misdn: Remove some more deadcode. ................ ........ Merged revisions 374536 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374537 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374538 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374539 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-05 18:42:14 +00:00
return RESPONSE_RELEASE_SETUP;
}
ch->bc = bc;
ch->l3id = bc->l3_id;
ch->addr = bc->addr;
{
struct ast_format_cap *cap = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK);
struct ast_format tmpfmt;
if (!(cap)) {
return RESPONSE_ERR;
}
ast_format_cap_add(cap, ast_format_set(&tmpfmt, AST_FORMAT_ALAW, 0));
chan = misdn_new(ch, AST_STATE_RESERVED, bc->dialed.number, bc->caller.number, cap, NULL, NULL, bc->port, bc->channel);
cap = ast_format_cap_destroy(cap);
}
if (!chan) {
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
chan_list_unref(ch, "Failed to create a new channel");
Merged revisions 374515-374535 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ................ r374515 | rmudgett | 2012-10-04 17:52:36 -0500 (Thu, 04 Oct 2012) | 10 lines chan_misdn: Remove some deadcode * Made setup_bc() static. Patches: patch1_unused-code.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882 ................ r374516 | rmudgett | 2012-10-04 18:01:01 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused bchan states Patches: patch2_unused-states.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374517 | rmudgett | 2012-10-04 18:17:51 -0500 (Thu, 04 Oct 2012) | 16 lines chan_misdn: Remove unnecessary null pointer checks and checks for stack->nt * cleanup_bc() is always called with valid bc (or it would've crashed before). * Value of stack->nt is known in advance at some places. * Rename handle_event() to handle_event_te(), handle_frm() to handle_frm_te(). Patches: patch3_checks.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882 ................ r374518 | rmudgett | 2012-10-04 18:21:59 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Fix spelling in log messages Patches: patch4_spelling.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374519 | rmudgett | 2012-10-04 18:31:59 -0500 (Thu, 04 Oct 2012) | 15 lines chan_misdn: Don't cleanup a bc twice. In handle_frm_te() after calling misdn_lib_send_event(bc, EVENT_RELEASE_COMPLETE) bc is emptied, cleaned and set not in use, although misdn_lib_send_event() already did the same. This is bad. When it's not in use we are not allowed to touch it. * Moved log message in front of the resulting actions and fixed it to match the case. Patches: patch5_bccleanup.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374520 | rmudgett | 2012-10-04 18:43:56 -0500 (Thu, 04 Oct 2012) | 12 lines chan_misdn: Fix memory leaks, bc, chan not cleaned up etc., really bad stuff. * Fix return codes of cb_events() for EVENT_SETUP to use caller's cleanup mechanisms. * Move cl_queue_chan() call after bearer check. Patches: patch6_leaks.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374521 | rmudgett | 2012-10-04 18:48:38 -0500 (Thu, 04 Oct 2012) | 11 lines chan_misdn: We must initialize cause on sending a DISCONNECT. We must initialize cause on sending a DISCONNECT, so it is later correctly indicated to ast_channel in case the answer (RELEASE/RELEASE_COMPLETE) does not include one. Patches: patch7_hangupcause.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374522 | rmudgett | 2012-10-04 19:03:56 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused code for upqueue Patches: patch8_unused-upqueue.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374523 | rmudgett | 2012-10-04 19:11:50 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Improve debugging (port number, messages fixed, dups removed) Patches: patch9_debug.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374533 | rmudgett | 2012-10-05 12:17:18 -0500 (Fri, 05 Oct 2012) | 8 lines chan_misdn: Better debug: we can print_bc_info even if there's no ast leg. Patches: patch10_debug-bc-2.diff (license #6372) patch uploaded by Guenther Kelleter Modified. JIRA ABE-2882 ................ r374534 | rmudgett | 2012-10-05 12:34:10 -0500 (Fri, 05 Oct 2012) | 16 lines chan_misdn: setup_bc() is called too early for an incoming SETUP on TE. This prevents the B channel from being setup for HDLC mode when requested by the bearer capability and config option hdlc=yes. It violates ETS300102 Ch.5.2.3.2: "The user, in any case, must not connect to the channel until a CONNECT ACKNOWLEDGE message has been received." * Call setup_bc() on receipt of CONNECT_ACKNOWLEGDE for PTMP, and on first response to SETUP for PTP. Patches: abe-2881-2.diff (license #6372) patch uploaded by Guenther Kelleter Modified. JIRA ABE-2881 ................ r374535 | rmudgett | 2012-10-05 12:41:05 -0500 (Fri, 05 Oct 2012) | 2 lines chan_misdn: Remove some more deadcode. ................ ........ Merged revisions 374536 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374537 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374538 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374539 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-05 18:42:14 +00:00
ast_log(LOG_ERROR, "cb_events: misdn_new failed!\n");
return RESPONSE_RELEASE_SETUP;
}
ast_channel_lock(chan);
pickup_cfg = ast_get_chan_features_pickup_config(chan);
if (!pickup_cfg) {
ast_log(LOG_ERROR, "Unable to retrieve pickup configuration options. Unable to detect call pickup extension\n");
pickupexten = "";
} else {
pickupexten = ast_strdupa(pickup_cfg->pickupexten);
}
ast_channel_unlock(chan);
if ((exceed = add_in_calls(bc->port))) {
char tmp[16];
snprintf(tmp, sizeof(tmp), "%d", exceed);
pbx_builtin_setvar_helper(chan, "MAX_OVERFLOW", tmp);
}
read_config(ch);
export_ch(chan, bc, ch);
ast_channel_lock(ch->ast);
ast_channel_rings_set(ch->ast, 1);
ast_setstate(ch->ast, AST_STATE_RINGING);
ast_channel_unlock(ch->ast);
/* Update asterisk channel caller information */
chan_misdn_log(2, bc->port, " --> TON: %s(%d)\n", misdn_to_str_ton(bc->caller.number_type), bc->caller.number_type);
chan_misdn_log(2, bc->port, " --> PLAN: %s(%d)\n", misdn_to_str_plan(bc->caller.number_plan), bc->caller.number_plan);
ast_channel_caller(chan)->id.number.plan = misdn_to_ast_ton(bc->caller.number_type)
| misdn_to_ast_plan(bc->caller.number_plan);
chan_misdn_log(2, bc->port, " --> PRES: %s(%d)\n", misdn_to_str_pres(bc->caller.presentation), bc->caller.presentation);
chan_misdn_log(2, bc->port, " --> SCREEN: %s(%d)\n", misdn_to_str_screen(bc->caller.screening), bc->caller.screening);
ast_channel_caller(chan)->id.number.presentation = misdn_to_ast_pres(bc->caller.presentation)
| misdn_to_ast_screen(bc->caller.screening);
ast_set_callerid(chan, bc->caller.number, NULL, bc->caller.number);
Enhancements to connected line and redirecting work. From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
misdn_cfg_get(bc->port, MISDN_CFG_APPEND_MSN_TO_CALLERID_TAG, &append_msn, sizeof(append_msn));
if (append_msn) {
strncat(bc->incoming_cid_tag, "_", sizeof(bc->incoming_cid_tag) - strlen(bc->incoming_cid_tag) - 1);
strncat(bc->incoming_cid_tag, bc->dialed.number, sizeof(bc->incoming_cid_tag) - strlen(bc->incoming_cid_tag) - 1);
}
ast_channel_lock(chan);
ast_channel_caller(chan)->id.tag = ast_strdup(bc->incoming_cid_tag);
Enhancements to connected line and redirecting work. From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
ast_channel_unlock(chan);
if (!ast_strlen_zero(bc->redirecting.from.number)) {
/* Add configured prefix to redirecting.from.number */
misdn_add_number_prefix(bc->port, bc->redirecting.from.number_type, bc->redirecting.from.number, sizeof(bc->redirecting.from.number));
/* Update asterisk channel redirecting information */
Enhancements to connected line and redirecting work. From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
misdn_copy_redirecting_to_ast(chan, &bc->redirecting, bc->incoming_cid_tag);
}
pbx_builtin_setvar_helper(chan, "TRANSFERCAPABILITY", ast_transfercapability2str(bc->capability));
ast_channel_transfercapability_set(chan, bc->capability);
switch (bc->capability) {
case INFO_CAPABILITY_DIGITAL_UNRESTRICTED:
pbx_builtin_setvar_helper(chan, "CALLTYPE", "DIGITAL");
break;
default:
pbx_builtin_setvar_helper(chan, "CALLTYPE", "SPEECH");
break;
}
if (!strstr(ch->allowed_bearers, "all")) {
int i;
for (i = 0; i < ARRAY_LEN(allowed_bearers_array); ++i) {
if (allowed_bearers_array[i].cap == bc->capability) {
if (strstr(ch->allowed_bearers, allowed_bearers_array[i].name)) {
/* The bearer capability is allowed */
if (allowed_bearers_array[i].deprecated) {
chan_misdn_log(0, bc->port, "%s in allowed_bearers list is deprecated\n",
allowed_bearers_array[i].name);
}
break;
}
}
}
if (i == ARRAY_LEN(allowed_bearers_array)) {
/* We did not find the bearer capability */
chan_misdn_log(0, bc->port, "Bearer capability not allowed: %s(%d)\n",
bearer2str(bc->capability), bc->capability);
ch->state = MISDN_EXTCANTMATCH;
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
chan_list_unref(ch, "BC not allowed, releasing call");
Merged revisions 374515-374535 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ................ r374515 | rmudgett | 2012-10-04 17:52:36 -0500 (Thu, 04 Oct 2012) | 10 lines chan_misdn: Remove some deadcode * Made setup_bc() static. Patches: patch1_unused-code.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882 ................ r374516 | rmudgett | 2012-10-04 18:01:01 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused bchan states Patches: patch2_unused-states.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374517 | rmudgett | 2012-10-04 18:17:51 -0500 (Thu, 04 Oct 2012) | 16 lines chan_misdn: Remove unnecessary null pointer checks and checks for stack->nt * cleanup_bc() is always called with valid bc (or it would've crashed before). * Value of stack->nt is known in advance at some places. * Rename handle_event() to handle_event_te(), handle_frm() to handle_frm_te(). Patches: patch3_checks.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882 ................ r374518 | rmudgett | 2012-10-04 18:21:59 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Fix spelling in log messages Patches: patch4_spelling.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374519 | rmudgett | 2012-10-04 18:31:59 -0500 (Thu, 04 Oct 2012) | 15 lines chan_misdn: Don't cleanup a bc twice. In handle_frm_te() after calling misdn_lib_send_event(bc, EVENT_RELEASE_COMPLETE) bc is emptied, cleaned and set not in use, although misdn_lib_send_event() already did the same. This is bad. When it's not in use we are not allowed to touch it. * Moved log message in front of the resulting actions and fixed it to match the case. Patches: patch5_bccleanup.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374520 | rmudgett | 2012-10-04 18:43:56 -0500 (Thu, 04 Oct 2012) | 12 lines chan_misdn: Fix memory leaks, bc, chan not cleaned up etc., really bad stuff. * Fix return codes of cb_events() for EVENT_SETUP to use caller's cleanup mechanisms. * Move cl_queue_chan() call after bearer check. Patches: patch6_leaks.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374521 | rmudgett | 2012-10-04 18:48:38 -0500 (Thu, 04 Oct 2012) | 11 lines chan_misdn: We must initialize cause on sending a DISCONNECT. We must initialize cause on sending a DISCONNECT, so it is later correctly indicated to ast_channel in case the answer (RELEASE/RELEASE_COMPLETE) does not include one. Patches: patch7_hangupcause.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374522 | rmudgett | 2012-10-04 19:03:56 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused code for upqueue Patches: patch8_unused-upqueue.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374523 | rmudgett | 2012-10-04 19:11:50 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Improve debugging (port number, messages fixed, dups removed) Patches: patch9_debug.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374533 | rmudgett | 2012-10-05 12:17:18 -0500 (Fri, 05 Oct 2012) | 8 lines chan_misdn: Better debug: we can print_bc_info even if there's no ast leg. Patches: patch10_debug-bc-2.diff (license #6372) patch uploaded by Guenther Kelleter Modified. JIRA ABE-2882 ................ r374534 | rmudgett | 2012-10-05 12:34:10 -0500 (Fri, 05 Oct 2012) | 16 lines chan_misdn: setup_bc() is called too early for an incoming SETUP on TE. This prevents the B channel from being setup for HDLC mode when requested by the bearer capability and config option hdlc=yes. It violates ETS300102 Ch.5.2.3.2: "The user, in any case, must not connect to the channel until a CONNECT ACKNOWLEDGE message has been received." * Call setup_bc() on receipt of CONNECT_ACKNOWLEGDE for PTMP, and on first response to SETUP for PTP. Patches: abe-2881-2.diff (license #6372) patch uploaded by Guenther Kelleter Modified. JIRA ABE-2881 ................ r374535 | rmudgett | 2012-10-05 12:41:05 -0500 (Fri, 05 Oct 2012) | 2 lines chan_misdn: Remove some more deadcode. ................ ........ Merged revisions 374536 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374537 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374538 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374539 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-05 18:42:14 +00:00
bc->out_cause = AST_CAUSE_INCOMPATIBLE_DESTINATION;
return RESPONSE_RELEASE_SETUP;
}
}
Merged revisions 374515-374535 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ................ r374515 | rmudgett | 2012-10-04 17:52:36 -0500 (Thu, 04 Oct 2012) | 10 lines chan_misdn: Remove some deadcode * Made setup_bc() static. Patches: patch1_unused-code.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882 ................ r374516 | rmudgett | 2012-10-04 18:01:01 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused bchan states Patches: patch2_unused-states.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374517 | rmudgett | 2012-10-04 18:17:51 -0500 (Thu, 04 Oct 2012) | 16 lines chan_misdn: Remove unnecessary null pointer checks and checks for stack->nt * cleanup_bc() is always called with valid bc (or it would've crashed before). * Value of stack->nt is known in advance at some places. * Rename handle_event() to handle_event_te(), handle_frm() to handle_frm_te(). Patches: patch3_checks.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882 ................ r374518 | rmudgett | 2012-10-04 18:21:59 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Fix spelling in log messages Patches: patch4_spelling.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374519 | rmudgett | 2012-10-04 18:31:59 -0500 (Thu, 04 Oct 2012) | 15 lines chan_misdn: Don't cleanup a bc twice. In handle_frm_te() after calling misdn_lib_send_event(bc, EVENT_RELEASE_COMPLETE) bc is emptied, cleaned and set not in use, although misdn_lib_send_event() already did the same. This is bad. When it's not in use we are not allowed to touch it. * Moved log message in front of the resulting actions and fixed it to match the case. Patches: patch5_bccleanup.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374520 | rmudgett | 2012-10-04 18:43:56 -0500 (Thu, 04 Oct 2012) | 12 lines chan_misdn: Fix memory leaks, bc, chan not cleaned up etc., really bad stuff. * Fix return codes of cb_events() for EVENT_SETUP to use caller's cleanup mechanisms. * Move cl_queue_chan() call after bearer check. Patches: patch6_leaks.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374521 | rmudgett | 2012-10-04 18:48:38 -0500 (Thu, 04 Oct 2012) | 11 lines chan_misdn: We must initialize cause on sending a DISCONNECT. We must initialize cause on sending a DISCONNECT, so it is later correctly indicated to ast_channel in case the answer (RELEASE/RELEASE_COMPLETE) does not include one. Patches: patch7_hangupcause.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374522 | rmudgett | 2012-10-04 19:03:56 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused code for upqueue Patches: patch8_unused-upqueue.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374523 | rmudgett | 2012-10-04 19:11:50 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Improve debugging (port number, messages fixed, dups removed) Patches: patch9_debug.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374533 | rmudgett | 2012-10-05 12:17:18 -0500 (Fri, 05 Oct 2012) | 8 lines chan_misdn: Better debug: we can print_bc_info even if there's no ast leg. Patches: patch10_debug-bc-2.diff (license #6372) patch uploaded by Guenther Kelleter Modified. JIRA ABE-2882 ................ r374534 | rmudgett | 2012-10-05 12:34:10 -0500 (Fri, 05 Oct 2012) | 16 lines chan_misdn: setup_bc() is called too early for an incoming SETUP on TE. This prevents the B channel from being setup for HDLC mode when requested by the bearer capability and config option hdlc=yes. It violates ETS300102 Ch.5.2.3.2: "The user, in any case, must not connect to the channel until a CONNECT ACKNOWLEDGE message has been received." * Call setup_bc() on receipt of CONNECT_ACKNOWLEGDE for PTMP, and on first response to SETUP for PTP. Patches: abe-2881-2.diff (license #6372) patch uploaded by Guenther Kelleter Modified. JIRA ABE-2881 ................ r374535 | rmudgett | 2012-10-05 12:41:05 -0500 (Fri, 05 Oct 2012) | 2 lines chan_misdn: Remove some more deadcode. ................ ........ Merged revisions 374536 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374537 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374538 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374539 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-05 18:42:14 +00:00
/** queue new chan **/
cl_queue_chan(ch);
if (bc->fac_in.Function != Fac_None) {
misdn_facility_ie_handler(event, bc, ch);
}
/* Check for Pickup Request first */
if (!strcmp(ast_channel_exten(chan), pickupexten)) {
if (!ch->noautorespond_on_setup) {
/* Sending SETUP_ACK */
misdn_lib_send_event(bc, EVENT_SETUP_ACKNOWLEDGE);
} else {
ch->state = MISDN_INCOMING_SETUP;
}
if (ast_pickup_call(chan)) {
hangup_chan(ch, bc);
} else {
ch->state = MISDN_CALLING_ACKNOWLEDGE;
hangup_chan(ch, bc);
ch->ast = NULL;
break;
}
}
/*
* added support for s extension hope it will help those poor cretains
* which haven't overlap dial.
*/
misdn_cfg_get(bc->port, MISDN_CFG_ALWAYS_IMMEDIATE, &ai, sizeof(ai));
if (ai) {
do_immediate_setup(bc, ch, chan);
break;
}
/* check if we should jump into s when we have no dialed.number */
misdn_cfg_get(bc->port, MISDN_CFG_IMMEDIATE, &im, sizeof(im));
if (im && ast_strlen_zero(bc->dialed.number)) {
do_immediate_setup(bc, ch, chan);
break;
}
chan_misdn_log(5, bc->port, "CONTEXT:%s\n", ch->context);
if (!ast_canmatch_extension(ch->ast, ch->context, bc->dialed.number, 1, bc->caller.number)) {
if (ast_exists_extension(ch->ast, ch->context, "i", 1, bc->caller.number)) {
ast_log(LOG_WARNING,
"Extension '%s@%s' can never match. Jumping to 'i' extension. port:%d\n",
bc->dialed.number, ch->context, bc->port);
pbx_builtin_setvar_helper(ch->ast, "INVALID_EXTEN", bc->dialed.number);
ast_channel_exten_set(ch->ast, "i");
misdn_lib_send_event(bc, EVENT_SETUP_ACKNOWLEDGE);
ch->state = MISDN_DIALING;
start_pbx(ch, bc, chan);
break;
}
ast_log(LOG_WARNING,
"Extension '%s@%s' can never match. Disconnecting. port:%d\n"
"\tMaybe you want to add an 'i' extension to catch this case.\n",
bc->dialed.number, ch->context, bc->port);
if (bc->nt) {
hanguptone_indicate(ch);
}
ch->state = MISDN_EXTCANTMATCH;
bc->out_cause = AST_CAUSE_UNALLOCATED;
misdn_lib_send_event(bc, bc->nt ? EVENT_RELEASE_COMPLETE : EVENT_RELEASE);
break;
}
/* Whatever happens, when sending_complete is set or we are PTMP TE, we will definitely
* jump into the dialplan, when the dialed extension does not exist, the 's' extension
* will be used by Asterisk automatically. */
if (bc->sending_complete || (!bc->nt && !misdn_lib_is_ptp(bc->port))) {
if (!ch->noautorespond_on_setup) {
ch->state=MISDN_DIALING;
misdn_lib_send_event(bc, EVENT_PROCEEDING);
} else {
ch->state = MISDN_INCOMING_SETUP;
}
start_pbx(ch, bc, chan);
break;
}
/*
* When we are NT and overlapdial is set and if
* the number is empty, we wait for the ISDN timeout
* instead of our own timer.
*/
if (ch->overlap_dial && bc->nt && !bc->dialed.number[0]) {
wait_for_digits(ch, bc, chan);
break;
}
/*
* If overlapdial we will definitely send a SETUP_ACKNOWLEDGE and wait for more
* Infos with a Interdigit Timeout.
* */
if (ch->overlap_dial) {
ast_mutex_lock(&ch->overlap_tv_lock);
ch->overlap_tv = ast_tvnow();
ast_mutex_unlock(&ch->overlap_tv_lock);
wait_for_digits(ch, bc, chan);
if (ch->overlap_dial_task == -1) {
ch->overlap_dial_task =
misdn_tasks_add_variable(ch->overlap_dial, misdn_overlap_dial_task, ch);
}
break;
}
/* If the extension does not exist and we're not TE_PTMP we wait for more digits
* without interdigit timeout.
* */
if (!ast_exists_extension(ch->ast, ch->context, bc->dialed.number, 1, bc->caller.number)) {
wait_for_digits(ch, bc, chan);
break;
}
/*
* If the extension exists let's just jump into it.
* */
if (ast_exists_extension(ch->ast, ch->context, bc->dialed.number, 1, bc->caller.number)) {
misdn_lib_send_event(bc, bc->need_more_infos ? EVENT_SETUP_ACKNOWLEDGE : EVENT_PROCEEDING);
ch->state = MISDN_DIALING;
start_pbx(ch, bc, chan);
break;
}
break;
}
#if defined(AST_MISDN_ENHANCEMENTS)
case EVENT_REGISTER:
if (bc->fac_in.Function != Fac_None) {
misdn_facility_ie_handler(event, bc, ch);
}
/*
* Shut down this connection immediately.
* The current design of chan_misdn data structures
* does not allow the proper handling of inbound call records
* without an assigned B channel. Therefore, we cannot
* be the CCBS User-B party in a point-to-point setup.
*/
bc->fac_out.Function = Fac_None;
bc->out_cause = AST_CAUSE_NORMAL_CLEARING;
misdn_lib_send_event(bc, EVENT_RELEASE_COMPLETE);
break;
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
case EVENT_SETUP_ACKNOWLEDGE:
ch->state = MISDN_CALLING_ACKNOWLEDGE;
if (bc->channel) {
update_name(ch->ast,bc->port,bc->channel);
}
if (bc->fac_in.Function != Fac_None) {
misdn_facility_ie_handler(event, bc, ch);
}
if (!ast_strlen_zero(bc->infos_pending)) {
/* TX Pending Infos */
strncat(bc->dialed.number, bc->infos_pending, sizeof(bc->dialed.number) - strlen(bc->dialed.number) - 1);
if (!ch->ast) {
break;
}
ast_channel_exten_set(ch->ast, bc->dialed.number);
ast_copy_string(bc->info_dad, bc->infos_pending, sizeof(bc->info_dad));
ast_copy_string(bc->infos_pending, "", sizeof(bc->infos_pending));
misdn_lib_send_event(bc, EVENT_INFORMATION);
}
break;
case EVENT_PROCEEDING:
if (misdn_cap_is_speech(bc->capability) &&
misdn_inband_avail(bc)) {
start_bc_tones(ch);
}
ch->state = MISDN_PROCEEDING;
if (bc->fac_in.Function != Fac_None) {
misdn_facility_ie_handler(event, bc, ch);
}
if (!ch->ast) {
break;
}
ast_queue_control(ch->ast, AST_CONTROL_PROCEEDING);
break;
case EVENT_PROGRESS:
if (bc->channel) {
update_name(ch->ast, bc->port, bc->channel);
}
if (bc->fac_in.Function != Fac_None) {
misdn_facility_ie_handler(event, bc, ch);
}
if (!bc->nt) {
if (misdn_cap_is_speech(bc->capability) &&
misdn_inband_avail(bc)) {
start_bc_tones(ch);
}
ch->state = MISDN_PROGRESS;
if (!ch->ast) {
break;
}
ast_queue_control(ch->ast, AST_CONTROL_PROGRESS);
}
break;
case EVENT_ALERTING:
ch->state = MISDN_ALERTING;
if (!ch->ast) {
break;
}
if (bc->fac_in.Function != Fac_None) {
misdn_facility_ie_handler(event, bc, ch);
}
ast_queue_control(ch->ast, AST_CONTROL_RINGING);
ast_channel_lock(ch->ast);
ast_setstate(ch->ast, AST_STATE_RINGING);
ast_channel_unlock(ch->ast);
cb_log(7, bc->port, " --> Set State Ringing\n");
if (misdn_cap_is_speech(bc->capability) && misdn_inband_avail(bc)) {
cb_log(1, bc->port, "Starting Tones, we have inband Data\n");
start_bc_tones(ch);
} else {
cb_log(3, bc->port, " --> We have no inband Data, the other end must create ringing\n");
if (ch->far_alerting) {
cb_log(1, bc->port, " --> The other end can not do ringing eh ?.. we must do all ourself..");
start_bc_tones(ch);
/*tone_indicate(ch, TONE_FAR_ALERTING);*/
}
}
break;
case EVENT_CONNECT:
if (bc->fac_in.Function != Fac_None) {
misdn_facility_ie_handler(event, bc, ch);
}
#if defined(AST_MISDN_ENHANCEMENTS)
if (bc->div_leg_3_rx_wanted) {
bc->div_leg_3_rx_wanted = 0;
if (ch->ast) {
Add private representation of caller, connected and redirecting party ids. This patch adds the feature "Private representation of caller, connected and redirecting party ids", as previously discussed with us (DATUS) and Digium. 1. Feature motivation Until now it is quite difficult to modify a party number or name which can only be seen by exactly one particular instantiated technology channel subscriber. One example where a modified party number or name on one channel is spread over several channels are supplementary services like call transfer or pickup. To implement these features Asterisk internally copies caller and connected ids from one channel to another. Another example are extension subscriptions. The monitoring entities (watchers) are notified of state changes and - if desired - of party numbers or names which represent the involving call parties. One major feature where a private representation of party names is essentially needed, i.e. where a party name shall be exclusively signaled to only one particular user, is a private user-specific name resolution for party numbers. A lookup in a private destination-dependent telephone book shall provide party names which cannot be seen by any other user at any time. 2. Feature Description This feature comes along with the implementation of additional private party id elements for caller id, connected id and redirecting ids inside Asterisk channels. The private party id elements can be read or set by the user using Asterisk dialplan functions. When a technology channel is initiating a call, receives an internal connected-line update event, or receives an internal redirecting update event, it merges the corresponding public id with the private id to create an effective party id. The effective party id is then used for protocol signaling. The channel technologies which initially support the private id representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and PRI (chan_dahdi). Once a private name or number on a channel is set and (implicitly) made valid, it is generally used for any further protocol signaling until it is rewritten or invalidated. To simplify the invalidation of private ids all internally generated connected/redirecting update events and also all connected/redirecting update events which are generated by technology channels -- receiving regarding protocol information - automatically trigger the invalidation of private ids. If not using the private party id representation feature at all, i.e. if using only the 'regular' caller-id, connected and redirecting related functions, the current characteristic of Asterisk is not affected by the new extended functionality. 3. User interface Description To grant access to the private name and number representation from the Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan functions are extended by the following data types. The formats of these data types are equal to the corresponding regular 'non-private' already existing data types: CALLERID: priv-all priv-name priv-name-valid priv-name-charset priv-name-pres priv-num priv-num-valid priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE: priv-name priv-name-valid priv-name-pres priv-name-charset priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag REDIRECTING: priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd priv-orig-tag priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd priv-to-tag Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2030/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-10 19:54:55 +00:00
struct ast_party_redirecting redirecting;
ast_channel_redirecting(ch->ast)->to.number.presentation =
AST_PRES_RESTRICTED | AST_PRES_USER_NUMBER_UNSCREENED;
Add private representation of caller, connected and redirecting party ids. This patch adds the feature "Private representation of caller, connected and redirecting party ids", as previously discussed with us (DATUS) and Digium. 1. Feature motivation Until now it is quite difficult to modify a party number or name which can only be seen by exactly one particular instantiated technology channel subscriber. One example where a modified party number or name on one channel is spread over several channels are supplementary services like call transfer or pickup. To implement these features Asterisk internally copies caller and connected ids from one channel to another. Another example are extension subscriptions. The monitoring entities (watchers) are notified of state changes and - if desired - of party numbers or names which represent the involving call parties. One major feature where a private representation of party names is essentially needed, i.e. where a party name shall be exclusively signaled to only one particular user, is a private user-specific name resolution for party numbers. A lookup in a private destination-dependent telephone book shall provide party names which cannot be seen by any other user at any time. 2. Feature Description This feature comes along with the implementation of additional private party id elements for caller id, connected id and redirecting ids inside Asterisk channels. The private party id elements can be read or set by the user using Asterisk dialplan functions. When a technology channel is initiating a call, receives an internal connected-line update event, or receives an internal redirecting update event, it merges the corresponding public id with the private id to create an effective party id. The effective party id is then used for protocol signaling. The channel technologies which initially support the private id representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and PRI (chan_dahdi). Once a private name or number on a channel is set and (implicitly) made valid, it is generally used for any further protocol signaling until it is rewritten or invalidated. To simplify the invalidation of private ids all internally generated connected/redirecting update events and also all connected/redirecting update events which are generated by technology channels -- receiving regarding protocol information - automatically trigger the invalidation of private ids. If not using the private party id representation feature at all, i.e. if using only the 'regular' caller-id, connected and redirecting related functions, the current characteristic of Asterisk is not affected by the new extended functionality. 3. User interface Description To grant access to the private name and number representation from the Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan functions are extended by the following data types. The formats of these data types are equal to the corresponding regular 'non-private' already existing data types: CALLERID: priv-all priv-name priv-name-valid priv-name-charset priv-name-pres priv-num priv-num-valid priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE: priv-name priv-name-valid priv-name-pres priv-name-charset priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag REDIRECTING: priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd priv-orig-tag priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd priv-to-tag Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2030/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-10 19:54:55 +00:00
ast_party_redirecting_init(&redirecting);
ast_party_redirecting_copy(&redirecting, ast_channel_redirecting(ch->ast));
/*
* Reset any earlier private redirecting id representations and
* make sure that it is invalidated at the remote end.
*/
ast_party_id_reset(&redirecting.priv_orig);
ast_party_id_reset(&redirecting.priv_from);
ast_party_id_reset(&redirecting.priv_to);
ast_channel_queue_redirecting_update(ch->ast, &redirecting, NULL);
ast_party_redirecting_free(&redirecting);
}
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
/* we answer when we've got our very new L3 ID from the NT stack */
misdn_lib_send_event(bc, EVENT_CONNECT_ACKNOWLEDGE);
if (!ch->ast) {
break;
}
stop_indicate(ch);
#if defined(AST_MISDN_ENHANCEMENTS)
if (ch->record_id != -1) {
/*
* We will delete the associated call completion
* record since we now have a completed call.
* We will not wait/depend on the network to tell
* us to delete it.
*/
AST_LIST_LOCK(&misdn_cc_records_db);
cc_record = misdn_cc_find_by_id(ch->record_id);
if (cc_record) {
if (cc_record->ptp && cc_record->mode.ptp.bc) {
/* Close the call-completion signaling link */
cc_record->mode.ptp.bc->fac_out.Function = Fac_None;
cc_record->mode.ptp.bc->out_cause = AST_CAUSE_NORMAL_CLEARING;
misdn_lib_send_event(cc_record->mode.ptp.bc, EVENT_RELEASE_COMPLETE);
}
misdn_cc_delete(cc_record);
}
AST_LIST_UNLOCK(&misdn_cc_records_db);
ch->record_id = -1;
if (ch->peer) {
misdn_cc_set_peer_var(ch->peer, MISDN_CC_RECORD_ID, "");
ao2_ref(ch->peer, -1);
ch->peer = NULL;
}
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
Enhancements to connected line and redirecting work. From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
if (!ast_strlen_zero(bc->connected.number)) {
/* Add configured prefix to connected.number */
misdn_add_number_prefix(bc->port, bc->connected.number_type, bc->connected.number, sizeof(bc->connected.number));
Enhancements to connected line and redirecting work. From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
/* Update the connected line information on the other channel */
misdn_update_remote_party(ch->ast, &bc->connected, AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER, bc->incoming_cid_tag);
Enhancements to connected line and redirecting work. From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
}
ch->l3id = bc->l3_id;
ch->addr = bc->addr;
start_bc_tones(ch);
ch->state = MISDN_CONNECTED;
ast_queue_control(ch->ast, AST_CONTROL_ANSWER);
break;
case EVENT_CONNECT_ACKNOWLEDGE:
ch->l3id = bc->l3_id;
ch->addr = bc->addr;
start_bc_tones(ch);
ch->state = MISDN_CONNECTED;
break;
case EVENT_DISCONNECT:
/* we might not have an ch->ast ptr here anymore */
if (ch) {
if (bc->fac_in.Function != Fac_None) {
misdn_facility_ie_handler(event, bc, ch);
}
chan_misdn_log(3, bc->port, " --> org:%d nt:%d, inbandavail:%d state:%d\n", ch->originator, bc->nt, misdn_inband_avail(bc), ch->state);
if (ch->originator == ORG_AST && !bc->nt && misdn_inband_avail(bc) && ch->state != MISDN_CONNECTED) {
/* If there's inband information available (e.g. a
recorded message saying what was wrong with the
dialled number, or perhaps even giving an
alternative number, then play it instead of
immediately releasing the call */
chan_misdn_log(1, bc->port, " --> Inband Info Avail, not sending RELEASE\n");
ch->state = MISDN_DISCONNECTED;
start_bc_tones(ch);
if (ch->ast) {
ast_channel_hangupcause_set(ch->ast, bc->cause);
if (bc->cause == AST_CAUSE_USER_BUSY) {
ast_queue_control(ch->ast, AST_CONTROL_BUSY);
}
}
ch->need_busy = 0;
break;
}
bc->need_disconnect = 0;
stop_bc_tones(ch);
/* Check for held channel, to implement transfer */
held_ch = find_hold_call(bc);
if (!held_ch || !ch->ast || misdn_attempt_transfer(ch, held_ch)) {
hangup_chan(ch, bc);
}
} else {
held_ch = find_hold_call_l3(bc->l3_id);
if (held_ch) {
if (bc->fac_in.Function != Fac_None) {
misdn_facility_ie_handler(event, bc, held_ch);
}
if (held_ch->hold.state == MISDN_HOLD_ACTIVE) {
bc->need_disconnect = 0;
#if defined(TRANSFER_ON_HELD_CALL_HANGUP)
/*
* Some phones disconnect the held call and the active call at the
* same time to do the transfer. Unfortunately, either call could
* be disconnected first.
*/
ch = find_hold_active_call(bc);
if (!ch || misdn_attempt_transfer(ch, held_ch)) {
held_ch->hold.state = MISDN_HOLD_DISCONNECT;
hangup_chan(held_ch, bc);
}
#else
hangup_chan(held_ch, bc);
#endif /* defined(TRANSFER_ON_HELD_CALL_HANGUP) */
}
}
}
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
if (held_ch) {
chan_list_unref(held_ch, "Done with held call");
}
bc->out_cause = -1;
if (bc->need_release) {
misdn_lib_send_event(bc, EVENT_RELEASE);
}
break;
case EVENT_RELEASE:
if (!ch) {
ch = find_hold_call_l3(bc->l3_id);
if (!ch) {
chan_misdn_log(1, bc->port,
" --> no Ch, so we've already released. (%s)\n",
manager_isdn_get_info(event));
return -1;
}
}
if (bc->fac_in.Function != Fac_None) {
misdn_facility_ie_handler(event, bc, ch);
}
bc->need_disconnect = 0;
bc->need_release = 0;
hangup_chan(ch, bc);
release_chan(ch, bc);
break;
case EVENT_RELEASE_COMPLETE:
if (!ch) {
ch = find_hold_call_l3(bc->l3_id);
}
bc->need_disconnect = 0;
bc->need_release = 0;
bc->need_release_complete = 0;
if (ch) {
if (bc->fac_in.Function != Fac_None) {
misdn_facility_ie_handler(event, bc, ch);
}
stop_bc_tones(ch);
hangup_chan(ch, bc);
release_chan(ch, bc);
} else {
#if defined(AST_MISDN_ENHANCEMENTS)
/*
* A call-completion signaling link established with
* REGISTER does not have a struct chan_list record
* associated with it.
*/
AST_LIST_LOCK(&misdn_cc_records_db);
cc_record = misdn_cc_find_by_bc(bc);
if (cc_record) {
/* The call-completion signaling link is closed. */
misdn_cc_delete(cc_record);
}
AST_LIST_UNLOCK(&misdn_cc_records_db);
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
chan_misdn_log(1, bc->port,
" --> no Ch, so we've already released. (%s)\n",
manager_isdn_get_info(event));
}
break;
case EVENT_BCHAN_ERROR:
case EVENT_CLEANUP:
stop_bc_tones(ch);
switch (ch->state) {
case MISDN_CALLING:
bc->cause = AST_CAUSE_DESTINATION_OUT_OF_ORDER;
break;
default:
break;
}
hangup_chan(ch, bc);
release_chan(ch, bc);
break;
case EVENT_TONE_GENERATE:
{
int tone_len = bc->tone_cnt;
struct ast_channel *ast = ch->ast;
void *tmp;
int res;
int (*generate)(struct ast_channel *chan, void *tmp, int datalen, int samples);
chan_misdn_log(9, bc->port, "TONE_GEN: len:%d\n", tone_len);
if (!ast) {
break;
}
if (!ast_channel_generator(ast)) {
break;
}
tmp = ast_channel_generatordata(ast);
ast_channel_generatordata_set(ast, NULL);
generate = ast_channel_generator(ast)->generate;
if (tone_len < 0 || tone_len > 512) {
ast_log(LOG_NOTICE, "TONE_GEN: len was %d, set to 128\n", tone_len);
tone_len = 128;
}
res = generate(ast, tmp, tone_len, tone_len);
ast_channel_generatordata_set(ast, tmp);
if (res) {
ast_log(LOG_WARNING, "Auto-deactivating generator\n");
ast_deactivate_generator(ast);
} else {
bc->tone_cnt = 0;
}
break;
}
case EVENT_BCHAN_DATA:
if (ch->bc->AOCD_need_export) {
export_aoc_vars(ch->originator, ch->ast, ch->bc);
}
if (!misdn_cap_is_speech(ch->bc->capability)) {
struct ast_frame frame;
/* In Data Modes we queue frames */
memset(&frame, 0, sizeof(frame));
frame.frametype = AST_FRAME_VOICE; /* we have no data frames yet */
ast_format_set(&frame.subclass.format, AST_FORMAT_ALAW, 0);
frame.datalen = bc->bframe_len;
frame.samples = bc->bframe_len;
frame.mallocd = 0;
frame.offset = 0;
frame.delivery = ast_tv(0, 0);
frame.src = NULL;
frame.data.ptr = bc->bframe;
if (ch->ast) {
ast_queue_frame(ch->ast, &frame);
}
} else {
struct pollfd pfd = { .fd = ch->pipe[1], .events = POLLOUT };
int t;
t = ast_poll(&pfd, 1, 0);
if (t < 0) {
chan_misdn_log(-1, bc->port, "poll() error (err=%s)\n", strerror(errno));
break;
}
if (!t) {
chan_misdn_log(9, bc->port, "poll() timed out\n");
break;
}
if (pfd.revents & POLLOUT) {
chan_misdn_log(9, bc->port, "writing %d bytes to asterisk\n", bc->bframe_len);
if (write(ch->pipe[1], bc->bframe, bc->bframe_len) <= 0) {
chan_misdn_log(0, bc->port, "Write returned <=0 (err=%s) --> hanging up channel\n", strerror(errno));
Merged revisions 49313 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r49313 | crichter | 2007-01-03 10:06:50 +0100 (Mi, 03 Jan 2007) | 41 lines Merged revisions 48319,48321,48467,48552,48576,49135,49303 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) | 1 line changed a few debugs to higher debug levels ........ r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) | 1 line added the export and import of the MISDN_ADDRESS_COMPLETE Variable to inidcate wether the extension is already completely dialed or if there might come additional digits by information elements. also added some docs for that. ........ r48467 | crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line removed FIXUP state. added check for channel allocation conflict when we create a setup while the other site creates a setup on the same channel, besides the check we resolve this conflict. ........ r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18 Dez 2006) | 1 line when our PTP Partner sends us a SETUP with a preselected channel we just accept it, even when we're NT. added some checks for segfaults. ........ r48576 | crichter | 2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line when we reject a channel, because it's in use already, we shouldn't process the setup anymore. made the channel allocation a bit easier and more understandable, removed a few unused lines ........ r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02 Jan 2007) | 1 line added check for channel ranges in the set/empty channel functions. set pmp_l1_check default to no. added misdn restart pid cli command. added cleaning of channel when we send a RELEASE_COMPLETE. ........ r49303 | crichter | 2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines * Added check for bridging in misdn_call to avoid setting echocancellation when 2 mISDN channels are involved and when bridging is set. That lead to a kernel panic before under different situations, because we switched about 2 times between hardware bridging and echocancelation * readded MISDN_URATE variable which got lost before, this should make app_v110 work again * fixed typo ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49321 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-03 11:15:02 +00:00
stop_bc_tones(ch);
hangup_chan(ch, bc);
release_chan(ch, bc);
}
} else {
chan_misdn_log(1, bc->port, "Write Pipe full!\n");
}
}
break;
case EVENT_TIMEOUT:
if (ch && bc) {
chan_misdn_log(1, bc->port, "--> state: %s\n", misdn_get_ch_state(ch));
}
switch (ch->state) {
case MISDN_DIALING:
case MISDN_PROGRESS:
if (bc->nt && !ch->nttimeout) {
break;
}
/* fall-through */
case MISDN_CALLING:
case MISDN_ALERTING:
case MISDN_PROCEEDING:
case MISDN_CALLING_ACKNOWLEDGE:
if (bc->nt) {
bc->progress_indicator = INFO_PI_INBAND_AVAILABLE;
hanguptone_indicate(ch);
}
bc->out_cause = AST_CAUSE_UNALLOCATED;
misdn_lib_send_event(bc, EVENT_DISCONNECT);
break;
case MISDN_WAITING4DIGS:
if (bc->nt) {
bc->progress_indicator = INFO_PI_INBAND_AVAILABLE;
bc->out_cause = AST_CAUSE_UNALLOCATED;
hanguptone_indicate(ch);
misdn_lib_send_event(bc, EVENT_DISCONNECT);
} else {
bc->out_cause = AST_CAUSE_NORMAL_CLEARING;
misdn_lib_send_event(bc, EVENT_RELEASE);
}
break;
case MISDN_CLEANING:
chan_misdn_log(1, bc->port, " --> in state cleaning .. so ignoring, the stack should clean it for us\n");
break;
default:
misdn_lib_send_event(bc, EVENT_RELEASE_COMPLETE);
break;
}
break;
/****************************/
/** Supplementary Services **/
/****************************/
case EVENT_RETRIEVE:
if (!ch) {
chan_misdn_log(4, bc->port, " --> no CH, searching for held call\n");
ch = find_hold_call_l3(bc->l3_id);
if (!ch || ch->hold.state != MISDN_HOLD_ACTIVE) {
ast_log(LOG_WARNING, "No held call found, cannot Retrieve\n");
misdn_lib_send_event(bc, EVENT_RETRIEVE_REJECT);
break;
}
}
/* remember the channel again */
ch->bc = bc;
ch->hold.state = MISDN_HOLD_IDLE;
ch->hold.port = 0;
ch->hold.channel = 0;
ast_queue_unhold(ch->ast);
if (misdn_lib_send_event(bc, EVENT_RETRIEVE_ACKNOWLEDGE) < 0) {
chan_misdn_log(4, bc->port, " --> RETRIEVE_ACK failed\n");
misdn_lib_send_event(bc, EVENT_RETRIEVE_REJECT);
}
break;
case EVENT_HOLD:
{
int hold_allowed;
RAII_VAR(struct ast_channel *, bridged, NULL, ast_channel_cleanup);
misdn_cfg_get(bc->port, MISDN_CFG_HOLD_ALLOWED, &hold_allowed, sizeof(hold_allowed));
if (!hold_allowed) {
chan_misdn_log(-1, bc->port, "Hold not allowed this port.\n");
misdn_lib_send_event(bc, EVENT_HOLD_REJECT);
break;
}
bridged = ast_channel_bridge_peer(ch->ast);
if (bridged) {
chan_misdn_log(2, bc->port, "Bridge Partner is of type: %s\n", ast_channel_tech(bridged)->type);
ch->l3id = bc->l3_id;
/* forget the channel now */
ch->bc = NULL;
ch->hold.state = MISDN_HOLD_ACTIVE;
ch->hold.port = bc->port;
ch->hold.channel = bc->channel;
ast_queue_hold(ch->ast, NULL);
misdn_lib_send_event(bc, EVENT_HOLD_ACKNOWLEDGE);
} else {
misdn_lib_send_event(bc, EVENT_HOLD_REJECT);
chan_misdn_log(0, bc->port, "We aren't bridged to anybody\n");
}
break;
}
case EVENT_NOTIFY:
if (bc->redirecting.to_changed) {
/* Add configured prefix to redirecting.to.number */
misdn_add_number_prefix(bc->port, bc->redirecting.to.number_type,
bc->redirecting.to.number, sizeof(bc->redirecting.to.number));
}
switch (bc->notify_description_code) {
case mISDN_NOTIFY_CODE_DIVERSION_ACTIVATED:
/* Ignore for now. */
bc->redirecting.to_changed = 0;
break;
case mISDN_NOTIFY_CODE_CALL_IS_DIVERTING:
Add private representation of caller, connected and redirecting party ids. This patch adds the feature "Private representation of caller, connected and redirecting party ids", as previously discussed with us (DATUS) and Digium. 1. Feature motivation Until now it is quite difficult to modify a party number or name which can only be seen by exactly one particular instantiated technology channel subscriber. One example where a modified party number or name on one channel is spread over several channels are supplementary services like call transfer or pickup. To implement these features Asterisk internally copies caller and connected ids from one channel to another. Another example are extension subscriptions. The monitoring entities (watchers) are notified of state changes and - if desired - of party numbers or names which represent the involving call parties. One major feature where a private representation of party names is essentially needed, i.e. where a party name shall be exclusively signaled to only one particular user, is a private user-specific name resolution for party numbers. A lookup in a private destination-dependent telephone book shall provide party names which cannot be seen by any other user at any time. 2. Feature Description This feature comes along with the implementation of additional private party id elements for caller id, connected id and redirecting ids inside Asterisk channels. The private party id elements can be read or set by the user using Asterisk dialplan functions. When a technology channel is initiating a call, receives an internal connected-line update event, or receives an internal redirecting update event, it merges the corresponding public id with the private id to create an effective party id. The effective party id is then used for protocol signaling. The channel technologies which initially support the private id representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and PRI (chan_dahdi). Once a private name or number on a channel is set and (implicitly) made valid, it is generally used for any further protocol signaling until it is rewritten or invalidated. To simplify the invalidation of private ids all internally generated connected/redirecting update events and also all connected/redirecting update events which are generated by technology channels -- receiving regarding protocol information - automatically trigger the invalidation of private ids. If not using the private party id representation feature at all, i.e. if using only the 'regular' caller-id, connected and redirecting related functions, the current characteristic of Asterisk is not affected by the new extended functionality. 3. User interface Description To grant access to the private name and number representation from the Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan functions are extended by the following data types. The formats of these data types are equal to the corresponding regular 'non-private' already existing data types: CALLERID: priv-all priv-name priv-name-valid priv-name-charset priv-name-pres priv-num priv-num-valid priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE: priv-name priv-name-valid priv-name-pres priv-name-charset priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag REDIRECTING: priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd priv-orig-tag priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd priv-to-tag Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2030/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-10 19:54:55 +00:00
{
struct ast_party_redirecting redirecting;
Merged revisions 287017 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r287017 | rmudgett | 2010-09-15 15:53:38 -0500 (Wed, 15 Sep 2010) | 65 lines Merged revision 287014 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r287014 | rmudgett | 2010-09-15 15:32:24 -0500 (Wed, 15 Sep 2010) | 58 lines The handling of call transfer signaling for mISDN PTMP is not fully implemented. The handling of call transfer signaling for mISDN PTMP is not fully implemented. The signaling of number updates with ISDN/DSS1 ECT supplementary services (ETS 300 369-1) comes along with a notification indicator IE and redirection number IE for PTMP. The implementation in the current Asterisk mISDN channel unfortunately can handle these information elements only in a NOTIFY message. These information elements are also signaled in a FACILTY message with a RequestSubaddress facility, when the subscriber is already in the active state (see 9.2.4 and 9.2.5 of ETS 300 369-1). ********** abe_2526_ast.patch * Added support to handle the notification indicator IE and redirection number IE with the RequestSubaddress facility. * Made misdn_update_connected_line() send a NOTIFY message if Asterisk originated the call and it is not connected yet. * Made misdn_update_connected_line() send a FACILITY message if the call is already connected. This patch requires the presence of the associated mISDN patches to compile. I had to enhance mISDN to allow the notification indicator IE and the redirection number IE to be used with a FACILITY message. Earlier versions of the Digium enhanced mISDN are no longer going to work. ********** abe_2526_misdn.patch * Made an incoming FACILITY message allow the presence of the notification indicator IE and the redirection number IE. ********** abe_2526_misdnuser_v3.patch * Added support to send and receive a FACILITY message with the notification indicator IE and the redirection number IE. * Added the ability to send a NOTIFY message in PTMP/NT mode to all responding subcalls in Q.931 states 6, 7, 8, 9, and 25. ********** Patches: abe_2526_ast.patch uploaded by rmudgett (license 664) abe_2526_misdn.patch uploaded by rmudgett (license 664) abe_2526_misdnuser_v3.patch uploaded by rmudgett (license 664) Tested by: rmudgett and reporter JIRA SWP-2146 JIRA ABE-2526 .......... ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287018 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-15 20:56:21 +00:00
if (!bc->redirecting.to_changed) {
break;
}
Merged revisions 287017 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r287017 | rmudgett | 2010-09-15 15:53:38 -0500 (Wed, 15 Sep 2010) | 65 lines Merged revision 287014 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r287014 | rmudgett | 2010-09-15 15:32:24 -0500 (Wed, 15 Sep 2010) | 58 lines The handling of call transfer signaling for mISDN PTMP is not fully implemented. The handling of call transfer signaling for mISDN PTMP is not fully implemented. The signaling of number updates with ISDN/DSS1 ECT supplementary services (ETS 300 369-1) comes along with a notification indicator IE and redirection number IE for PTMP. The implementation in the current Asterisk mISDN channel unfortunately can handle these information elements only in a NOTIFY message. These information elements are also signaled in a FACILTY message with a RequestSubaddress facility, when the subscriber is already in the active state (see 9.2.4 and 9.2.5 of ETS 300 369-1). ********** abe_2526_ast.patch * Added support to handle the notification indicator IE and redirection number IE with the RequestSubaddress facility. * Made misdn_update_connected_line() send a NOTIFY message if Asterisk originated the call and it is not connected yet. * Made misdn_update_connected_line() send a FACILITY message if the call is already connected. This patch requires the presence of the associated mISDN patches to compile. I had to enhance mISDN to allow the notification indicator IE and the redirection number IE to be used with a FACILITY message. Earlier versions of the Digium enhanced mISDN are no longer going to work. ********** abe_2526_misdn.patch * Made an incoming FACILITY message allow the presence of the notification indicator IE and the redirection number IE. ********** abe_2526_misdnuser_v3.patch * Added support to send and receive a FACILITY message with the notification indicator IE and the redirection number IE. * Added the ability to send a NOTIFY message in PTMP/NT mode to all responding subcalls in Q.931 states 6, 7, 8, 9, and 25. ********** Patches: abe_2526_ast.patch uploaded by rmudgett (license 664) abe_2526_misdn.patch uploaded by rmudgett (license 664) abe_2526_misdnuser_v3.patch uploaded by rmudgett (license 664) Tested by: rmudgett and reporter JIRA SWP-2146 JIRA ABE-2526 .......... ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287018 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-15 20:56:21 +00:00
bc->redirecting.to_changed = 0;
if (!ch || !ch->ast) {
break;
}
switch (ch->state) {
case MISDN_ALERTING:
/* Call is deflecting after we have seen an ALERTING message */
bc->redirecting.reason = mISDN_REDIRECTING_REASON_NO_REPLY;
break;
default:
/* Call is deflecting for call forwarding unconditional or busy reason. */
bc->redirecting.reason = mISDN_REDIRECTING_REASON_UNKNOWN;
break;
}
misdn_copy_redirecting_to_ast(ch->ast, &bc->redirecting, bc->incoming_cid_tag);
Add private representation of caller, connected and redirecting party ids. This patch adds the feature "Private representation of caller, connected and redirecting party ids", as previously discussed with us (DATUS) and Digium. 1. Feature motivation Until now it is quite difficult to modify a party number or name which can only be seen by exactly one particular instantiated technology channel subscriber. One example where a modified party number or name on one channel is spread over several channels are supplementary services like call transfer or pickup. To implement these features Asterisk internally copies caller and connected ids from one channel to another. Another example are extension subscriptions. The monitoring entities (watchers) are notified of state changes and - if desired - of party numbers or names which represent the involving call parties. One major feature where a private representation of party names is essentially needed, i.e. where a party name shall be exclusively signaled to only one particular user, is a private user-specific name resolution for party numbers. A lookup in a private destination-dependent telephone book shall provide party names which cannot be seen by any other user at any time. 2. Feature Description This feature comes along with the implementation of additional private party id elements for caller id, connected id and redirecting ids inside Asterisk channels. The private party id elements can be read or set by the user using Asterisk dialplan functions. When a technology channel is initiating a call, receives an internal connected-line update event, or receives an internal redirecting update event, it merges the corresponding public id with the private id to create an effective party id. The effective party id is then used for protocol signaling. The channel technologies which initially support the private id representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and PRI (chan_dahdi). Once a private name or number on a channel is set and (implicitly) made valid, it is generally used for any further protocol signaling until it is rewritten or invalidated. To simplify the invalidation of private ids all internally generated connected/redirecting update events and also all connected/redirecting update events which are generated by technology channels -- receiving regarding protocol information - automatically trigger the invalidation of private ids. If not using the private party id representation feature at all, i.e. if using only the 'regular' caller-id, connected and redirecting related functions, the current characteristic of Asterisk is not affected by the new extended functionality. 3. User interface Description To grant access to the private name and number representation from the Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan functions are extended by the following data types. The formats of these data types are equal to the corresponding regular 'non-private' already existing data types: CALLERID: priv-all priv-name priv-name-valid priv-name-charset priv-name-pres priv-num priv-num-valid priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE: priv-name priv-name-valid priv-name-pres priv-name-charset priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag REDIRECTING: priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd priv-orig-tag priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd priv-to-tag Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2030/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-10 19:54:55 +00:00
ast_party_redirecting_init(&redirecting);
ast_party_redirecting_copy(&redirecting, ast_channel_redirecting(ch->ast));
/*
* Reset any earlier private redirecting id representations and
* make sure that it is invalidated at the remote end.
*/
ast_party_id_reset(&redirecting.priv_orig);
ast_party_id_reset(&redirecting.priv_from);
ast_party_id_reset(&redirecting.priv_to);
ast_channel_queue_redirecting_update(ch->ast, &redirecting, NULL);
ast_party_redirecting_free(&redirecting);
break;
Add private representation of caller, connected and redirecting party ids. This patch adds the feature "Private representation of caller, connected and redirecting party ids", as previously discussed with us (DATUS) and Digium. 1. Feature motivation Until now it is quite difficult to modify a party number or name which can only be seen by exactly one particular instantiated technology channel subscriber. One example where a modified party number or name on one channel is spread over several channels are supplementary services like call transfer or pickup. To implement these features Asterisk internally copies caller and connected ids from one channel to another. Another example are extension subscriptions. The monitoring entities (watchers) are notified of state changes and - if desired - of party numbers or names which represent the involving call parties. One major feature where a private representation of party names is essentially needed, i.e. where a party name shall be exclusively signaled to only one particular user, is a private user-specific name resolution for party numbers. A lookup in a private destination-dependent telephone book shall provide party names which cannot be seen by any other user at any time. 2. Feature Description This feature comes along with the implementation of additional private party id elements for caller id, connected id and redirecting ids inside Asterisk channels. The private party id elements can be read or set by the user using Asterisk dialplan functions. When a technology channel is initiating a call, receives an internal connected-line update event, or receives an internal redirecting update event, it merges the corresponding public id with the private id to create an effective party id. The effective party id is then used for protocol signaling. The channel technologies which initially support the private id representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and PRI (chan_dahdi). Once a private name or number on a channel is set and (implicitly) made valid, it is generally used for any further protocol signaling until it is rewritten or invalidated. To simplify the invalidation of private ids all internally generated connected/redirecting update events and also all connected/redirecting update events which are generated by technology channels -- receiving regarding protocol information - automatically trigger the invalidation of private ids. If not using the private party id representation feature at all, i.e. if using only the 'regular' caller-id, connected and redirecting related functions, the current characteristic of Asterisk is not affected by the new extended functionality. 3. User interface Description To grant access to the private name and number representation from the Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan functions are extended by the following data types. The formats of these data types are equal to the corresponding regular 'non-private' already existing data types: CALLERID: priv-all priv-name priv-name-valid priv-name-charset priv-name-pres priv-num priv-num-valid priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE: priv-name priv-name-valid priv-name-pres priv-name-charset priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag REDIRECTING: priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd priv-orig-tag priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd priv-to-tag Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2030/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-10 19:54:55 +00:00
}
case mISDN_NOTIFY_CODE_CALL_TRANSFER_ALERTING:
/*
* It would be preferable to update the connected line information
* only when the message callStatus is active. However, the
* optional redirection number may not be present in the active
* message if an alerting message were received earlier.
*
* The consequences if we wind up sending two updates is benign.
* The other end will think that it got transferred twice.
*/
Merged revisions 287017 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r287017 | rmudgett | 2010-09-15 15:53:38 -0500 (Wed, 15 Sep 2010) | 65 lines Merged revision 287014 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r287014 | rmudgett | 2010-09-15 15:32:24 -0500 (Wed, 15 Sep 2010) | 58 lines The handling of call transfer signaling for mISDN PTMP is not fully implemented. The handling of call transfer signaling for mISDN PTMP is not fully implemented. The signaling of number updates with ISDN/DSS1 ECT supplementary services (ETS 300 369-1) comes along with a notification indicator IE and redirection number IE for PTMP. The implementation in the current Asterisk mISDN channel unfortunately can handle these information elements only in a NOTIFY message. These information elements are also signaled in a FACILTY message with a RequestSubaddress facility, when the subscriber is already in the active state (see 9.2.4 and 9.2.5 of ETS 300 369-1). ********** abe_2526_ast.patch * Added support to handle the notification indicator IE and redirection number IE with the RequestSubaddress facility. * Made misdn_update_connected_line() send a NOTIFY message if Asterisk originated the call and it is not connected yet. * Made misdn_update_connected_line() send a FACILITY message if the call is already connected. This patch requires the presence of the associated mISDN patches to compile. I had to enhance mISDN to allow the notification indicator IE and the redirection number IE to be used with a FACILITY message. Earlier versions of the Digium enhanced mISDN are no longer going to work. ********** abe_2526_misdn.patch * Made an incoming FACILITY message allow the presence of the notification indicator IE and the redirection number IE. ********** abe_2526_misdnuser_v3.patch * Added support to send and receive a FACILITY message with the notification indicator IE and the redirection number IE. * Added the ability to send a NOTIFY message in PTMP/NT mode to all responding subcalls in Q.931 states 6, 7, 8, 9, and 25. ********** Patches: abe_2526_ast.patch uploaded by rmudgett (license 664) abe_2526_misdn.patch uploaded by rmudgett (license 664) abe_2526_misdnuser_v3.patch uploaded by rmudgett (license 664) Tested by: rmudgett and reporter JIRA SWP-2146 JIRA ABE-2526 .......... ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287018 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-15 20:56:21 +00:00
if (!bc->redirecting.to_changed) {
break;
}
Merged revisions 287017 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r287017 | rmudgett | 2010-09-15 15:53:38 -0500 (Wed, 15 Sep 2010) | 65 lines Merged revision 287014 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r287014 | rmudgett | 2010-09-15 15:32:24 -0500 (Wed, 15 Sep 2010) | 58 lines The handling of call transfer signaling for mISDN PTMP is not fully implemented. The handling of call transfer signaling for mISDN PTMP is not fully implemented. The signaling of number updates with ISDN/DSS1 ECT supplementary services (ETS 300 369-1) comes along with a notification indicator IE and redirection number IE for PTMP. The implementation in the current Asterisk mISDN channel unfortunately can handle these information elements only in a NOTIFY message. These information elements are also signaled in a FACILTY message with a RequestSubaddress facility, when the subscriber is already in the active state (see 9.2.4 and 9.2.5 of ETS 300 369-1). ********** abe_2526_ast.patch * Added support to handle the notification indicator IE and redirection number IE with the RequestSubaddress facility. * Made misdn_update_connected_line() send a NOTIFY message if Asterisk originated the call and it is not connected yet. * Made misdn_update_connected_line() send a FACILITY message if the call is already connected. This patch requires the presence of the associated mISDN patches to compile. I had to enhance mISDN to allow the notification indicator IE and the redirection number IE to be used with a FACILITY message. Earlier versions of the Digium enhanced mISDN are no longer going to work. ********** abe_2526_misdn.patch * Made an incoming FACILITY message allow the presence of the notification indicator IE and the redirection number IE. ********** abe_2526_misdnuser_v3.patch * Added support to send and receive a FACILITY message with the notification indicator IE and the redirection number IE. * Added the ability to send a NOTIFY message in PTMP/NT mode to all responding subcalls in Q.931 states 6, 7, 8, 9, and 25. ********** Patches: abe_2526_ast.patch uploaded by rmudgett (license 664) abe_2526_misdn.patch uploaded by rmudgett (license 664) abe_2526_misdnuser_v3.patch uploaded by rmudgett (license 664) Tested by: rmudgett and reporter JIRA SWP-2146 JIRA ABE-2526 .......... ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287018 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-15 20:56:21 +00:00
bc->redirecting.to_changed = 0;
if (!ch || !ch->ast) {
break;
}
misdn_update_remote_party(ch->ast, &bc->redirecting.to,
AST_CONNECTED_LINE_UPDATE_SOURCE_TRANSFER_ALERTING,
bc->incoming_cid_tag);
break;
case mISDN_NOTIFY_CODE_CALL_TRANSFER_ACTIVE:
Merged revisions 287017 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r287017 | rmudgett | 2010-09-15 15:53:38 -0500 (Wed, 15 Sep 2010) | 65 lines Merged revision 287014 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r287014 | rmudgett | 2010-09-15 15:32:24 -0500 (Wed, 15 Sep 2010) | 58 lines The handling of call transfer signaling for mISDN PTMP is not fully implemented. The handling of call transfer signaling for mISDN PTMP is not fully implemented. The signaling of number updates with ISDN/DSS1 ECT supplementary services (ETS 300 369-1) comes along with a notification indicator IE and redirection number IE for PTMP. The implementation in the current Asterisk mISDN channel unfortunately can handle these information elements only in a NOTIFY message. These information elements are also signaled in a FACILTY message with a RequestSubaddress facility, when the subscriber is already in the active state (see 9.2.4 and 9.2.5 of ETS 300 369-1). ********** abe_2526_ast.patch * Added support to handle the notification indicator IE and redirection number IE with the RequestSubaddress facility. * Made misdn_update_connected_line() send a NOTIFY message if Asterisk originated the call and it is not connected yet. * Made misdn_update_connected_line() send a FACILITY message if the call is already connected. This patch requires the presence of the associated mISDN patches to compile. I had to enhance mISDN to allow the notification indicator IE and the redirection number IE to be used with a FACILITY message. Earlier versions of the Digium enhanced mISDN are no longer going to work. ********** abe_2526_misdn.patch * Made an incoming FACILITY message allow the presence of the notification indicator IE and the redirection number IE. ********** abe_2526_misdnuser_v3.patch * Added support to send and receive a FACILITY message with the notification indicator IE and the redirection number IE. * Added the ability to send a NOTIFY message in PTMP/NT mode to all responding subcalls in Q.931 states 6, 7, 8, 9, and 25. ********** Patches: abe_2526_ast.patch uploaded by rmudgett (license 664) abe_2526_misdn.patch uploaded by rmudgett (license 664) abe_2526_misdnuser_v3.patch uploaded by rmudgett (license 664) Tested by: rmudgett and reporter JIRA SWP-2146 JIRA ABE-2526 .......... ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287018 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-15 20:56:21 +00:00
if (!bc->redirecting.to_changed) {
break;
}
Merged revisions 287017 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r287017 | rmudgett | 2010-09-15 15:53:38 -0500 (Wed, 15 Sep 2010) | 65 lines Merged revision 287014 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r287014 | rmudgett | 2010-09-15 15:32:24 -0500 (Wed, 15 Sep 2010) | 58 lines The handling of call transfer signaling for mISDN PTMP is not fully implemented. The handling of call transfer signaling for mISDN PTMP is not fully implemented. The signaling of number updates with ISDN/DSS1 ECT supplementary services (ETS 300 369-1) comes along with a notification indicator IE and redirection number IE for PTMP. The implementation in the current Asterisk mISDN channel unfortunately can handle these information elements only in a NOTIFY message. These information elements are also signaled in a FACILTY message with a RequestSubaddress facility, when the subscriber is already in the active state (see 9.2.4 and 9.2.5 of ETS 300 369-1). ********** abe_2526_ast.patch * Added support to handle the notification indicator IE and redirection number IE with the RequestSubaddress facility. * Made misdn_update_connected_line() send a NOTIFY message if Asterisk originated the call and it is not connected yet. * Made misdn_update_connected_line() send a FACILITY message if the call is already connected. This patch requires the presence of the associated mISDN patches to compile. I had to enhance mISDN to allow the notification indicator IE and the redirection number IE to be used with a FACILITY message. Earlier versions of the Digium enhanced mISDN are no longer going to work. ********** abe_2526_misdn.patch * Made an incoming FACILITY message allow the presence of the notification indicator IE and the redirection number IE. ********** abe_2526_misdnuser_v3.patch * Added support to send and receive a FACILITY message with the notification indicator IE and the redirection number IE. * Added the ability to send a NOTIFY message in PTMP/NT mode to all responding subcalls in Q.931 states 6, 7, 8, 9, and 25. ********** Patches: abe_2526_ast.patch uploaded by rmudgett (license 664) abe_2526_misdn.patch uploaded by rmudgett (license 664) abe_2526_misdnuser_v3.patch uploaded by rmudgett (license 664) Tested by: rmudgett and reporter JIRA SWP-2146 JIRA ABE-2526 .......... ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287018 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-15 20:56:21 +00:00
bc->redirecting.to_changed = 0;
if (!ch || !ch->ast) {
break;
}
misdn_update_remote_party(ch->ast, &bc->redirecting.to,
AST_CONNECTED_LINE_UPDATE_SOURCE_TRANSFER, bc->incoming_cid_tag);
break;
default:
bc->redirecting.to_changed = 0;
chan_misdn_log(0, bc->port," --> not yet handled: notify code:0x%02X\n",
bc->notify_description_code);
break;
}
Merged revisions 287017 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r287017 | rmudgett | 2010-09-15 15:53:38 -0500 (Wed, 15 Sep 2010) | 65 lines Merged revision 287014 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r287014 | rmudgett | 2010-09-15 15:32:24 -0500 (Wed, 15 Sep 2010) | 58 lines The handling of call transfer signaling for mISDN PTMP is not fully implemented. The handling of call transfer signaling for mISDN PTMP is not fully implemented. The signaling of number updates with ISDN/DSS1 ECT supplementary services (ETS 300 369-1) comes along with a notification indicator IE and redirection number IE for PTMP. The implementation in the current Asterisk mISDN channel unfortunately can handle these information elements only in a NOTIFY message. These information elements are also signaled in a FACILTY message with a RequestSubaddress facility, when the subscriber is already in the active state (see 9.2.4 and 9.2.5 of ETS 300 369-1). ********** abe_2526_ast.patch * Added support to handle the notification indicator IE and redirection number IE with the RequestSubaddress facility. * Made misdn_update_connected_line() send a NOTIFY message if Asterisk originated the call and it is not connected yet. * Made misdn_update_connected_line() send a FACILITY message if the call is already connected. This patch requires the presence of the associated mISDN patches to compile. I had to enhance mISDN to allow the notification indicator IE and the redirection number IE to be used with a FACILITY message. Earlier versions of the Digium enhanced mISDN are no longer going to work. ********** abe_2526_misdn.patch * Made an incoming FACILITY message allow the presence of the notification indicator IE and the redirection number IE. ********** abe_2526_misdnuser_v3.patch * Added support to send and receive a FACILITY message with the notification indicator IE and the redirection number IE. * Added the ability to send a NOTIFY message in PTMP/NT mode to all responding subcalls in Q.931 states 6, 7, 8, 9, and 25. ********** Patches: abe_2526_ast.patch uploaded by rmudgett (license 664) abe_2526_misdn.patch uploaded by rmudgett (license 664) abe_2526_misdnuser_v3.patch uploaded by rmudgett (license 664) Tested by: rmudgett and reporter JIRA SWP-2146 JIRA ABE-2526 .......... ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287018 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-15 20:56:21 +00:00
bc->notify_description_code = mISDN_NOTIFY_CODE_INVALID;
break;
case EVENT_FACILITY:
if (bc->fac_in.Function == Fac_None) {
/* This is a FACILITY message so we MUST have a facility ie */
chan_misdn_log(0, bc->port," --> Missing facility ie or unknown facility ie contents.\n");
} else {
misdn_facility_ie_handler(event, bc, ch);
}
Merged revisions 287017 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r287017 | rmudgett | 2010-09-15 15:53:38 -0500 (Wed, 15 Sep 2010) | 65 lines Merged revision 287014 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r287014 | rmudgett | 2010-09-15 15:32:24 -0500 (Wed, 15 Sep 2010) | 58 lines The handling of call transfer signaling for mISDN PTMP is not fully implemented. The handling of call transfer signaling for mISDN PTMP is not fully implemented. The signaling of number updates with ISDN/DSS1 ECT supplementary services (ETS 300 369-1) comes along with a notification indicator IE and redirection number IE for PTMP. The implementation in the current Asterisk mISDN channel unfortunately can handle these information elements only in a NOTIFY message. These information elements are also signaled in a FACILTY message with a RequestSubaddress facility, when the subscriber is already in the active state (see 9.2.4 and 9.2.5 of ETS 300 369-1). ********** abe_2526_ast.patch * Added support to handle the notification indicator IE and redirection number IE with the RequestSubaddress facility. * Made misdn_update_connected_line() send a NOTIFY message if Asterisk originated the call and it is not connected yet. * Made misdn_update_connected_line() send a FACILITY message if the call is already connected. This patch requires the presence of the associated mISDN patches to compile. I had to enhance mISDN to allow the notification indicator IE and the redirection number IE to be used with a FACILITY message. Earlier versions of the Digium enhanced mISDN are no longer going to work. ********** abe_2526_misdn.patch * Made an incoming FACILITY message allow the presence of the notification indicator IE and the redirection number IE. ********** abe_2526_misdnuser_v3.patch * Added support to send and receive a FACILITY message with the notification indicator IE and the redirection number IE. * Added the ability to send a NOTIFY message in PTMP/NT mode to all responding subcalls in Q.931 states 6, 7, 8, 9, and 25. ********** Patches: abe_2526_ast.patch uploaded by rmudgett (license 664) abe_2526_misdn.patch uploaded by rmudgett (license 664) abe_2526_misdnuser_v3.patch uploaded by rmudgett (license 664) Tested by: rmudgett and reporter JIRA SWP-2146 JIRA ABE-2526 .......... ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287018 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-15 20:56:21 +00:00
/* In case it came in on a FACILITY message and we did not handle it. */
bc->redirecting.to_changed = 0;
bc->notify_description_code = mISDN_NOTIFY_CODE_INVALID;
break;
case EVENT_RESTART:
if (!bc->dummy) {
stop_bc_tones(ch);
release_chan(ch, bc);
}
break;
default:
chan_misdn_log(1, 0, "Got Unknown Event\n");
break;
}
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
if (ch) {
chan_list_unref(ch, "cb_event complete OK");
}
return RESPONSE_OK;
}
/** TE STUFF END **/
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief Get call completion record information.
*
* \param chan Asterisk channel to operate upon. (Not used)
* \param function_name Name of the function that called us.
* \param function_args Argument string passed to function (Could be NULL)
* \param buf Buffer to put returned string.
* \param size Size of the supplied buffer including the null terminator.
*
* \retval 0 on success.
* \retval -1 on error.
*/
static int misdn_cc_read(struct ast_channel *chan, const char *function_name,
char *function_args, char *buf, size_t size)
{
char *parse;
struct misdn_cc_record *cc_record;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(cc_id); /* Call completion record ID value. */
AST_APP_ARG(get_name); /* Name of what to get */
AST_APP_ARG(other); /* Any extraneous garbage arguments */
);
/* Ensure that the buffer is empty */
*buf = 0;
if (ast_strlen_zero(function_args)) {
ast_log(LOG_ERROR, "Function '%s' requires arguments.\n", function_name);
return -1;
}
parse = ast_strdupa(function_args);
AST_STANDARD_APP_ARGS(args, parse);
if (!args.argc || ast_strlen_zero(args.cc_id)) {
ast_log(LOG_ERROR, "Function '%s' missing call completion record ID.\n",
function_name);
return -1;
}
if (!isdigit(*args.cc_id)) {
ast_log(LOG_ERROR, "Function '%s' call completion record ID must be numeric.\n",
function_name);
return -1;
}
if (ast_strlen_zero(args.get_name)) {
ast_log(LOG_ERROR, "Function '%s' missing what-to-get parameter.\n",
function_name);
return -1;
}
AST_LIST_LOCK(&misdn_cc_records_db);
cc_record = misdn_cc_find_by_id(atoi(args.cc_id));
if (cc_record) {
if (!strcasecmp("a-all", args.get_name)) {
snprintf(buf, size, "\"%s\" <%s>", cc_record->redial.caller.name,
cc_record->redial.caller.number);
} else if (!strcasecmp("a-name", args.get_name)) {
ast_copy_string(buf, cc_record->redial.caller.name, size);
} else if (!strncasecmp("a-num", args.get_name, 5)) {
ast_copy_string(buf, cc_record->redial.caller.number, size);
} else if (!strcasecmp("a-ton", args.get_name)) {
snprintf(buf, size, "%d",
misdn_to_ast_plan(cc_record->redial.caller.number_plan)
| misdn_to_ast_ton(cc_record->redial.caller.number_type));
} else if (!strncasecmp("a-pres", args.get_name, 6)) {
ast_copy_string(buf, ast_named_caller_presentation(
misdn_to_ast_pres(cc_record->redial.caller.presentation)
| misdn_to_ast_screen(cc_record->redial.caller.screening)), size);
} else if (!strcasecmp("a-busy", args.get_name)) {
ast_copy_string(buf, cc_record->party_a_free ? "no" : "yes", size);
} else if (!strncasecmp("b-num", args.get_name, 5)) {
ast_copy_string(buf, cc_record->redial.dialed.number, size);
} else if (!strcasecmp("b-ton", args.get_name)) {
snprintf(buf, size, "%d",
misdn_to_ast_plan(cc_record->redial.dialed.number_plan)
| misdn_to_ast_ton(cc_record->redial.dialed.number_type));
} else if (!strcasecmp("port", args.get_name)) {
snprintf(buf, size, "%d", cc_record->port);
} else if (!strcasecmp("available-notify-priority", args.get_name)) {
snprintf(buf, size, "%d", cc_record->remote_user_free.priority);
} else if (!strcasecmp("available-notify-exten", args.get_name)) {
ast_copy_string(buf, cc_record->remote_user_free.exten, size);
} else if (!strcasecmp("available-notify-context", args.get_name)) {
ast_copy_string(buf, cc_record->remote_user_free.context, size);
} else if (!strcasecmp("busy-notify-priority", args.get_name)) {
snprintf(buf, size, "%d", cc_record->b_free.priority);
} else if (!strcasecmp("busy-notify-exten", args.get_name)) {
ast_copy_string(buf, cc_record->b_free.exten, size);
} else if (!strcasecmp("busy-notify-context", args.get_name)) {
ast_copy_string(buf, cc_record->b_free.context, size);
} else {
AST_LIST_UNLOCK(&misdn_cc_records_db);
ast_log(LOG_ERROR, "Function '%s': Unknown what-to-get '%s'.\n", function_name, args.get_name);
return -1;
}
}
AST_LIST_UNLOCK(&misdn_cc_records_db);
return 0;
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
static struct ast_custom_function misdn_cc_function = {
.name = "mISDN_CC",
.synopsis = "Get call completion record information.",
.syntax = "mISDN_CC(${MISDN_CC_RECORD_ID},<what-to-get>)",
.desc =
"mISDN_CC(${MISDN_CC_RECORD_ID},<what-to-get>)\n"
"The following can be retrieved:\n"
"\"a-num\", \"a-name\", \"a-all\", \"a-ton\", \"a-pres\", \"a-busy\",\n"
"\"b-num\", \"b-ton\", \"port\",\n"
" User-A is available for call completion:\n"
" \"available-notify-priority\",\n"
" \"available-notify-exten\",\n"
" \"available-notify-context\",\n"
" User-A is busy:\n"
" \"busy-notify-priority\",\n"
" \"busy-notify-exten\",\n"
" \"busy-notify-context\"\n",
.read = misdn_cc_read,
};
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
/******************************************
*
* Asterisk Channel Endpoint END
*
*
*******************************************/
static int unload_module(void)
{
/* First, take us out of the channel loop */
ast_verb(0, "-- Unregistering mISDN Channel Driver --\n");
misdn_tasks_destroy();
if (!g_config_initialized) {
return 0;
}
ast_cli_unregister_multiple(chan_misdn_clis, sizeof(chan_misdn_clis) / sizeof(struct ast_cli_entry));
/* ast_unregister_application("misdn_crypt"); */
ast_unregister_application("misdn_set_opt");
ast_unregister_application("misdn_facility");
ast_unregister_application("misdn_check_l2l1");
#if defined(AST_MISDN_ENHANCEMENTS)
ast_unregister_application(misdn_command_name);
ast_custom_function_unregister(&misdn_cc_function);
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
ast_channel_unregister(&misdn_tech);
free_robin_list();
misdn_cfg_destroy();
misdn_lib_destroy();
ast_free(misdn_out_calls);
ast_free(misdn_in_calls);
ast_free(misdn_debug_only);
ast_free(misdn_ports);
ast_free(misdn_debug);
#if defined(AST_MISDN_ENHANCEMENTS)
misdn_cc_destroy();
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
misdn_tech.capabilities = ast_format_cap_destroy(misdn_tech.capabilities);
return 0;
}
/*!
* \brief Load the module
*
* Module loading including tests for configuration or dependencies.
* This function can return AST_MODULE_LOAD_FAILURE, AST_MODULE_LOAD_DECLINE,
* or AST_MODULE_LOAD_SUCCESS. If a dependency or environment variable fails
* tests return AST_MODULE_LOAD_FAILURE. If the module can not load the
* configuration file or other non-critical problem return
* AST_MODULE_LOAD_DECLINE. On success return AST_MODULE_LOAD_SUCCESS.
*/
static int load_module(void)
{
int i, port;
int ntflags = 0, ntkc = 0;
char ports[256] = "";
char tempbuf[BUFFERSIZE + 1];
char ntfile[BUFFERSIZE + 1];
struct misdn_lib_iface iface = {
.cb_event = cb_events,
.cb_log = chan_misdn_log,
.cb_jb_empty = chan_misdn_jb_empty,
};
if (!(misdn_tech.capabilities = ast_format_cap_alloc(0))) {
return AST_MODULE_LOAD_DECLINE;
}
ast_format_set(&prefformat, AST_FORMAT_ALAW, 0);
ast_format_cap_add(misdn_tech.capabilities, &prefformat);
max_ports = misdn_lib_maxports_get();
if (max_ports <= 0) {
ast_log(LOG_ERROR, "Unable to initialize mISDN\n");
return AST_MODULE_LOAD_DECLINE;
}
if (misdn_cfg_init(max_ports, 0)) {
ast_log(LOG_ERROR, "Unable to initialize misdn_config.\n");
return AST_MODULE_LOAD_DECLINE;
}
g_config_initialized = 1;
#if defined(AST_MISDN_ENHANCEMENTS)
misdn_cc_init();
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
misdn_debug = ast_malloc(sizeof(int) * (max_ports + 1));
if (!misdn_debug) {
ast_log(LOG_ERROR, "Out of memory for misdn_debug\n");
return AST_MODULE_LOAD_DECLINE;
}
misdn_ports = ast_malloc(sizeof(int) * (max_ports + 1));
if (!misdn_ports) {
ast_free(misdn_debug);
ast_log(LOG_ERROR, "Out of memory for misdn_ports\n");
return AST_MODULE_LOAD_DECLINE;
}
misdn_cfg_get(0, MISDN_GEN_DEBUG, &misdn_debug[0], sizeof(misdn_debug[0]));
for (i = 1; i <= max_ports; i++) {
misdn_debug[i] = misdn_debug[0];
misdn_ports[i] = i;
}
*misdn_ports = 0;
misdn_debug_only = ast_calloc(max_ports + 1, sizeof(int));
if (!misdn_debug_only) {
ast_free(misdn_ports);
ast_free(misdn_debug);
ast_log(LOG_ERROR, "Out of memory for misdn_debug_only\n");
return AST_MODULE_LOAD_DECLINE;
}
misdn_cfg_get(0, MISDN_GEN_TRACEFILE, tempbuf, sizeof(tempbuf));
if (!ast_strlen_zero(tempbuf)) {
tracing = 1;
}
misdn_in_calls = ast_malloc(sizeof(int) * (max_ports + 1));
if (!misdn_in_calls) {
ast_free(misdn_debug_only);
ast_free(misdn_ports);
ast_free(misdn_debug);
ast_log(LOG_ERROR, "Out of memory for misdn_in_calls\n");
return AST_MODULE_LOAD_DECLINE;
}
misdn_out_calls = ast_malloc(sizeof(int) * (max_ports + 1));
if (!misdn_out_calls) {
ast_free(misdn_in_calls);
ast_free(misdn_debug_only);
ast_free(misdn_ports);
ast_free(misdn_debug);
ast_log(LOG_ERROR, "Out of memory for misdn_out_calls\n");
return AST_MODULE_LOAD_DECLINE;
}
for (i = 1; i <= max_ports; i++) {
misdn_in_calls[i] = 0;
misdn_out_calls[i] = 0;
}
ast_mutex_init(&cl_te_lock);
ast_mutex_init(&release_lock);
misdn_cfg_update_ptp();
misdn_cfg_get_ports_string(ports);
if (!ast_strlen_zero(ports)) {
chan_misdn_log(0, 0, "Got: %s from get_ports\n", ports);
}
if (misdn_lib_init(ports, &iface, NULL)) {
chan_misdn_log(0, 0, "No te ports initialized\n");
}
misdn_cfg_get(0, MISDN_GEN_NTDEBUGFLAGS, &ntflags, sizeof(ntflags));
misdn_cfg_get(0, MISDN_GEN_NTDEBUGFILE, &ntfile, sizeof(ntfile));
misdn_cfg_get(0, MISDN_GEN_NTKEEPCALLS, &ntkc, sizeof(ntkc));
misdn_lib_nt_keepcalls(ntkc);
misdn_lib_nt_debug_init(ntflags, ntfile);
if (ast_channel_register(&misdn_tech)) {
ast_log(LOG_ERROR, "Unable to register channel class %s\n", misdn_type);
unload_module();
return AST_MODULE_LOAD_DECLINE;
}
ast_cli_register_multiple(chan_misdn_clis, sizeof(chan_misdn_clis) / sizeof(struct ast_cli_entry));
ast_register_application("misdn_set_opt", misdn_set_opt_exec, "misdn_set_opt",
"misdn_set_opt(:<opt><optarg>:<opt><optarg>...):\n"
"Sets mISDN opts. and optargs\n"
"\n"
"The available options are:\n"
" a - Have Asterisk detect DTMF tones on called channel\n"
" c - Make crypted outgoing call, optarg is keyindex\n"
" d - Send display text to called phone, text is the optarg\n"
" e - Perform echo cancellation on this channel,\n"
" takes taps as optarg (32,64,128,256)\n"
" e! - Disable echo cancellation on this channel\n"
" f - Enable fax detection\n"
" h - Make digital outgoing call\n"
" h1 - Make HDLC mode digital outgoing call\n"
" i - Ignore detected DTMF tones, don't signal them to Asterisk,\n"
" they will be transported inband.\n"
" jb - Set jitter buffer length, optarg is length\n"
" jt - Set jitter buffer upper threshold, optarg is threshold\n"
" jn - Disable jitter buffer\n"
" n - Disable mISDN DSP on channel.\n"
" Disables: echo cancel, DTMF detection, and volume control.\n"
" p - Caller ID presentation,\n"
" optarg is either 'allowed' or 'restricted'\n"
" s - Send Non-inband DTMF as inband\n"
" vr - Rx gain control, optarg is gain\n"
" vt - Tx gain control, optarg is gain\n"
);
ast_register_application("misdn_facility", misdn_facility_exec, "misdn_facility",
"misdn_facility(<FACILITY_TYPE>|<ARG1>|..)\n"
"Sends the Facility Message FACILITY_TYPE with \n"
"the given Arguments to the current ISDN Channel\n"
"Supported Facilities are:\n"
"\n"
"type=calldeflect args=Nr where to deflect\n"
#if defined(AST_MISDN_ENHANCEMENTS)
"type=callrerouting args=Nr where to deflect\n"
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
);
ast_register_application("misdn_check_l2l1", misdn_check_l2l1, "misdn_check_l2l1",
"misdn_check_l2l1(<port>||g:<groupname>,timeout)\n"
"Checks if the L2 and L1 are up on either the given <port> or\n"
"on the ports in the group with <groupname>\n"
"If the L1/L2 are down, check_l2l1 gets up the L1/L2 and waits\n"
"for <timeout> seconds that this happens. Otherwise, nothing happens\n"
"\n"
"This application, ensures the L1/L2 state of the Ports in a group\n"
"it is intended to make the pmp_l1_check option redundant and to\n"
"fix a buggy switch config from your provider\n"
"\n"
"a sample dialplan would look like:\n\n"
"exten => _X.,1,misdn_check_l2l1(g:out|2)\n"
"exten => _X.,n,dial(mISDN/g:out/${EXTEN})\n"
);
#if defined(AST_MISDN_ENHANCEMENTS)
ast_register_application(misdn_command_name, misdn_command_exec, misdn_command_name,
"misdn_command(<command>[,<options>])\n"
"The following commands are defined:\n"
"cc-initialize\n"
" Setup mISDN support for call completion\n"
" Must call before doing any Dial() involving call completion.\n"
"ccnr-request,${MISDN_CC_RECORD_ID},<notify-context>,<user-a-extension>,<priority>\n"
" Request Call Completion No Reply activation\n"
"ccbs-request,${MISDN_CC_RECORD_ID},<notify-context>,<user-a-extension>,<priority>\n"
" Request Call Completion Busy Subscriber activation\n"
"cc-b-free,${MISDN_CC_RECORD_ID},<notify-context>,<user-a-extension>,<priority>\n"
" Set the dialplan location to notify when User-B is available but User-A is busy.\n"
" Setting this dialplan location is optional.\n"
"cc-a-busy,${MISDN_CC_RECORD_ID},<yes/no>\n"
" Set the busy status of call completion User-A\n"
"cc-deactivate,${MISDN_CC_RECORD_ID}\n"
" Deactivate the identified call completion request\n"
"\n"
"MISDN_CC_RECORD_ID is set when Dial() returns and call completion is possible\n"
"MISDN_CC_STATUS is set to ACTIVATED or ERROR after the call completion\n"
"activation request.\n"
"MISDN_ERROR_MSG is set to a descriptive message on error.\n"
);
ast_custom_function_register(&misdn_cc_function);
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
misdn_cfg_get(0, MISDN_GEN_TRACEFILE, global_tracefile, sizeof(global_tracefile));
/* start the l1 watchers */
for (port = misdn_cfg_get_next_port(0); port >= 0; port = misdn_cfg_get_next_port(port)) {
int l1timeout;
misdn_cfg_get(port, MISDN_CFG_L1_TIMEOUT, &l1timeout, sizeof(l1timeout));
if (l1timeout) {
chan_misdn_log(4, 0, "Adding L1watcher task: port:%d timeout:%ds\n", port, l1timeout);
misdn_tasks_add(l1timeout * 1000, misdn_l1_task, &misdn_ports[port]);
}
}
chan_misdn_log(0, 0, "-- mISDN Channel Driver Registered --\n");
return 0;
}
static int reload(void)
{
reload_config();
return 0;
}
/*** SOME APPS ;)***/
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \brief misdn_command arguments container.
*/
AST_DEFINE_APP_ARGS_TYPE(misdn_command_args,
AST_APP_ARG(name); /* Subcommand name */
AST_APP_ARG(arg)[10 + 1]; /* Subcommand arguments */
);
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
static void misdn_cc_caller_destroy(void *obj)
{
/* oh snap! */
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
static struct misdn_cc_caller *misdn_cc_caller_alloc(struct ast_channel *chan)
{
struct misdn_cc_caller *cc_caller;
if (!(cc_caller = ao2_alloc(sizeof(*cc_caller), misdn_cc_caller_destroy))) {
return NULL;
}
cc_caller->chan = chan;
return cc_caller;
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief misdn_command(cc-initialize) subcommand handler
*
* \param chan Asterisk channel to operate upon.
* \param subcommand Arguments for the subcommand
*
* \retval 0 on success.
* \retval -1 on error.
*/
static int misdn_command_cc_initialize(struct ast_channel *chan, struct misdn_command_args *subcommand)
{
struct misdn_cc_caller *cc_caller;
struct ast_datastore *datastore;
if (!(cc_caller = misdn_cc_caller_alloc(chan))) {
return -1;
}
if (!(datastore = ast_datastore_alloc(&misdn_cc_ds_info, NULL))) {
ao2_ref(cc_caller, -1);
return -1;
}
ast_channel_lock(chan);
/* Inherit reference */
datastore->data = cc_caller;
cc_caller = NULL;
datastore->inheritance = DATASTORE_INHERIT_FOREVER;
ast_channel_datastore_add(chan, datastore);
ast_channel_unlock(chan);
return 0;
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief misdn_command(cc-deactivate) subcommand handler
*
* \details
* misdn_command(cc-deactivate,${MISDN_CC_RECORD_ID})
* Deactivate a call completion service instance.
*
* \param chan Asterisk channel to operate upon.
* \param subcommand Arguments for the subcommand
*
* \retval 0 on success.
* \retval -1 on error.
*/
static int misdn_command_cc_deactivate(struct ast_channel *chan, struct misdn_command_args *subcommand)
{
long record_id;
const char *error_str;
struct misdn_cc_record *cc_record;
struct misdn_bchannel *bc;
struct misdn_bchannel dummy;
static const char cmd_help[] = "%s(%s,${MISDN_CC_RECORD_ID})\n";
if (ast_strlen_zero(subcommand->arg[0]) || !isdigit(*subcommand->arg[0])) {
ast_log(LOG_WARNING, cmd_help, misdn_command_name, subcommand->name);
return -1;
}
record_id = atol(subcommand->arg[0]);
AST_LIST_LOCK(&misdn_cc_records_db);
cc_record = misdn_cc_find_by_id(record_id);
if (cc_record && 0 <= cc_record->port) {
if (cc_record->ptp) {
if (cc_record->mode.ptp.bc) {
/* Close the call-completion signaling link */
bc = cc_record->mode.ptp.bc;
bc->fac_out.Function = Fac_None;
bc->out_cause = AST_CAUSE_NORMAL_CLEARING;
misdn_lib_send_event(bc, EVENT_RELEASE_COMPLETE);
}
misdn_cc_delete(cc_record);
} else if (cc_record->activated) {
cc_record->error_code = FacError_None;
cc_record->reject_code = FacReject_None;
cc_record->invoke_id = ++misdn_invoke_id;
cc_record->outstanding_message = 1;
/* Build message */
misdn_make_dummy(&dummy, cc_record->port, 0, misdn_lib_port_is_nt(cc_record->port), 0);
dummy.fac_out.Function = Fac_CCBSDeactivate;
dummy.fac_out.u.CCBSDeactivate.InvokeID = cc_record->invoke_id;
dummy.fac_out.u.CCBSDeactivate.ComponentType = FacComponent_Invoke;
dummy.fac_out.u.CCBSDeactivate.Component.Invoke.CCBSReference = cc_record->mode.ptmp.reference_id;
/* Send message */
print_facility(&dummy.fac_out, &dummy);
misdn_lib_send_event(&dummy, EVENT_FACILITY);
}
}
AST_LIST_UNLOCK(&misdn_cc_records_db);
/* Wait for the response to the call completion deactivation request. */
misdn_cc_response_wait(chan, MISDN_CC_REQUEST_WAIT_MAX, record_id);
AST_LIST_LOCK(&misdn_cc_records_db);
cc_record = misdn_cc_find_by_id(record_id);
if (cc_record) {
if (cc_record->port < 0) {
/* The network did not tell us that call completion was available. */
error_str = NULL;
} else if (cc_record->outstanding_message) {
cc_record->outstanding_message = 0;
error_str = misdn_no_response_from_network;
} else if (cc_record->reject_code != FacReject_None) {
error_str = misdn_to_str_reject_code(cc_record->reject_code);
} else if (cc_record->error_code != FacError_None) {
error_str = misdn_to_str_error_code(cc_record->error_code);
} else {
error_str = NULL;
}
misdn_cc_delete(cc_record);
} else {
error_str = NULL;
}
AST_LIST_UNLOCK(&misdn_cc_records_db);
if (error_str) {
ast_verb(1, "%s(%s) diagnostic '%s' on channel %s\n",
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
misdn_command_name, subcommand->name, error_str, ast_channel_name(chan));
pbx_builtin_setvar_helper(chan, MISDN_ERROR_MSG, error_str);
}
return 0;
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief misdn_command(cc-a-busy) subcommand handler
*
* \details
* misdn_command(cc-a-busy,${MISDN_CC_RECORD_ID},<yes/no>)
* Set the status of User-A for a call completion service instance.
*
* \param chan Asterisk channel to operate upon.
* \param subcommand Arguments for the subcommand
*
* \retval 0 on success.
* \retval -1 on error.
*/
static int misdn_command_cc_a_busy(struct ast_channel *chan, struct misdn_command_args *subcommand)
{
long record_id;
int party_a_free;
struct misdn_cc_record *cc_record;
struct misdn_bchannel *bc;
static const char cmd_help[] = "%s(%s,${MISDN_CC_RECORD_ID},<yes/no>)\n";
if (ast_strlen_zero(subcommand->arg[0]) || !isdigit(*subcommand->arg[0])) {
ast_log(LOG_WARNING, cmd_help, misdn_command_name, subcommand->name);
return -1;
}
record_id = atol(subcommand->arg[0]);
if (ast_true(subcommand->arg[1])) {
party_a_free = 0;
} else if (ast_false(subcommand->arg[1])) {
party_a_free = 1;
} else {
ast_log(LOG_WARNING, cmd_help, misdn_command_name, subcommand->name);
return -1;
}
AST_LIST_LOCK(&misdn_cc_records_db);
cc_record = misdn_cc_find_by_id(record_id);
if (cc_record && cc_record->party_a_free != party_a_free) {
/* User-A's status has changed */
cc_record->party_a_free = party_a_free;
if (cc_record->ptp && cc_record->mode.ptp.bc) {
cc_record->error_code = FacError_None;
cc_record->reject_code = FacReject_None;
/* Build message */
bc = cc_record->mode.ptp.bc;
if (cc_record->party_a_free) {
bc->fac_out.Function = Fac_CCBS_T_Resume;
bc->fac_out.u.CCBS_T_Resume.InvokeID = ++misdn_invoke_id;
} else {
bc->fac_out.Function = Fac_CCBS_T_Suspend;
bc->fac_out.u.CCBS_T_Suspend.InvokeID = ++misdn_invoke_id;
}
/* Send message */
print_facility(&bc->fac_out, bc);
misdn_lib_send_event(bc, EVENT_FACILITY);
}
}
AST_LIST_UNLOCK(&misdn_cc_records_db);
return 0;
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief misdn_command(cc-b-free) subcommand handler
*
* \details
* misdn_command(cc-b-free,${MISDN_CC_RECORD_ID},<notify-context>,<user-a-extension>,<priority>)
* Set the dialplan location to notify when User-B is free and User-A is busy.
*
* \param chan Asterisk channel to operate upon.
* \param subcommand Arguments for the subcommand
*
* \retval 0 on success.
* \retval -1 on error.
*/
static int misdn_command_cc_b_free(struct ast_channel *chan, struct misdn_command_args *subcommand)
{
unsigned index;
long record_id;
int priority;
char *context;
char *exten;
struct misdn_cc_record *cc_record;
static const char cmd_help[] = "%s(%s,${MISDN_CC_RECORD_ID},<notify-context>,<user-a-extension>,<priority>)\n";
/* Check that all arguments are present */
for (index = 0; index < 4; ++index) {
if (ast_strlen_zero(subcommand->arg[index])) {
ast_log(LOG_WARNING, cmd_help, misdn_command_name, subcommand->name);
return -1;
}
}
/* These must be numeric */
if (!isdigit(*subcommand->arg[0]) || !isdigit(*subcommand->arg[3])) {
ast_log(LOG_WARNING, cmd_help, misdn_command_name, subcommand->name);
return -1;
}
record_id = atol(subcommand->arg[0]);
context = subcommand->arg[1];
exten = subcommand->arg[2];
priority = atoi(subcommand->arg[3]);
AST_LIST_LOCK(&misdn_cc_records_db);
cc_record = misdn_cc_find_by_id(record_id);
if (cc_record) {
/* Save User-B free information */
ast_copy_string(cc_record->b_free.context, context, sizeof(cc_record->b_free.context));
ast_copy_string(cc_record->b_free.exten, exten, sizeof(cc_record->b_free.exten));
cc_record->b_free.priority = priority;
}
AST_LIST_UNLOCK(&misdn_cc_records_db);
return 0;
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
struct misdn_cc_request {
enum FacFunction ptmp;
enum FacFunction ptp;
};
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief misdn_command(ccbs-request/ccnr-request) subcommand handler helper
*
* \details
* misdn_command(ccbs-request,${MISDN_CC_RECORD_ID},<notify-context>,<user-a-extension>,<priority>)
* misdn_command(ccnr-request,${MISDN_CC_RECORD_ID},<notify-context>,<user-a-extension>,<priority>)
* Set the dialplan location to notify when User-B is free and User-A is free.
*
* \param chan Asterisk channel to operate upon.
* \param subcommand Arguments for the subcommand
* \param request Which call-completion request message to generate.
*
* \retval 0 on success.
* \retval -1 on error.
*/
static int misdn_command_cc_request(struct ast_channel *chan, struct misdn_command_args *subcommand, const struct misdn_cc_request *request)
{
unsigned index;
int request_retention;
long record_id;
int priority;
char *context;
char *exten;
const char *error_str;
struct misdn_cc_record *cc_record;
struct misdn_bchannel *bc;
struct misdn_bchannel dummy;
struct misdn_party_id id;
static const char cmd_help[] = "%s(%s,${MISDN_CC_RECORD_ID},<notify-context>,<user-a-extension>,<priority>)\n";
/* Check that all arguments are present */
for (index = 0; index < 4; ++index) {
if (ast_strlen_zero(subcommand->arg[index])) {
ast_log(LOG_WARNING, cmd_help, misdn_command_name, subcommand->name);
return -1;
}
}
/* These must be numeric */
if (!isdigit(*subcommand->arg[0]) || !isdigit(*subcommand->arg[3])) {
ast_log(LOG_WARNING, cmd_help, misdn_command_name, subcommand->name);
return -1;
}
record_id = atol(subcommand->arg[0]);
context = subcommand->arg[1];
exten = subcommand->arg[2];
priority = atoi(subcommand->arg[3]);
AST_LIST_LOCK(&misdn_cc_records_db);
cc_record = misdn_cc_find_by_id(record_id);
if (cc_record) {
/* Save User-B free information */
ast_copy_string(cc_record->remote_user_free.context, context,
sizeof(cc_record->remote_user_free.context));
ast_copy_string(cc_record->remote_user_free.exten, exten,
sizeof(cc_record->remote_user_free.exten));
cc_record->remote_user_free.priority = priority;
if (0 <= cc_record->port) {
if (cc_record->ptp) {
if (!cc_record->mode.ptp.bc) {
bc = misdn_lib_get_register_bc(cc_record->port);
if (bc) {
cc_record->mode.ptp.bc = bc;
cc_record->error_code = FacError_None;
cc_record->reject_code = FacReject_None;
cc_record->invoke_id = ++misdn_invoke_id;
cc_record->outstanding_message = 1;
cc_record->activation_requested = 1;
misdn_cfg_get(bc->port, MISDN_CFG_CC_REQUEST_RETENTION,
&request_retention, sizeof(request_retention));
cc_record->mode.ptp.requested_retention = request_retention ? 1 : 0;
/* Build message */
bc->fac_out.Function = request->ptp;
bc->fac_out.u.CCBS_T_Request.InvokeID = cc_record->invoke_id;
bc->fac_out.u.CCBS_T_Request.ComponentType = FacComponent_Invoke;
bc->fac_out.u.CCBS_T_Request.Component.Invoke.Q931ie =
cc_record->redial.setup_bc_hlc_llc;
memset(&id, 0, sizeof(id));
id.number_plan = cc_record->redial.dialed.number_plan;
id.number_type = cc_record->redial.dialed.number_type;
ast_copy_string(id.number, cc_record->redial.dialed.number,
sizeof(id.number));
misdn_Address_fill(
&bc->fac_out.u.CCBS_T_Request.Component.Invoke.Destination,
&id);
misdn_Address_fill(
&bc->fac_out.u.CCBS_T_Request.Component.Invoke.Originating,
&cc_record->redial.caller);
bc->fac_out.u.CCBS_T_Request.Component.Invoke.PresentationAllowedIndicatorPresent = 1;
bc->fac_out.u.CCBS_T_Request.Component.Invoke.PresentationAllowedIndicator =
(cc_record->redial.caller.presentation != 0) ? 0 : 1;
bc->fac_out.u.CCBS_T_Request.Component.Invoke.RetentionSupported =
request_retention ? 1 : 0;
/* Send message */
print_facility(&bc->fac_out, bc);
misdn_lib_send_event(bc, EVENT_REGISTER);
}
}
} else {
cc_record->error_code = FacError_None;
cc_record->reject_code = FacReject_None;
cc_record->invoke_id = ++misdn_invoke_id;
cc_record->outstanding_message = 1;
cc_record->activation_requested = 1;
/* Build message */
misdn_make_dummy(&dummy, cc_record->port, 0,
misdn_lib_port_is_nt(cc_record->port), 0);
dummy.fac_out.Function = request->ptmp;
dummy.fac_out.u.CCBSRequest.InvokeID = cc_record->invoke_id;
dummy.fac_out.u.CCBSRequest.ComponentType = FacComponent_Invoke;
dummy.fac_out.u.CCBSRequest.Component.Invoke.CallLinkageID =
cc_record->mode.ptmp.linkage_id;
/* Send message */
print_facility(&dummy.fac_out, &dummy);
misdn_lib_send_event(&dummy, EVENT_FACILITY);
}
}
}
AST_LIST_UNLOCK(&misdn_cc_records_db);
/* Wait for the response to the call completion request. */
misdn_cc_response_wait(chan, MISDN_CC_REQUEST_WAIT_MAX, record_id);
AST_LIST_LOCK(&misdn_cc_records_db);
cc_record = misdn_cc_find_by_id(record_id);
if (cc_record) {
if (!cc_record->activated) {
if (cc_record->port < 0) {
/* The network did not tell us that call completion was available. */
error_str = "No port number";
} else if (cc_record->outstanding_message) {
cc_record->outstanding_message = 0;
error_str = misdn_no_response_from_network;
} else if (cc_record->reject_code != FacReject_None) {
error_str = misdn_to_str_reject_code(cc_record->reject_code);
} else if (cc_record->error_code != FacError_None) {
error_str = misdn_to_str_error_code(cc_record->error_code);
} else if (cc_record->ptp) {
if (cc_record->mode.ptp.bc) {
error_str = "Call-completion already requested";
} else {
error_str = "Could not allocate call-completion signaling link";
}
} else {
/* Should never happen. */
error_str = "Unexpected error";
}
/* No need to keep the call completion record. */
if (cc_record->ptp && cc_record->mode.ptp.bc) {
/* Close the call-completion signaling link */
bc = cc_record->mode.ptp.bc;
bc->fac_out.Function = Fac_None;
bc->out_cause = AST_CAUSE_NORMAL_CLEARING;
misdn_lib_send_event(bc, EVENT_RELEASE_COMPLETE);
}
misdn_cc_delete(cc_record);
} else {
error_str = NULL;
}
} else {
error_str = misdn_cc_record_not_found;
}
AST_LIST_UNLOCK(&misdn_cc_records_db);
if (error_str) {
ast_verb(1, "%s(%s) diagnostic '%s' on channel %s\n",
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
misdn_command_name, subcommand->name, error_str, ast_channel_name(chan));
pbx_builtin_setvar_helper(chan, MISDN_ERROR_MSG, error_str);
pbx_builtin_setvar_helper(chan, MISDN_CC_STATUS, "ERROR");
} else {
pbx_builtin_setvar_helper(chan, MISDN_CC_STATUS, "ACTIVATED");
}
return 0;
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief misdn_command(ccbs-request) subcommand handler
*
* \details
* misdn_command(ccbs-request,${MISDN_CC_RECORD_ID},<notify-context>,<user-a-extension>,<priority>)
* Set the dialplan location to notify when User-B is free and User-A is free.
*
* \param chan Asterisk channel to operate upon.
* \param subcommand Arguments for the subcommand
*
* \retval 0 on success.
* \retval -1 on error.
*/
static int misdn_command_ccbs_request(struct ast_channel *chan, struct misdn_command_args *subcommand)
{
static const struct misdn_cc_request request = {
.ptmp = Fac_CCBSRequest,
.ptp = Fac_CCBS_T_Request
};
return misdn_command_cc_request(chan, subcommand, &request);
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief misdn_command(ccnr-request) subcommand handler
*
* \details
* misdn_command(ccnr-request,${MISDN_CC_RECORD_ID},<notify-context>,<user-a-extension>,<priority>)
* Set the dialplan location to notify when User-B is free and User-A is free.
*
* \param chan Asterisk channel to operate upon.
* \param subcommand Arguments for the subcommand
*
* \retval 0 on success.
* \retval -1 on error.
*/
static int misdn_command_ccnr_request(struct ast_channel *chan, struct misdn_command_args *subcommand)
{
static const struct misdn_cc_request request = {
.ptmp = Fac_CCNRRequest,
.ptp = Fac_CCNR_T_Request
};
return misdn_command_cc_request(chan, subcommand, &request);
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
struct misdn_command_table {
/*! \brief subcommand name */
const char *name;
/*! \brief subcommand handler */
int (*func)(struct ast_channel *chan, struct misdn_command_args *subcommand);
/*! \brief TRUE if the subcommand can only be executed on mISDN channels */
int misdn_only;
};
static const struct misdn_command_table misdn_commands[] = {
/* *INDENT-OFF* */
/* subcommand-name subcommand-handler mISDN only */
{ "cc-initialize", misdn_command_cc_initialize, 0 },
{ "cc-deactivate", misdn_command_cc_deactivate, 0 },
{ "cc-a-busy", misdn_command_cc_a_busy, 0 },
{ "cc-b-free", misdn_command_cc_b_free, 0 },
{ "ccbs-request", misdn_command_ccbs_request, 0 },
{ "ccnr-request", misdn_command_ccnr_request, 0 },
/* *INDENT-ON* */
};
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
#if defined(AST_MISDN_ENHANCEMENTS)
/*!
* \internal
* \brief misdn_command() dialplan application.
*
* \param chan Asterisk channel to operate upon.
* \param data Application options string.
*
* \retval 0 on success.
* \retval -1 on error.
*/
static int misdn_command_exec(struct ast_channel *chan, const char *data)
{
char *parse;
unsigned index;
struct misdn_command_args subcommand;
if (ast_strlen_zero((char *) data)) {
ast_log(LOG_ERROR, "%s requires arguments\n", misdn_command_name);
return -1;
}
ast_debug(1, "%s(%s)\n", misdn_command_name, (char *) data);
parse = ast_strdupa(data);
AST_STANDARD_APP_ARGS(subcommand, parse);
if (!subcommand.argc || ast_strlen_zero(subcommand.name)) {
ast_log(LOG_ERROR, "%s requires a subcommand\n", misdn_command_name);
return -1;
}
for (index = 0; index < ARRAY_LEN(misdn_commands); ++index) {
if (strcasecmp(misdn_commands[index].name, subcommand.name) == 0) {
strcpy(subcommand.name, misdn_commands[index].name);
if (misdn_commands[index].misdn_only
&& strcasecmp(ast_channel_tech(chan)->type, misdn_type) != 0) {
ast_log(LOG_WARNING,
"%s(%s) only makes sense with %s channels!\n",
misdn_command_name, subcommand.name, misdn_type);
return -1;
}
return misdn_commands[index].func(chan, &subcommand);
}
}
ast_log(LOG_WARNING, "%s(%s) subcommand is unknown\n", misdn_command_name,
subcommand.name);
return -1;
}
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
static int misdn_facility_exec(struct ast_channel *chan, const char *data)
{
struct chan_list *ch = MISDN_ASTERISK_TECH_PVT(chan);
char *parse;
unsigned max_len;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(facility_type);
AST_APP_ARG(arg)[99];
);
chan_misdn_log(0, 0, "TYPE: %s\n", ast_channel_tech(chan)->type);
if (strcasecmp(ast_channel_tech(chan)->type, misdn_type)) {
ast_log(LOG_WARNING, "misdn_facility only makes sense with %s channels!\n", misdn_type);
return -1;
}
if (ast_strlen_zero((char *) data)) {
ast_log(LOG_WARNING, "misdn_facility requires arguments: facility_type[,<args>]\n");
return -1;
}
parse = ast_strdupa(data);
AST_STANDARD_APP_ARGS(args, parse);
if (ast_strlen_zero(args.facility_type)) {
ast_log(LOG_WARNING, "misdn_facility requires arguments: facility_type[,<args>]\n");
return -1;
}
if (!strcasecmp(args.facility_type, "calldeflect")) {
if (ast_strlen_zero(args.arg[0])) {
ast_log(LOG_WARNING, "Facility: Call Deflection requires an argument: Number\n");
}
#if defined(AST_MISDN_ENHANCEMENTS)
max_len = sizeof(ch->bc->fac_out.u.CallDeflection.Component.Invoke.Deflection.Party.Number) - 1;
if (max_len < strlen(args.arg[0])) {
ast_log(LOG_WARNING,
"Facility: Number argument too long (up to %u digits are allowed). Ignoring.\n",
max_len);
return 0;
}
ch->bc->fac_out.Function = Fac_CallDeflection;
ch->bc->fac_out.u.CallDeflection.InvokeID = ++misdn_invoke_id;
ch->bc->fac_out.u.CallDeflection.ComponentType = FacComponent_Invoke;
ch->bc->fac_out.u.CallDeflection.Component.Invoke.PresentationAllowedToDivertedToUserPresent = 1;
ch->bc->fac_out.u.CallDeflection.Component.Invoke.PresentationAllowedToDivertedToUser = 0;
ch->bc->fac_out.u.CallDeflection.Component.Invoke.Deflection.Party.Type = 0;/* unknown */
ch->bc->fac_out.u.CallDeflection.Component.Invoke.Deflection.Party.LengthOfNumber = strlen(args.arg[0]);
strcpy((char *) ch->bc->fac_out.u.CallDeflection.Component.Invoke.Deflection.Party.Number, args.arg[0]);
ch->bc->fac_out.u.CallDeflection.Component.Invoke.Deflection.Subaddress.Length = 0;
#else /* !defined(AST_MISDN_ENHANCEMENTS) */
max_len = sizeof(ch->bc->fac_out.u.CDeflection.DeflectedToNumber) - 1;
if (max_len < strlen(args.arg[0])) {
ast_log(LOG_WARNING,
"Facility: Number argument too long (up to %u digits are allowed). Ignoring.\n",
max_len);
return 0;
}
ch->bc->fac_out.Function = Fac_CD;
ch->bc->fac_out.u.CDeflection.PresentationAllowed = 0;
//ch->bc->fac_out.u.CDeflection.DeflectedToSubaddress[0] = 0;
strcpy((char *) ch->bc->fac_out.u.CDeflection.DeflectedToNumber, args.arg[0]);
#endif /* !defined(AST_MISDN_ENHANCEMENTS) */
/* Send message */
print_facility(&ch->bc->fac_out, ch->bc);
misdn_lib_send_event(ch->bc, EVENT_FACILITY);
#if defined(AST_MISDN_ENHANCEMENTS)
} else if (!strcasecmp(args.facility_type, "callrerouteing")
|| !strcasecmp(args.facility_type, "callrerouting")) {
if (ast_strlen_zero(args.arg[0])) {
ast_log(LOG_WARNING, "Facility: Call rerouting requires an argument: Number\n");
}
max_len = sizeof(ch->bc->fac_out.u.CallRerouteing.Component.Invoke.CalledAddress.Party.Number) - 1;
if (max_len < strlen(args.arg[0])) {
ast_log(LOG_WARNING,
"Facility: Number argument too long (up to %u digits are allowed). Ignoring.\n",
max_len);
return 0;
}
ch->bc->fac_out.Function = Fac_CallRerouteing;
ch->bc->fac_out.u.CallRerouteing.InvokeID = ++misdn_invoke_id;
ch->bc->fac_out.u.CallRerouteing.ComponentType = FacComponent_Invoke;
ch->bc->fac_out.u.CallRerouteing.Component.Invoke.ReroutingReason = 0;/* unknown */
ch->bc->fac_out.u.CallRerouteing.Component.Invoke.ReroutingCounter = 1;
ch->bc->fac_out.u.CallRerouteing.Component.Invoke.CalledAddress.Party.Type = 0;/* unknown */
ch->bc->fac_out.u.CallRerouteing.Component.Invoke.CalledAddress.Party.LengthOfNumber = strlen(args.arg[0]);
strcpy((char *) ch->bc->fac_out.u.CallRerouteing.Component.Invoke.CalledAddress.Party.Number, args.arg[0]);
ch->bc->fac_out.u.CallRerouteing.Component.Invoke.CalledAddress.Subaddress.Length = 0;
ch->bc->fac_out.u.CallRerouteing.Component.Invoke.CallingPartySubaddress.Length = 0;
/* 0x90 0x90 0xa3 3.1 kHz audio, circuit mode, 64kbit/sec, level1/a-Law */
ch->bc->fac_out.u.CallRerouteing.Component.Invoke.Q931ie.Bc.Length = 3;
ch->bc->fac_out.u.CallRerouteing.Component.Invoke.Q931ie.Bc.Contents[0] = 0x90;
ch->bc->fac_out.u.CallRerouteing.Component.Invoke.Q931ie.Bc.Contents[1] = 0x90;
ch->bc->fac_out.u.CallRerouteing.Component.Invoke.Q931ie.Bc.Contents[2] = 0xa3;
ch->bc->fac_out.u.CallRerouteing.Component.Invoke.Q931ie.Hlc.Length = 0;
ch->bc->fac_out.u.CallRerouteing.Component.Invoke.Q931ie.Llc.Length = 0;
ch->bc->fac_out.u.CallRerouteing.Component.Invoke.Q931ie.UserInfo.Length = 0;
ch->bc->fac_out.u.CallRerouteing.Component.Invoke.LastRerouting.Type = 1;/* presentationRestricted */
ch->bc->fac_out.u.CallRerouteing.Component.Invoke.SubscriptionOption = 0;/* no notification to caller */
/* Send message */
print_facility(&ch->bc->fac_out, ch->bc);
misdn_lib_send_event(ch->bc, EVENT_FACILITY);
#endif /* defined(AST_MISDN_ENHANCEMENTS) */
} else {
chan_misdn_log(1, ch->bc->port, "Unknown Facility: %s\n", args.facility_type);
}
return 0;
}
static int misdn_check_l2l1(struct ast_channel *chan, const char *data)
{
char *parse;
char group[BUFFERSIZE + 1];
char *port_str;
int port = 0;
int timeout;
int dowait = 0;
int port_up;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(grouppar);
AST_APP_ARG(timeout);
);
if (ast_strlen_zero((char *) data)) {
ast_log(LOG_WARNING, "misdn_check_l2l1 Requires arguments\n");
return -1;
}
parse = ast_strdupa(data);
AST_STANDARD_APP_ARGS(args, parse);
if (args.argc != 2) {
ast_log(LOG_WARNING, "Wrong argument count\n");
return 0;
}
/*ast_log(LOG_NOTICE, "Arguments: group/port '%s' timeout '%s'\n", args.grouppar, args.timeout);*/
timeout = atoi(args.timeout);
port_str = args.grouppar;
if (port_str[0] == 'g' && port_str[1] == ':') {
/* We make a group call lets checkout which ports are in my group */
port_str += 2;
ast_copy_string(group, port_str, sizeof(group));
chan_misdn_log(2, 0, "Checking Ports in group: %s\n", group);
for (port = misdn_cfg_get_next_port(port);
port > 0;
port = misdn_cfg_get_next_port(port)) {
char cfg_group[BUFFERSIZE + 1];
chan_misdn_log(2, 0, "trying port %d\n", port);
misdn_cfg_get(port, MISDN_CFG_GROUPNAME, cfg_group, sizeof(cfg_group));
if (!strcasecmp(cfg_group, group)) {
port_up = misdn_lib_port_up(port, 1);
if (!port_up) {
chan_misdn_log(2, 0, " --> port '%d'\n", port);
misdn_lib_get_port_up(port);
dowait = 1;
}
}
}
} else {
port = atoi(port_str);
chan_misdn_log(2, 0, "Checking Port: %d\n", port);
port_up = misdn_lib_port_up(port, 1);
if (!port_up) {
misdn_lib_get_port_up(port);
dowait = 1;
}
}
if (dowait) {
chan_misdn_log(2, 0, "Waiting for '%d' seconds\n", timeout);
ast_safe_sleep(chan, timeout * 1000);
}
return 0;
}
static int misdn_set_opt_exec(struct ast_channel *chan, const char *data)
{
struct chan_list *ch = MISDN_ASTERISK_TECH_PVT(chan);
char *tok;
char *tokb;
char *parse;
int keyidx = 0;
int rxgain = 0;
int txgain = 0;
int change_jitter = 0;
if (strcasecmp(ast_channel_tech(chan)->type, misdn_type)) {
ast_log(LOG_WARNING, "misdn_set_opt makes sense only with %s channels!\n", misdn_type);
return -1;
}
if (ast_strlen_zero((char *) data)) {
ast_log(LOG_WARNING, "misdn_set_opt Requires arguments\n");
return -1;
}
parse = ast_strdupa(data);
for (tok = strtok_r(parse, ":", &tokb);
tok;
tok = strtok_r(NULL, ":", &tokb)) {
int neglect = 0;
if (tok[0] == '!') {
neglect = 1;
tok++;
}
switch(tok[0]) {
case 'd' :
ast_copy_string(ch->bc->display, ++tok, sizeof(ch->bc->display));
chan_misdn_log(1, ch->bc->port, "SETOPT: Display:%s\n", ch->bc->display);
break;
case 'n':
chan_misdn_log(1, ch->bc->port, "SETOPT: No DSP\n");
ch->bc->nodsp = 1;
break;
case 'j':
chan_misdn_log(1, ch->bc->port, "SETOPT: jitter\n");
tok++;
change_jitter = 1;
switch (tok[0]) {
case 'b':
ch->jb_len = atoi(++tok);
chan_misdn_log(1, ch->bc->port, " --> buffer_len:%d\n", ch->jb_len);
break;
case 't' :
ch->jb_upper_threshold = atoi(++tok);
chan_misdn_log(1, ch->bc->port, " --> upper_threshold:%d\n", ch->jb_upper_threshold);
break;
case 'n':
ch->bc->nojitter = 1;
chan_misdn_log(1, ch->bc->port, " --> nojitter\n");
break;
default:
ch->jb_len = 4000;
ch->jb_upper_threshold = 0;
chan_misdn_log(1, ch->bc->port, " --> buffer_len:%d (default)\n", ch->jb_len);
chan_misdn_log(1, ch->bc->port, " --> upper_threshold:%d (default)\n", ch->jb_upper_threshold);
break;
}
break;
case 'v':
tok++;
switch (tok[0]) {
case 'r' :
rxgain = atoi(++tok);
if (rxgain < -8) {
rxgain = -8;
}
if (rxgain > 8) {
rxgain = 8;
}
ch->bc->rxgain = rxgain;
chan_misdn_log(1, ch->bc->port, "SETOPT: Volume:%d\n", rxgain);
break;
case 't':
txgain = atoi(++tok);
if (txgain < -8) {
txgain = -8;
}
if (txgain > 8) {
txgain = 8;
}
ch->bc->txgain = txgain;
chan_misdn_log(1, ch->bc->port, "SETOPT: Volume:%d\n", txgain);
break;
}
break;
case 'c':
keyidx = atoi(++tok);
{
char keys[4096];
char *key = NULL;
char *tmp = keys;
int i;
misdn_cfg_get(0, MISDN_GEN_CRYPT_KEYS, keys, sizeof(keys));
for (i = 0; i < keyidx; i++) {
key = strsep(&tmp, ",");
}
if (key) {
ast_copy_string(ch->bc->crypt_key, key, sizeof(ch->bc->crypt_key));
}
chan_misdn_log(0, ch->bc->port, "SETOPT: crypt with key:%s\n", ch->bc->crypt_key);
break;
}
case 'e':
chan_misdn_log(1, ch->bc->port, "SETOPT: EchoCancel\n");
if (neglect) {
chan_misdn_log(1, ch->bc->port, " --> disabled\n");
#ifdef MISDN_1_2
*ch->bc->pipeline = 0;
#else
ch->bc->ec_enable = 0;
#endif
} else {
#ifdef MISDN_1_2
update_pipeline_config(ch->bc);
#else
ch->bc->ec_enable = 1;
ch->bc->orig = ch->originator;
tok++;
if (*tok) {
ch->bc->ec_deftaps = atoi(tok);
}
#endif
}
break;
case 'h':
chan_misdn_log(1, ch->bc->port, "SETOPT: Digital\n");
if (strlen(tok) > 1 && tok[1] == '1') {
chan_misdn_log(1, ch->bc->port, "SETOPT: HDLC \n");
if (!ch->bc->hdlc) {
ch->bc->hdlc = 1;
}
}
ch->bc->capability = INFO_CAPABILITY_DIGITAL_UNRESTRICTED;
break;
case 's':
chan_misdn_log(1, ch->bc->port, "SETOPT: Send DTMF\n");
ch->bc->send_dtmf = 1;
break;
case 'f':
chan_misdn_log(1, ch->bc->port, "SETOPT: Faxdetect\n");
ch->faxdetect = 1;
misdn_cfg_get(ch->bc->port, MISDN_CFG_FAXDETECT_TIMEOUT, &ch->faxdetect_timeout, sizeof(ch->faxdetect_timeout));
break;
case 'a':
chan_misdn_log(1, ch->bc->port, "SETOPT: AST_DSP (for DTMF)\n");
ch->ast_dsp = 1;
break;
case 'p':
chan_misdn_log(1, ch->bc->port, "SETOPT: callerpres: %s\n", &tok[1]);
/* CRICH: callingpres!!! */
if (strstr(tok, "allowed")) {
ch->bc->presentation = 0;
ch->bc->set_presentation = 1;
} else if (strstr(tok, "restricted")) {
ch->bc->presentation = 1;
ch->bc->set_presentation = 1;
} else if (strstr(tok, "not_screened")) {
chan_misdn_log(0, ch->bc->port, "SETOPT: callerpres: not_screened is deprecated\n");
ch->bc->presentation = 1;
ch->bc->set_presentation = 1;
}
break;
case 'i' :
chan_misdn_log(1, ch->bc->port, "Ignoring dtmf tones, just use them inband\n");
ch->ignore_dtmf = 1;
break;
default:
break;
}
}
if (change_jitter) {
config_jitterbuffer(ch);
}
if (ch->faxdetect || ch->ast_dsp) {
if (!ch->dsp) {
ch->dsp = ast_dsp_new();
}
if (ch->dsp) {
ast_dsp_set_features(ch->dsp, DSP_FEATURE_DIGIT_DETECT | DSP_FEATURE_FAX_DETECT);
}
}
if (ch->ast_dsp) {
chan_misdn_log(1, ch->bc->port, "SETOPT: with AST_DSP we deactivate mISDN_dsp\n");
ch->bc->nodsp = 1;
}
return 0;
}
int chan_misdn_jb_empty(struct misdn_bchannel *bc, char *buf, int len)
{
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
struct chan_list *ch;
int res;
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
ch = find_chan_by_bc(bc);
if (!ch) {
return 0;
}
Merged revisions 294125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 Nov 2010) | 33 lines valgrind reported references to freed memory during a mISDN hangup collision. Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08 17:19:04 +00:00
if (ch->jb) {
res = misdn_jb_empty(ch->jb, buf, len);
} else {
res = 0;
}
chan_list_unref(ch, "Done emptying jb");
return res;
}
/*******************************************************/
/***************** JITTERBUFFER ************************/
/*******************************************************/
/* allocates the jb-structure and initialize the elements*/
struct misdn_jb *misdn_jb_init(int size, int upper_threshold)
{
struct misdn_jb *jb;
jb = ast_calloc(1, sizeof(*jb));
if (!jb) {
chan_misdn_log(-1, 0, "No free Mem for jb\n");
return NULL;
}
jb->size = size;
jb->upper_threshold = upper_threshold;
//jb->wp = 0;
//jb->rp = 0;
//jb->state_full = 0;
//jb->state_empty = 0;
//jb->bytes_wrote = 0;
jb->samples = ast_calloc(size, sizeof(*jb->samples));
if (!jb->samples) {
ast_free(jb);
chan_misdn_log(-1, 0, "No free Mem for jb->samples\n");
return NULL;
}
jb->ok = ast_calloc(size, sizeof(*jb->ok));
if (!jb->ok) {
ast_free(jb->samples);
ast_free(jb);
chan_misdn_log(-1, 0, "No free Mem for jb->ok\n");
return NULL;
}
ast_mutex_init(&jb->mutexjb);
return jb;
}
/* frees the data and destroys the given jitterbuffer struct */
void misdn_jb_destroy(struct misdn_jb *jb)
{
ast_mutex_destroy(&jb->mutexjb);
ast_free(jb->ok);
ast_free(jb->samples);
ast_free(jb);
}
/* fills the jitterbuffer with len data returns < 0 if there was an
error (buffer overflow). */
int misdn_jb_fill(struct misdn_jb *jb, const char *data, int len)
{
int i;
int j;
int rp;
int wp;
if (!jb || ! data) {
return 0;
}
ast_mutex_lock(&jb->mutexjb);
wp = jb->wp;
rp = jb->rp;
for (i = 0; i < len; i++) {
jb->samples[wp] = data[i];
jb->ok[wp] = 1;
wp = (wp != jb->size - 1) ? wp + 1 : 0;
if (wp == jb->rp) {
jb->state_full = 1;
}
}
if (wp >= rp) {
jb->state_buffer = wp - rp;
} else {
jb->state_buffer = jb->size - rp + wp;
}
chan_misdn_log(9, 0, "misdn_jb_fill: written:%d | Buffer status:%d p:%p\n", len, jb->state_buffer, jb);
if (jb->state_full) {
jb->wp = wp;
rp = wp;
for (j = 0; j < jb->upper_threshold; j++) {
rp = (rp != 0) ? rp - 1 : jb->size - 1;
}
jb->rp = rp;
jb->state_full = 0;
jb->state_empty = 1;
ast_mutex_unlock(&jb->mutexjb);
return -1;
}
if (!jb->state_empty) {
jb->bytes_wrote += len;
if (jb->bytes_wrote >= jb->upper_threshold) {
jb->state_empty = 1;
jb->bytes_wrote = 0;
}
}
jb->wp = wp;
ast_mutex_unlock(&jb->mutexjb);
return 0;
}
/* gets len bytes out of the jitterbuffer if available, else only the
available data is returned and the return value indicates the number
of data. */
int misdn_jb_empty(struct misdn_jb *jb, char *data, int len)
{
int i;
int wp;
int rp;
int read = 0;
ast_mutex_lock(&jb->mutexjb);
rp = jb->rp;
wp = jb->wp;
if (jb->state_empty) {
for (i = 0; i < len; i++) {
if (wp == rp) {
jb->rp = rp;
jb->state_empty = 0;
ast_mutex_unlock(&jb->mutexjb);
return read;
} else {
if (jb->ok[rp] == 1) {
data[i] = jb->samples[rp];
jb->ok[rp] = 0;
rp = (rp != jb->size - 1) ? rp + 1 : 0;
read += 1;
}
}
}
if (wp >= rp) {
jb->state_buffer = wp - rp;
} else {
jb->state_buffer = jb->size - rp + wp;
}
chan_misdn_log(9, 0, "misdn_jb_empty: read:%d | Buffer status:%d p:%p\n", len, jb->state_buffer, jb);
jb->rp = rp;
} else {
chan_misdn_log(9, 0, "misdn_jb_empty: Wait...requested:%d p:%p\n", len, jb);
}
ast_mutex_unlock(&jb->mutexjb);
return read;
}
/*******************************************************/
/*************** JITTERBUFFER END *********************/
/*******************************************************/
static void chan_misdn_log(int level, int port, char *tmpl, ...)
{
va_list ap;
char buf[1024];
char port_buf[8];
if (!(0 <= port && port <= max_ports)) {
Merged revisions 374515-374535 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ................ r374515 | rmudgett | 2012-10-04 17:52:36 -0500 (Thu, 04 Oct 2012) | 10 lines chan_misdn: Remove some deadcode * Made setup_bc() static. Patches: patch1_unused-code.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882 ................ r374516 | rmudgett | 2012-10-04 18:01:01 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused bchan states Patches: patch2_unused-states.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374517 | rmudgett | 2012-10-04 18:17:51 -0500 (Thu, 04 Oct 2012) | 16 lines chan_misdn: Remove unnecessary null pointer checks and checks for stack->nt * cleanup_bc() is always called with valid bc (or it would've crashed before). * Value of stack->nt is known in advance at some places. * Rename handle_event() to handle_event_te(), handle_frm() to handle_frm_te(). Patches: patch3_checks.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882 ................ r374518 | rmudgett | 2012-10-04 18:21:59 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Fix spelling in log messages Patches: patch4_spelling.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374519 | rmudgett | 2012-10-04 18:31:59 -0500 (Thu, 04 Oct 2012) | 15 lines chan_misdn: Don't cleanup a bc twice. In handle_frm_te() after calling misdn_lib_send_event(bc, EVENT_RELEASE_COMPLETE) bc is emptied, cleaned and set not in use, although misdn_lib_send_event() already did the same. This is bad. When it's not in use we are not allowed to touch it. * Moved log message in front of the resulting actions and fixed it to match the case. Patches: patch5_bccleanup.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374520 | rmudgett | 2012-10-04 18:43:56 -0500 (Thu, 04 Oct 2012) | 12 lines chan_misdn: Fix memory leaks, bc, chan not cleaned up etc., really bad stuff. * Fix return codes of cb_events() for EVENT_SETUP to use caller's cleanup mechanisms. * Move cl_queue_chan() call after bearer check. Patches: patch6_leaks.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374521 | rmudgett | 2012-10-04 18:48:38 -0500 (Thu, 04 Oct 2012) | 11 lines chan_misdn: We must initialize cause on sending a DISCONNECT. We must initialize cause on sending a DISCONNECT, so it is later correctly indicated to ast_channel in case the answer (RELEASE/RELEASE_COMPLETE) does not include one. Patches: patch7_hangupcause.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374522 | rmudgett | 2012-10-04 19:03:56 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused code for upqueue Patches: patch8_unused-upqueue.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374523 | rmudgett | 2012-10-04 19:11:50 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Improve debugging (port number, messages fixed, dups removed) Patches: patch9_debug.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374533 | rmudgett | 2012-10-05 12:17:18 -0500 (Fri, 05 Oct 2012) | 8 lines chan_misdn: Better debug: we can print_bc_info even if there's no ast leg. Patches: patch10_debug-bc-2.diff (license #6372) patch uploaded by Guenther Kelleter Modified. JIRA ABE-2882 ................ r374534 | rmudgett | 2012-10-05 12:34:10 -0500 (Fri, 05 Oct 2012) | 16 lines chan_misdn: setup_bc() is called too early for an incoming SETUP on TE. This prevents the B channel from being setup for HDLC mode when requested by the bearer capability and config option hdlc=yes. It violates ETS300102 Ch.5.2.3.2: "The user, in any case, must not connect to the channel until a CONNECT ACKNOWLEDGE message has been received." * Call setup_bc() on receipt of CONNECT_ACKNOWLEGDE for PTMP, and on first response to SETUP for PTP. Patches: abe-2881-2.diff (license #6372) patch uploaded by Guenther Kelleter Modified. JIRA ABE-2881 ................ r374535 | rmudgett | 2012-10-05 12:41:05 -0500 (Fri, 05 Oct 2012) | 2 lines chan_misdn: Remove some more deadcode. ................ ........ Merged revisions 374536 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374537 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374538 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374539 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-05 18:42:14 +00:00
ast_log(LOG_WARNING, "chan_misdn_log called with out-of-range port number! (%d)\n", port);
port = 0;
level = -1;
} else if (!(level == -1
|| (misdn_debug_only[port]
? (level == 1 && misdn_debug[port]) || level == misdn_debug[port]
: level <= misdn_debug[port])
|| (level <= misdn_debug[0] && !ast_strlen_zero(global_tracefile)))) {
/*
* We are not going to print anything so lets not
* go to all the work of generating a string.
*/
return;
}
snprintf(port_buf, sizeof(port_buf), "P[%2d] ", port);
va_start(ap, tmpl);
vsnprintf(buf, sizeof(buf), tmpl, ap);
va_end(ap);
if (level == -1) {
ast_log(LOG_WARNING, "%s", buf);
} else if (misdn_debug_only[port]
? (level == 1 && misdn_debug[port]) || level == misdn_debug[port]
: level <= misdn_debug[port]) {
ast_verbose("%s%s", port_buf, buf);
}
if (level <= misdn_debug[0] && !ast_strlen_zero(global_tracefile)) {
char ctimebuf[30];
time_t tm;
char *tmp;
char *p;
FILE *fp;
fp = fopen(global_tracefile, "a+");
if (!fp) {
ast_verbose("Error opening Tracefile: [ %s ] %s\n", global_tracefile, strerror(errno));
return;
}
tm = time(NULL);
tmp = ctime_r(&tm, ctimebuf);
p = strchr(tmp, '\n');
if (p) {
*p = ':';
}
fputs(tmp, fp);
fputs(" ", fp);
fputs(port_buf, fp);
fputs(" ", fp);
fputs(buf, fp);
fclose(fp);
}
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Channel driver for mISDN Support (BRI/PRI)",
.load = load_module,
.unload = unload_module,
.reload = reload,
.load_pri = AST_MODPRI_CHANNEL_DRIVER,
);