asterisk/channels/sig_ss7.c

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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2010 Digium, Inc.
*
* Richard Mudgett <rmudgett@digium.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 SS7 signaling module.
*
* \author Matthew Fredrickson <creslin@digium.com>
* \author Richard Mudgett <rmudgett@digium.com>
*
* See Also:
* \arg \ref AstCREDITS
*/
/*** MODULEINFO
<support_level>core</support_level>
***/
#include "asterisk.h"
#if defined(HAVE_SS7)
#include <signal.h>
#include "asterisk/pbx.h"
#include "asterisk/causes.h"
#include "asterisk/musiconhold.h"
#include "asterisk/cli.h"
#include "asterisk/transcap.h"
#include "sig_ss7.h"
/* ------------------------------------------------------------------- */
static const char *sig_ss7_call_level2str(enum sig_ss7_call_level level)
{
switch (level) {
case SIG_SS7_CALL_LEVEL_IDLE:
return "Idle";
case SIG_SS7_CALL_LEVEL_ALLOCATED:
return "Allocated";
case SIG_SS7_CALL_LEVEL_CONTINUITY:
return "Continuity";
case SIG_SS7_CALL_LEVEL_SETUP:
return "Setup";
case SIG_SS7_CALL_LEVEL_PROCEEDING:
return "Proceeding";
case SIG_SS7_CALL_LEVEL_ALERTING:
return "Alerting";
case SIG_SS7_CALL_LEVEL_CONNECT:
return "Connect";
case SIG_SS7_CALL_LEVEL_GLARE:
return "Glare";
}
return "Unknown";
}
static void sig_ss7_unlock_private(struct sig_ss7_chan *p)
{
if (p->calls->unlock_private) {
p->calls->unlock_private(p->chan_pvt);
}
}
static void sig_ss7_lock_private(struct sig_ss7_chan *p)
{
if (p->calls->lock_private) {
p->calls->lock_private(p->chan_pvt);
}
}
static void sig_ss7_deadlock_avoidance_private(struct sig_ss7_chan *p)
{
if (p->calls->deadlock_avoidance_private) {
p->calls->deadlock_avoidance_private(p->chan_pvt);
} else {
/* Fallback to the old way if callback not present. */
sig_ss7_unlock_private(p);
sched_yield();
sig_ss7_lock_private(p);
}
}
void sig_ss7_set_alarm(struct sig_ss7_chan *p, int in_alarm)
{
p->inalarm = in_alarm;
if (p->calls->set_alarm) {
p->calls->set_alarm(p->chan_pvt, in_alarm);
}
}
static void sig_ss7_set_dialing(struct sig_ss7_chan *p, int is_dialing)
{
if (p->calls->set_dialing) {
p->calls->set_dialing(p->chan_pvt, is_dialing);
}
}
static void sig_ss7_set_digital(struct sig_ss7_chan *p, int is_digital)
{
if (p->calls->set_digital) {
p->calls->set_digital(p->chan_pvt, is_digital);
}
}
static void sig_ss7_set_outgoing(struct sig_ss7_chan *p, int is_outgoing)
{
p->outgoing = is_outgoing;
if (p->calls->set_outgoing) {
p->calls->set_outgoing(p->chan_pvt, is_outgoing);
}
}
static void sig_ss7_set_inservice(struct sig_ss7_chan *p, int is_inservice)
{
if (p->calls->set_inservice) {
p->calls->set_inservice(p->chan_pvt, is_inservice);
}
}
static void sig_ss7_set_locallyblocked(struct sig_ss7_chan *p, int is_blocked)
{
p->locallyblocked = is_blocked;
if (p->calls->set_locallyblocked) {
p->calls->set_locallyblocked(p->chan_pvt, is_blocked);
}
}
static void sig_ss7_set_remotelyblocked(struct sig_ss7_chan *p, int is_blocked)
{
p->remotelyblocked = is_blocked;
if (p->calls->set_remotelyblocked) {
p->calls->set_remotelyblocked(p->chan_pvt, is_blocked);
}
}
/*!
* \internal
* \brief Open the SS7 channel media path.
* \since 1.8.12
*
* \param p Channel private control structure.
*
* \return Nothing
*/
static void sig_ss7_open_media(struct sig_ss7_chan *p)
{
if (p->calls->open_media) {
p->calls->open_media(p->chan_pvt);
}
}
/*!
* \internal
* \brief Set the caller id information in the parent module.
* \since 1.8
*
* \param p sig_ss7 channel structure.
*
* \return Nothing
*/
static void sig_ss7_set_caller_id(struct sig_ss7_chan *p)
{
struct ast_party_caller caller;
if (p->calls->set_callerid) {
ast_party_caller_init(&caller);
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
caller.id.name.str = p->cid_name;
caller.id.name.presentation = p->callingpres;
caller.id.name.valid = 1;
caller.id.number.str = p->cid_num;
caller.id.number.plan = p->cid_ton;
caller.id.number.presentation = p->callingpres;
caller.id.number.valid = 1;
if (!ast_strlen_zero(p->cid_subaddr)) {
caller.id.subaddress.valid = 1;
//caller.id.subaddress.type = 0;/* nsap */
//caller.id.subaddress.odd_even_indicator = 0;
caller.id.subaddress.str = p->cid_subaddr;
}
caller.ani.number.str = p->cid_ani;
//caller.ani.number.plan = p->xxx;
//caller.ani.number.presentation = p->xxx;
caller.ani.number.valid = 1;
caller.ani2 = p->cid_ani2;
p->calls->set_callerid(p->chan_pvt, &caller);
}
}
/*!
* \internal
* \brief Set the Dialed Number Identifier.
* \since 1.8
*
* \param p sig_ss7 channel structure.
* \param dnid Dialed Number Identifier string.
*
* \return Nothing
*/
static void sig_ss7_set_dnid(struct sig_ss7_chan *p, const char *dnid)
{
if (p->calls->set_dnid) {
p->calls->set_dnid(p->chan_pvt, dnid);
}
}
static int sig_ss7_play_tone(struct sig_ss7_chan *p, enum sig_ss7_tone tone)
{
int res;
if (p->calls->play_tone) {
res = p->calls->play_tone(p->chan_pvt, tone);
} else {
res = -1;
}
return res;
}
static int sig_ss7_set_echocanceller(struct sig_ss7_chan *p, int enable)
{
if (p->calls->set_echocanceller) {
return p->calls->set_echocanceller(p->chan_pvt, enable);
}
return -1;
}
static void sig_ss7_loopback(struct sig_ss7_chan *p, int enable)
{
if (p->loopedback != enable) {
p->loopedback = enable;
if (p->calls->set_loopback) {
p->calls->set_loopback(p->chan_pvt, enable);
}
}
}
static struct ast_channel *sig_ss7_new_ast_channel(struct sig_ss7_chan *p, int state, int ulaw, int transfercapability, char *exten, const struct ast_channel *requestor)
{
struct ast_channel *ast;
if (p->calls->new_ast_channel) {
ast = p->calls->new_ast_channel(p->chan_pvt, state, ulaw, exten, requestor);
} else {
return NULL;
}
if (!ast) {
return NULL;
}
if (!p->owner) {
p->owner = ast;
}
p->alreadyhungup = 0;
ast_channel_transfercapability_set(ast, transfercapability);
pbx_builtin_setvar_helper(ast, "TRANSFERCAPABILITY",
ast_transfercapability2str(transfercapability));
if (transfercapability & AST_TRANS_CAP_DIGITAL) {
sig_ss7_set_digital(p, 1);
}
return ast;
}
static void sig_ss7_handle_link_exception(struct sig_ss7_linkset *linkset, int which)
{
if (linkset->calls->handle_link_exception) {
linkset->calls->handle_link_exception(linkset, which);
}
}
/*!
* \internal
* \brief Obtain the sig_ss7 owner channel lock if the owner exists.
* \since 1.8
*
* \param ss7 SS7 linkset control structure.
* \param chanpos Channel position in the span.
*
* \note Assumes the ss7->lock is already obtained.
* \note Assumes the sig_ss7_lock_private(ss7->pvts[chanpos]) is already obtained.
*
* \return Nothing
*/
static void sig_ss7_lock_owner(struct sig_ss7_linkset *ss7, int chanpos)
{
for (;;) {
if (!ss7->pvts[chanpos]->owner) {
/* There is no owner lock to get. */
break;
}
if (!ast_channel_trylock(ss7->pvts[chanpos]->owner)) {
/* We got the lock */
break;
}
/* Avoid deadlock */
sig_ss7_unlock_private(ss7->pvts[chanpos]);
DEADLOCK_AVOIDANCE(&ss7->lock);
sig_ss7_lock_private(ss7->pvts[chanpos]);
}
}
/*!
* \internal
* \brief Queue the given frame onto the owner channel.
* \since 1.8
*
* \param ss7 SS7 linkset control structure.
* \param chanpos Channel position in the span.
* \param frame Frame to queue onto the owner channel.
*
* \note Assumes the ss7->lock is already obtained.
* \note Assumes the sig_ss7_lock_private(ss7->pvts[chanpos]) is already obtained.
*
* \return Nothing
*/
static void sig_ss7_queue_frame(struct sig_ss7_linkset *ss7, int chanpos, struct ast_frame *frame)
{
sig_ss7_lock_owner(ss7, chanpos);
if (ss7->pvts[chanpos]->owner) {
ast_queue_frame(ss7->pvts[chanpos]->owner, frame);
ast_channel_unlock(ss7->pvts[chanpos]->owner);
}
}
/*!
* \internal
* \brief Queue a control frame of the specified subclass onto the owner channel.
* \since 1.8
*
* \param ss7 SS7 linkset control structure.
* \param chanpos Channel position in the span.
* \param subclass Control frame subclass to queue onto the owner channel.
*
* \note Assumes the ss7->lock is already obtained.
* \note Assumes the sig_ss7_lock_private(ss7->pvts[chanpos]) is already obtained.
*
* \return Nothing
*/
static void sig_ss7_queue_control(struct sig_ss7_linkset *ss7, int chanpos, int subclass)
{
struct ast_frame f = {AST_FRAME_CONTROL, };
struct sig_ss7_chan *p = ss7->pvts[chanpos];
if (p->calls->queue_control) {
p->calls->queue_control(p->chan_pvt, subclass);
}
f.subclass.integer = subclass;
sig_ss7_queue_frame(ss7, chanpos, &f);
}
/*!
* \internal
* \brief Find the channel position by CIC/DPC.
*
* \param linkset SS7 linkset control structure.
* \param cic Circuit Identification Code
* \param dpc Destination Point Code
*
* \retval chanpos on success.
* \retval -1 on error.
*/
static int ss7_find_cic(struct sig_ss7_linkset *linkset, int cic, unsigned int dpc)
{
int i;
int winner = -1;
for (i = 0; i < linkset->numchans; i++) {
if (linkset->pvts[i] && (linkset->pvts[i]->dpc == dpc && linkset->pvts[i]->cic == cic)) {
winner = i;
break;
}
}
return winner;
}
/*!
* \internal
* \brief Find the channel position by CIC/DPC and gripe if not found.
*
* \param linkset SS7 linkset control structure.
* \param cic Circuit Identification Code
* \param dpc Destination Point Code
* \param msg_name Message type name that failed.
*
* \retval chanpos on success.
* \retval -1 on error.
*/
static int ss7_find_cic_gripe(struct sig_ss7_linkset *linkset, int cic, unsigned int dpc, const char *msg_name)
{
int chanpos;
chanpos = ss7_find_cic(linkset, cic, dpc);
if (chanpos < 0) {
ast_log(LOG_WARNING, "Linkset %d: SS7 %s requested unconfigured CIC/DPC %d/%d.\n",
linkset->span, msg_name, cic, dpc);
return -1;
}
return chanpos;
}
static void ss7_handle_cqm(struct sig_ss7_linkset *linkset, int startcic, int endcic, unsigned int dpc)
{
unsigned char status[32];
struct sig_ss7_chan *p = NULL;
int i, offset;
for (i = 0; i < linkset->numchans; i++) {
if (linkset->pvts[i] && (linkset->pvts[i]->dpc == dpc && ((linkset->pvts[i]->cic >= startcic) && (linkset->pvts[i]->cic <= endcic)))) {
p = linkset->pvts[i];
offset = p->cic - startcic;
status[offset] = 0;
if (p->locallyblocked)
status[offset] |= (1 << 0) | (1 << 4);
if (p->remotelyblocked)
status[offset] |= (1 << 1) | (1 << 5);
if (p->ss7call) {
if (p->outgoing)
status[offset] |= (1 << 3);
else
status[offset] |= (1 << 2);
} else
status[offset] |= 0x3 << 2;
}
}
if (p)
isup_cqr(linkset->ss7, startcic, endcic, dpc, status);
else
ast_log(LOG_WARNING, "Could not find any equipped circuits within CQM CICs\n");
}
static inline void ss7_hangup_cics(struct sig_ss7_linkset *linkset, int startcic, int endcic, unsigned int dpc)
{
int i;
for (i = 0; i < linkset->numchans; i++) {
if (linkset->pvts[i] && (linkset->pvts[i]->dpc == dpc && ((linkset->pvts[i]->cic >= startcic) && (linkset->pvts[i]->cic <= endcic)))) {
sig_ss7_lock_private(linkset->pvts[i]);
sig_ss7_lock_owner(linkset, i);
if (linkset->pvts[i]->owner) {
ast_softhangup_nolock(linkset->pvts[i]->owner, AST_SOFTHANGUP_DEV);
ast_channel_unlock(linkset->pvts[i]->owner);
}
sig_ss7_unlock_private(linkset->pvts[i]);
}
}
}
static inline void ss7_block_cics(struct sig_ss7_linkset *linkset, int startcic, int endcic, unsigned int dpc, unsigned char state[], int block)
{
int i;
/* XXX the use of state here seems questionable about matching up with the linkset channels */
for (i = 0; i < linkset->numchans; i++) {
if (linkset->pvts[i] && (linkset->pvts[i]->dpc == dpc && ((linkset->pvts[i]->cic >= startcic) && (linkset->pvts[i]->cic <= endcic)))) {
if (state) {
if (state[i])
sig_ss7_set_remotelyblocked(linkset->pvts[i], block);
} else
sig_ss7_set_remotelyblocked(linkset->pvts[i], block);
}
}
}
static void ss7_inservice(struct sig_ss7_linkset *linkset, int startcic, int endcic, unsigned int dpc)
{
int i;
for (i = 0; i < linkset->numchans; i++) {
if (linkset->pvts[i] && (linkset->pvts[i]->dpc == dpc && ((linkset->pvts[i]->cic >= startcic) && (linkset->pvts[i]->cic <= endcic))))
sig_ss7_set_inservice(linkset->pvts[i], 1);
}
}
static void ss7_reset_linkset(struct sig_ss7_linkset *linkset)
{
int i, startcic = -1, endcic, dpc;
if (linkset->numchans <= 0)
return;
startcic = linkset->pvts[0]->cic;
/* DB: CIC's DPC fix */
dpc = linkset->pvts[0]->dpc;
for (i = 0; i < linkset->numchans; i++) {
if (linkset->pvts[i+1] && linkset->pvts[i+1]->dpc == dpc && ((linkset->pvts[i+1]->cic - linkset->pvts[i]->cic) == 1) && (linkset->pvts[i]->cic - startcic < 31)) {
continue;
} else {
endcic = linkset->pvts[i]->cic;
ast_verbose("Resetting CICs %d to %d\n", startcic, endcic);
isup_grs(linkset->ss7, startcic, endcic, dpc);
/* DB: CIC's DPC fix */
if (linkset->pvts[i+1]) {
startcic = linkset->pvts[i+1]->cic;
dpc = linkset->pvts[i+1]->dpc;
}
}
}
}
/* This function is assumed to be called with the private channel lock and linkset lock held */
static void ss7_start_call(struct sig_ss7_chan *p, struct sig_ss7_linkset *linkset)
{
struct ss7 *ss7 = linkset->ss7;
int law;
struct ast_channel *c;
char tmp[256];
if (!(linkset->flags & LINKSET_FLAG_EXPLICITACM)) {
Merged revisions 303771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303771 | rmudgett | 2011-01-25 11:49:20 -0600 (Tue, 25 Jan 2011) | 54 lines Merged revisions 303769 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303769 | rmudgett | 2011-01-25 11:42:42 -0600 (Tue, 25 Jan 2011) | 47 lines Merged revisions 303765 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303765 | rmudgett | 2011-01-25 11:36:50 -0600 (Tue, 25 Jan 2011) | 40 lines Sending out unnecessary PROCEEDING messages breaks overlap dialing. Issue #16789 was a good idea. Unfortunately, it breaks overlap dialing through Asterisk. There is not enough information available at this point to know if dialing is complete. The ast_exists_extension(), ast_matchmore_extension(), and ast_canmatch_extension() calls are not adequate to detect a dial through extension pattern of "_9!". Workaround is to use the dialplan Proceeding() application early in non-dial through extensions. * Effectively revert issue #16789. * Allow outgoing overlap dialing to hear dialtone and other early media. A PROGRESS "inband-information is now available" message is now sent after the SETUP_ACKNOWLEDGE message for non-digital calls. An AST_CONTROL_PROGRESS is now generated for incoming SETUP_ACKNOWLEDGE messages for non-digital calls. * Handling of the AST_CONTROL_CONGESTION in chan_dahdi/sig_pri was inconsistent with the cause codes. * Added better protection from sending out of sequence messages by combining several flags into a single enum value representing call progress level. * Added diagnostic messages for deferred overlap digits handling corner cases. (closes issue #17085) Reported by: shawkris (closes issue #18509) Reported by: wimpy Patches: issue18509_early_media_v1.8_v3.patch uploaded by rmudgett (license 664) Expanded upon issue18509_early_media_v1.8_v3.patch to include analog and SS7 because of backporting requirements. Tested by: wimpy, rmudgett ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-25 17:58:00 +00:00
p->call_level = SIG_SS7_CALL_LEVEL_PROCEEDING;
isup_acm(ss7, p->ss7call);
Merged revisions 303771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303771 | rmudgett | 2011-01-25 11:49:20 -0600 (Tue, 25 Jan 2011) | 54 lines Merged revisions 303769 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303769 | rmudgett | 2011-01-25 11:42:42 -0600 (Tue, 25 Jan 2011) | 47 lines Merged revisions 303765 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303765 | rmudgett | 2011-01-25 11:36:50 -0600 (Tue, 25 Jan 2011) | 40 lines Sending out unnecessary PROCEEDING messages breaks overlap dialing. Issue #16789 was a good idea. Unfortunately, it breaks overlap dialing through Asterisk. There is not enough information available at this point to know if dialing is complete. The ast_exists_extension(), ast_matchmore_extension(), and ast_canmatch_extension() calls are not adequate to detect a dial through extension pattern of "_9!". Workaround is to use the dialplan Proceeding() application early in non-dial through extensions. * Effectively revert issue #16789. * Allow outgoing overlap dialing to hear dialtone and other early media. A PROGRESS "inband-information is now available" message is now sent after the SETUP_ACKNOWLEDGE message for non-digital calls. An AST_CONTROL_PROGRESS is now generated for incoming SETUP_ACKNOWLEDGE messages for non-digital calls. * Handling of the AST_CONTROL_CONGESTION in chan_dahdi/sig_pri was inconsistent with the cause codes. * Added better protection from sending out of sequence messages by combining several flags into a single enum value representing call progress level. * Added diagnostic messages for deferred overlap digits handling corner cases. (closes issue #17085) Reported by: shawkris (closes issue #18509) Reported by: wimpy Patches: issue18509_early_media_v1.8_v3.patch uploaded by rmudgett (license 664) Expanded upon issue18509_early_media_v1.8_v3.patch to include analog and SS7 because of backporting requirements. Tested by: wimpy, rmudgett ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-25 17:58:00 +00:00
} else {
p->call_level = SIG_SS7_CALL_LEVEL_SETUP;
}
if (linkset->type == SS7_ITU) {
law = SIG_SS7_ALAW;
} else {
law = SIG_SS7_ULAW;
}
/*
* Release the SS7 lock while we create the channel so other
* threads can send messages. We must also release the private
* lock to prevent deadlock while creating the channel.
*/
ast_mutex_unlock(&linkset->lock);
sig_ss7_unlock_private(p);
c = sig_ss7_new_ast_channel(p, AST_STATE_RING, law, 0, p->exten, NULL);
if (!c) {
ast_log(LOG_WARNING, "Unable to start PBX on CIC %d\n", p->cic);
ast_mutex_lock(&linkset->lock);
sig_ss7_lock_private(p);
isup_rel(linkset->ss7, p->ss7call, -1);
Merged revisions 303771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303771 | rmudgett | 2011-01-25 11:49:20 -0600 (Tue, 25 Jan 2011) | 54 lines Merged revisions 303769 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303769 | rmudgett | 2011-01-25 11:42:42 -0600 (Tue, 25 Jan 2011) | 47 lines Merged revisions 303765 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303765 | rmudgett | 2011-01-25 11:36:50 -0600 (Tue, 25 Jan 2011) | 40 lines Sending out unnecessary PROCEEDING messages breaks overlap dialing. Issue #16789 was a good idea. Unfortunately, it breaks overlap dialing through Asterisk. There is not enough information available at this point to know if dialing is complete. The ast_exists_extension(), ast_matchmore_extension(), and ast_canmatch_extension() calls are not adequate to detect a dial through extension pattern of "_9!". Workaround is to use the dialplan Proceeding() application early in non-dial through extensions. * Effectively revert issue #16789. * Allow outgoing overlap dialing to hear dialtone and other early media. A PROGRESS "inband-information is now available" message is now sent after the SETUP_ACKNOWLEDGE message for non-digital calls. An AST_CONTROL_PROGRESS is now generated for incoming SETUP_ACKNOWLEDGE messages for non-digital calls. * Handling of the AST_CONTROL_CONGESTION in chan_dahdi/sig_pri was inconsistent with the cause codes. * Added better protection from sending out of sequence messages by combining several flags into a single enum value representing call progress level. * Added diagnostic messages for deferred overlap digits handling corner cases. (closes issue #17085) Reported by: shawkris (closes issue #18509) Reported by: wimpy Patches: issue18509_early_media_v1.8_v3.patch uploaded by rmudgett (license 664) Expanded upon issue18509_early_media_v1.8_v3.patch to include analog and SS7 because of backporting requirements. Tested by: wimpy, rmudgett ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-25 17:58:00 +00:00
p->call_level = SIG_SS7_CALL_LEVEL_IDLE;
p->alreadyhungup = 1;
return;
}
/* Hold the channel and private lock while we setup the channel. */
ast_channel_lock(c);
sig_ss7_lock_private(p);
sig_ss7_set_echocanceller(p, 1);
/*
* It is reasonably safe to set the following
* channel variables while the channel private
* structure is locked. The PBX has not been
* started yet and it is unlikely that any other task
* will do anything with the channel we have just
* created.
*
* We only reference these variables in the context of the ss7_linkset function
* when receiving either and IAM or a COT message.
*/
if (!ast_strlen_zero(p->charge_number)) {
pbx_builtin_setvar_helper(c, "SS7_CHARGE_NUMBER", p->charge_number);
/* Clear this after we set it */
p->charge_number[0] = 0;
}
if (!ast_strlen_zero(p->gen_add_number)) {
pbx_builtin_setvar_helper(c, "SS7_GENERIC_ADDRESS", p->gen_add_number);
/* Clear this after we set it */
p->gen_add_number[0] = 0;
}
if (!ast_strlen_zero(p->jip_number)) {
pbx_builtin_setvar_helper(c, "SS7_JIP", p->jip_number);
/* Clear this after we set it */
p->jip_number[0] = 0;
}
if (!ast_strlen_zero(p->gen_dig_number)) {
pbx_builtin_setvar_helper(c, "SS7_GENERIC_DIGITS", p->gen_dig_number);
/* Clear this after we set it */
p->gen_dig_number[0] = 0;
}
if (!ast_strlen_zero(p->orig_called_num)) {
pbx_builtin_setvar_helper(c, "SS7_ORIG_CALLED_NUM", p->orig_called_num);
/* Clear this after we set it */
p->orig_called_num[0] = 0;
}
snprintf(tmp, sizeof(tmp), "%d", p->gen_dig_type);
pbx_builtin_setvar_helper(c, "SS7_GENERIC_DIGTYPE", tmp);
/* Clear this after we set it */
p->gen_dig_type = 0;
snprintf(tmp, sizeof(tmp), "%d", p->gen_dig_scheme);
pbx_builtin_setvar_helper(c, "SS7_GENERIC_DIGSCHEME", tmp);
/* Clear this after we set it */
p->gen_dig_scheme = 0;
if (!ast_strlen_zero(p->lspi_ident)) {
pbx_builtin_setvar_helper(c, "SS7_LSPI_IDENT", p->lspi_ident);
/* Clear this after we set it */
p->lspi_ident[0] = 0;
}
snprintf(tmp, sizeof(tmp), "%d", p->call_ref_ident);
pbx_builtin_setvar_helper(c, "SS7_CALLREF_IDENT", tmp);
/* Clear this after we set it */
p->call_ref_ident = 0;
snprintf(tmp, sizeof(tmp), "%d", p->call_ref_pc);
pbx_builtin_setvar_helper(c, "SS7_CALLREF_PC", tmp);
/* Clear this after we set it */
p->call_ref_pc = 0;
snprintf(tmp, sizeof(tmp), "%d", p->calling_party_cat);
pbx_builtin_setvar_helper(c, "SS7_CALLING_PARTY_CATEGORY", tmp);
/* Clear this after we set it */
p->calling_party_cat = 0;
if (!ast_strlen_zero(p->redirecting_num)) {
pbx_builtin_setvar_helper(c, "SS7_REDIRECTING_NUMBER", p->redirecting_num);
/* Clear this after we set it */
p->redirecting_num[0] = 0;
}
if (!ast_strlen_zero(p->generic_name)) {
pbx_builtin_setvar_helper(c, "SS7_GENERIC_NAME", p->generic_name);
/* Clear this after we set it */
p->generic_name[0] = 0;
}
sig_ss7_unlock_private(p);
ast_channel_unlock(c);
if (ast_pbx_start(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
ast_log(LOG_WARNING, "Unable to start PBX on %s (CIC %d)\n", ast_channel_name(c), p->cic);
ast_hangup(c);
} else {
ast_verb(3, "Accepting call to '%s' on CIC %d\n", p->exten, p->cic);
}
/* Must return with linkset and private lock. */
ast_mutex_lock(&linkset->lock);
sig_ss7_lock_private(p);
}
static void ss7_apply_plan_to_number(char *buf, size_t size, const struct sig_ss7_linkset *ss7, const char *number, const unsigned nai)
{
if (ast_strlen_zero(number)) { /* make sure a number exists so prefix isn't placed on an empty string */
if (size) {
*buf = '\0';
}
return;
}
switch (nai) {
case SS7_NAI_INTERNATIONAL:
snprintf(buf, size, "%s%s", ss7->internationalprefix, number);
break;
case SS7_NAI_NATIONAL:
snprintf(buf, size, "%s%s", ss7->nationalprefix, number);
break;
case SS7_NAI_SUBSCRIBER:
snprintf(buf, size, "%s%s", ss7->subscriberprefix, number);
break;
case SS7_NAI_UNKNOWN:
snprintf(buf, size, "%s%s", ss7->unknownprefix, number);
break;
default:
snprintf(buf, size, "%s", number);
break;
}
}
static int ss7_pres_scr2cid_pres(char presentation_ind, char screening_ind)
{
return ((presentation_ind & 0x3) << 5) | (screening_ind & 0x3);
}
/* This is a thread per linkset that handles all received events from libss7. */
void *ss7_linkset(void *data)
{
int res, i;
struct timeval *next = NULL, tv;
struct sig_ss7_linkset *linkset = (struct sig_ss7_linkset *) data;
struct ss7 *ss7 = linkset->ss7;
ss7_event *e = NULL;
struct sig_ss7_chan *p;
int chanpos;
struct pollfd pollers[SIG_SS7_NUM_DCHANS];
int nextms = 0;
pthread_setcancelstate(PTHREAD_CANCEL_DISABLE, NULL);
ss7_set_debug(ss7, SIG_SS7_DEBUG_DEFAULT);
ast_mutex_lock(&linkset->lock);
ss7_start(ss7);
ast_mutex_unlock(&linkset->lock);
for (;;) {
ast_mutex_lock(&linkset->lock);
if ((next = ss7_schedule_next(ss7))) {
tv = ast_tvnow();
tv.tv_sec = next->tv_sec - tv.tv_sec;
tv.tv_usec = next->tv_usec - tv.tv_usec;
if (tv.tv_usec < 0) {
tv.tv_usec += 1000000;
tv.tv_sec -= 1;
}
if (tv.tv_sec < 0) {
tv.tv_sec = 0;
tv.tv_usec = 0;
}
nextms = tv.tv_sec * 1000;
nextms += tv.tv_usec / 1000;
}
for (i = 0; i < linkset->numsigchans; i++) {
pollers[i].fd = linkset->fds[i];
pollers[i].events = ss7_pollflags(ss7, linkset->fds[i]);
pollers[i].revents = 0;
}
ast_mutex_unlock(&linkset->lock);
pthread_setcancelstate(PTHREAD_CANCEL_ENABLE, NULL);
pthread_testcancel();
res = poll(pollers, linkset->numsigchans, nextms);
pthread_testcancel();
pthread_setcancelstate(PTHREAD_CANCEL_DISABLE, NULL);
if ((res < 0) && (errno != EINTR)) {
ast_log(LOG_ERROR, "poll(%s)\n", strerror(errno));
} else if (!res) {
ast_mutex_lock(&linkset->lock);
ss7_schedule_run(ss7);
ast_mutex_unlock(&linkset->lock);
continue;
}
ast_mutex_lock(&linkset->lock);
for (i = 0; i < linkset->numsigchans; i++) {
if (pollers[i].revents & POLLPRI) {
sig_ss7_handle_link_exception(linkset, i);
}
if (pollers[i].revents & POLLIN) {
res = ss7_read(ss7, pollers[i].fd);
}
if (pollers[i].revents & POLLOUT) {
res = ss7_write(ss7, pollers[i].fd);
if (res < 0) {
ast_debug(1, "Error in write %s\n", strerror(errno));
}
}
}
while ((e = ss7_check_event(ss7))) {
if (linkset->debug) {
ast_verbose("Linkset %d: Processing event: %s\n",
linkset->span, ss7_event2str(e->e));
}
switch (e->e) {
case SS7_EVENT_UP:
if (linkset->state != LINKSET_STATE_UP) {
ast_verbose("--- SS7 Up ---\n");
ss7_reset_linkset(linkset);
}
linkset->state = LINKSET_STATE_UP;
break;
case SS7_EVENT_DOWN:
ast_verbose("--- SS7 Down ---\n");
linkset->state = LINKSET_STATE_DOWN;
for (i = 0; i < linkset->numchans; i++) {
p = linkset->pvts[i];
if (p) {
sig_ss7_set_alarm(p, 1);
}
}
break;
case MTP2_LINK_UP:
ast_verbose("MTP2 link up (SLC %d)\n", e->gen.data);
break;
case MTP2_LINK_DOWN:
ast_log(LOG_WARNING, "MTP2 link down (SLC %d)\n", e->gen.data);
break;
case ISUP_EVENT_CPG:
chanpos = ss7_find_cic_gripe(linkset, e->cpg.cic, e->cpg.opc, "CPG");
if (chanpos < 0) {
break;
}
p = linkset->pvts[chanpos];
sig_ss7_lock_private(p);
switch (e->cpg.event) {
case CPG_EVENT_ALERTING:
Merged revisions 303771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303771 | rmudgett | 2011-01-25 11:49:20 -0600 (Tue, 25 Jan 2011) | 54 lines Merged revisions 303769 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303769 | rmudgett | 2011-01-25 11:42:42 -0600 (Tue, 25 Jan 2011) | 47 lines Merged revisions 303765 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303765 | rmudgett | 2011-01-25 11:36:50 -0600 (Tue, 25 Jan 2011) | 40 lines Sending out unnecessary PROCEEDING messages breaks overlap dialing. Issue #16789 was a good idea. Unfortunately, it breaks overlap dialing through Asterisk. There is not enough information available at this point to know if dialing is complete. The ast_exists_extension(), ast_matchmore_extension(), and ast_canmatch_extension() calls are not adequate to detect a dial through extension pattern of "_9!". Workaround is to use the dialplan Proceeding() application early in non-dial through extensions. * Effectively revert issue #16789. * Allow outgoing overlap dialing to hear dialtone and other early media. A PROGRESS "inband-information is now available" message is now sent after the SETUP_ACKNOWLEDGE message for non-digital calls. An AST_CONTROL_PROGRESS is now generated for incoming SETUP_ACKNOWLEDGE messages for non-digital calls. * Handling of the AST_CONTROL_CONGESTION in chan_dahdi/sig_pri was inconsistent with the cause codes. * Added better protection from sending out of sequence messages by combining several flags into a single enum value representing call progress level. * Added diagnostic messages for deferred overlap digits handling corner cases. (closes issue #17085) Reported by: shawkris (closes issue #18509) Reported by: wimpy Patches: issue18509_early_media_v1.8_v3.patch uploaded by rmudgett (license 664) Expanded upon issue18509_early_media_v1.8_v3.patch to include analog and SS7 because of backporting requirements. Tested by: wimpy, rmudgett ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-25 17:58:00 +00:00
if (p->call_level < SIG_SS7_CALL_LEVEL_ALERTING) {
p->call_level = SIG_SS7_CALL_LEVEL_ALERTING;
}
sig_ss7_lock_owner(linkset, chanpos);
if (p->owner) {
ast_setstate(p->owner, AST_STATE_RINGING);
ast_channel_unlock(p->owner);
}
sig_ss7_queue_control(linkset, chanpos, AST_CONTROL_RINGING);
break;
case CPG_EVENT_PROGRESS:
case CPG_EVENT_INBANDINFO:
{
ast_debug(1, "Queuing frame PROGRESS on CIC %d\n", p->cic);
sig_ss7_queue_control(linkset, chanpos, AST_CONTROL_PROGRESS);
p->progress = 1;
sig_ss7_set_dialing(p, 0);
sig_ss7_open_media(p);
}
break;
default:
ast_debug(1, "Do not handle CPG with event type 0x%x\n", e->cpg.event);
break;
}
sig_ss7_unlock_private(p);
break;
case ISUP_EVENT_RSC:
ast_verbose("Resetting CIC %d\n", e->rsc.cic);
chanpos = ss7_find_cic_gripe(linkset, e->rsc.cic, e->rsc.opc, "RSC");
if (chanpos < 0) {
break;
}
p = linkset->pvts[chanpos];
sig_ss7_lock_private(p);
sig_ss7_set_inservice(p, 1);
sig_ss7_set_remotelyblocked(p, 0);
isup_set_call_dpc(e->rsc.call, p->dpc);
sig_ss7_lock_owner(linkset, chanpos);
p->ss7call = NULL;
if (p->owner) {
ast_softhangup_nolock(p->owner, AST_SOFTHANGUP_DEV);
ast_channel_unlock(p->owner);
}
sig_ss7_unlock_private(p);
isup_rlc(ss7, e->rsc.call);
break;
case ISUP_EVENT_GRS:
ast_debug(1, "Got Reset for CICs %d to %d: Acknowledging\n", e->grs.startcic, e->grs.endcic);
chanpos = ss7_find_cic_gripe(linkset, e->grs.startcic, e->grs.opc, "GRS");
if (chanpos < 0) {
break;
}
p = linkset->pvts[chanpos];
isup_gra(ss7, e->grs.startcic, e->grs.endcic, e->grs.opc);
ss7_block_cics(linkset, e->grs.startcic, e->grs.endcic, e->grs.opc, NULL, 0);
ss7_hangup_cics(linkset, e->grs.startcic, e->grs.endcic, e->grs.opc);
break;
case ISUP_EVENT_CQM:
ast_debug(1, "Got Circuit group query message from CICs %d to %d\n", e->cqm.startcic, e->cqm.endcic);
ss7_handle_cqm(linkset, e->cqm.startcic, e->cqm.endcic, e->cqm.opc);
break;
case ISUP_EVENT_GRA:
ast_verbose("Got reset acknowledgement from CIC %d to %d.\n", e->gra.startcic, e->gra.endcic);
ss7_inservice(linkset, e->gra.startcic, e->gra.endcic, e->gra.opc);
ss7_block_cics(linkset, e->gra.startcic, e->gra.endcic, e->gra.opc, e->gra.status, 1);
break;
case ISUP_EVENT_IAM:
ast_debug(1, "Got IAM for CIC %d and called number %s, calling number %s\n", e->iam.cic, e->iam.called_party_num, e->iam.calling_party_num);
chanpos = ss7_find_cic_gripe(linkset, e->iam.cic, e->iam.opc, "IAM");
if (chanpos < 0) {
isup_rel(ss7, e->iam.call, -1);
break;
}
p = linkset->pvts[chanpos];
sig_ss7_lock_private(p);
sig_ss7_lock_owner(linkset, chanpos);
if (p->call_level != SIG_SS7_CALL_LEVEL_IDLE) {
/*
* Detected glare/dual-seizure
*
* Always abort both calls since we can't implement the dual
* seizure procedures due to our channel assignment architecture
* and the fact that we cannot tell libss7 to discard its call
* structure to ignore the incoming IAM.
*/
ast_debug(1,
"Linkset %d: SS7 IAM glare on CIC/DPC %d/%d. Dropping both calls.\n",
linkset->span, e->iam.cic, e->iam.opc);
if (p->call_level == SIG_SS7_CALL_LEVEL_ALLOCATED) {
/*
* We have not sent our IAM yet and we never will at this point.
*/
p->alreadyhungup = 1;
isup_rel(ss7, e->iam.call, -1);
}
p->call_level = SIG_SS7_CALL_LEVEL_GLARE;
if (p->owner) {
ast_channel_hangupcause_set(p->owner, AST_CAUSE_NORMAL_CLEARING);
ast_softhangup_nolock(p->owner, AST_SOFTHANGUP_DEV);
ast_channel_unlock(p->owner);
}
sig_ss7_unlock_private(p);
break;
}
/*
* The channel should not have an owner at this point since we
* are in the process of creating an owner for it.
*/
ast_assert(!p->owner);
/* Mark channel as in use so no outgoing call will steal it. */
p->call_level = SIG_SS7_CALL_LEVEL_ALLOCATED;
p->ss7call = e->iam.call;
isup_set_call_dpc(p->ss7call, p->dpc);
if ((p->use_callerid) && (!ast_strlen_zero(e->iam.calling_party_num))) {
ss7_apply_plan_to_number(p->cid_num, sizeof(p->cid_num), linkset, e->iam.calling_party_num, e->iam.calling_nai);
p->callingpres = ss7_pres_scr2cid_pres(e->iam.presentation_ind, e->iam.screening_ind);
} else
p->cid_num[0] = 0;
/* Set DNID */
if (!ast_strlen_zero(e->iam.called_party_num)) {
ss7_apply_plan_to_number(p->exten, sizeof(p->exten), linkset,
e->iam.called_party_num, e->iam.called_nai);
} else {
p->exten[0] = '\0';
}
sig_ss7_set_dnid(p, p->exten);
if (p->immediate) {
p->exten[0] = 's';
p->exten[1] = '\0';
} else if (!ast_strlen_zero(e->iam.called_party_num)) {
char *st;
ss7_apply_plan_to_number(p->exten, sizeof(p->exten), linkset, e->iam.called_party_num, e->iam.called_nai);
st = strchr(p->exten, '#');
if (st) {
*st = '\0';
}
} else {
p->exten[0] = '\0';
}
p->cid_ani[0] = '\0';
if ((p->use_callerid) && (!ast_strlen_zero(e->iam.generic_name)))
ast_copy_string(p->cid_name, e->iam.generic_name, sizeof(p->cid_name));
else
p->cid_name[0] = '\0';
p->cid_ani2 = e->iam.oli_ani2;
p->cid_ton = 0;
ast_copy_string(p->charge_number, e->iam.charge_number, sizeof(p->charge_number));
ast_copy_string(p->gen_add_number, e->iam.gen_add_number, sizeof(p->gen_add_number));
p->gen_add_type = e->iam.gen_add_type;
p->gen_add_nai = e->iam.gen_add_nai;
p->gen_add_pres_ind = e->iam.gen_add_pres_ind;
p->gen_add_num_plan = e->iam.gen_add_num_plan;
ast_copy_string(p->gen_dig_number, e->iam.gen_dig_number, sizeof(p->gen_dig_number));
p->gen_dig_type = e->iam.gen_dig_type;
p->gen_dig_scheme = e->iam.gen_dig_scheme;
ast_copy_string(p->jip_number, e->iam.jip_number, sizeof(p->jip_number));
ast_copy_string(p->orig_called_num, e->iam.orig_called_num, sizeof(p->orig_called_num));
ast_copy_string(p->redirecting_num, e->iam.redirecting_num, sizeof(p->redirecting_num));
ast_copy_string(p->generic_name, e->iam.generic_name, sizeof(p->generic_name));
p->calling_party_cat = e->iam.calling_party_cat;
sig_ss7_set_caller_id(p);
if (ast_exists_extension(NULL, p->context, p->exten, 1, p->cid_num)) {
if (e->iam.cot_check_required) {
p->call_level = SIG_SS7_CALL_LEVEL_CONTINUITY;
sig_ss7_loopback(p, 1);
} else {
ss7_start_call(p, linkset);
}
} else {
ast_debug(1, "Call on CIC for unconfigured extension %s\n", p->exten);
p->alreadyhungup = 1;
isup_rel(ss7, e->iam.call, AST_CAUSE_UNALLOCATED);
}
sig_ss7_unlock_private(p);
break;
case ISUP_EVENT_COT:
chanpos = ss7_find_cic_gripe(linkset, e->cot.cic, e->cot.opc, "COT");
if (chanpos < 0) {
isup_rel(ss7, e->cot.call, -1);
break;
}
p = linkset->pvts[chanpos];
sig_ss7_lock_private(p);
if (p->loopedback) {
sig_ss7_loopback(p, 0);
ss7_start_call(p, linkset);
}
sig_ss7_unlock_private(p);
break;
case ISUP_EVENT_CCR:
ast_debug(1, "Got CCR request on CIC %d\n", e->ccr.cic);
chanpos = ss7_find_cic_gripe(linkset, e->ccr.cic, e->ccr.opc, "CCR");
if (chanpos < 0) {
break;
}
p = linkset->pvts[chanpos];
sig_ss7_lock_private(p);
sig_ss7_loopback(p, 1);
sig_ss7_unlock_private(p);
isup_lpa(linkset->ss7, e->ccr.cic, p->dpc);
break;
case ISUP_EVENT_CVT:
ast_debug(1, "Got CVT request on CIC %d\n", e->cvt.cic);
chanpos = ss7_find_cic_gripe(linkset, e->cvt.cic, e->cvt.opc, "CVT");
if (chanpos < 0) {
break;
}
p = linkset->pvts[chanpos];
sig_ss7_lock_private(p);
sig_ss7_loopback(p, 1);
sig_ss7_unlock_private(p);
isup_cvr(linkset->ss7, e->cvt.cic, p->dpc);
break;
case ISUP_EVENT_REL:
chanpos = ss7_find_cic_gripe(linkset, e->rel.cic, e->rel.opc, "REL");
if (chanpos < 0) {
/* Continue hanging up the call anyway. */
isup_rlc(ss7, e->rel.call);
break;
}
p = linkset->pvts[chanpos];
sig_ss7_lock_private(p);
sig_ss7_lock_owner(linkset, chanpos);
if (p->owner) {
ast_channel_hangupcause_set(p->owner, e->rel.cause);
ast_softhangup_nolock(p->owner, AST_SOFTHANGUP_DEV);
ast_channel_unlock(p->owner);
}
/* End the loopback if we have one */
sig_ss7_loopback(p, 0);
isup_rlc(ss7, e->rel.call);
p->ss7call = NULL;
sig_ss7_unlock_private(p);
break;
case ISUP_EVENT_ACM:
chanpos = ss7_find_cic_gripe(linkset, e->acm.cic, e->acm.opc, "ACM");
if (chanpos < 0) {
isup_rel(ss7, e->acm.call, -1);
break;
}
{
p = linkset->pvts[chanpos];
ast_debug(1, "Queueing frame from SS7_EVENT_ACM on CIC %d\n", p->cic);
if (e->acm.call_ref_ident > 0) {
p->rlt = 1; /* Setting it but not using it here*/
}
sig_ss7_lock_private(p);
sig_ss7_queue_control(linkset, chanpos, AST_CONTROL_PROCEEDING);
Merged revisions 303771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303771 | rmudgett | 2011-01-25 11:49:20 -0600 (Tue, 25 Jan 2011) | 54 lines Merged revisions 303769 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303769 | rmudgett | 2011-01-25 11:42:42 -0600 (Tue, 25 Jan 2011) | 47 lines Merged revisions 303765 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303765 | rmudgett | 2011-01-25 11:36:50 -0600 (Tue, 25 Jan 2011) | 40 lines Sending out unnecessary PROCEEDING messages breaks overlap dialing. Issue #16789 was a good idea. Unfortunately, it breaks overlap dialing through Asterisk. There is not enough information available at this point to know if dialing is complete. The ast_exists_extension(), ast_matchmore_extension(), and ast_canmatch_extension() calls are not adequate to detect a dial through extension pattern of "_9!". Workaround is to use the dialplan Proceeding() application early in non-dial through extensions. * Effectively revert issue #16789. * Allow outgoing overlap dialing to hear dialtone and other early media. A PROGRESS "inband-information is now available" message is now sent after the SETUP_ACKNOWLEDGE message for non-digital calls. An AST_CONTROL_PROGRESS is now generated for incoming SETUP_ACKNOWLEDGE messages for non-digital calls. * Handling of the AST_CONTROL_CONGESTION in chan_dahdi/sig_pri was inconsistent with the cause codes. * Added better protection from sending out of sequence messages by combining several flags into a single enum value representing call progress level. * Added diagnostic messages for deferred overlap digits handling corner cases. (closes issue #17085) Reported by: shawkris (closes issue #18509) Reported by: wimpy Patches: issue18509_early_media_v1.8_v3.patch uploaded by rmudgett (license 664) Expanded upon issue18509_early_media_v1.8_v3.patch to include analog and SS7 because of backporting requirements. Tested by: wimpy, rmudgett ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-25 17:58:00 +00:00
if (p->call_level < SIG_SS7_CALL_LEVEL_PROCEEDING) {
p->call_level = SIG_SS7_CALL_LEVEL_PROCEEDING;
}
sig_ss7_set_dialing(p, 0);
/* Send alerting if subscriber is free */
if (e->acm.called_party_status_ind == 1) {
Merged revisions 303771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303771 | rmudgett | 2011-01-25 11:49:20 -0600 (Tue, 25 Jan 2011) | 54 lines Merged revisions 303769 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303769 | rmudgett | 2011-01-25 11:42:42 -0600 (Tue, 25 Jan 2011) | 47 lines Merged revisions 303765 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303765 | rmudgett | 2011-01-25 11:36:50 -0600 (Tue, 25 Jan 2011) | 40 lines Sending out unnecessary PROCEEDING messages breaks overlap dialing. Issue #16789 was a good idea. Unfortunately, it breaks overlap dialing through Asterisk. There is not enough information available at this point to know if dialing is complete. The ast_exists_extension(), ast_matchmore_extension(), and ast_canmatch_extension() calls are not adequate to detect a dial through extension pattern of "_9!". Workaround is to use the dialplan Proceeding() application early in non-dial through extensions. * Effectively revert issue #16789. * Allow outgoing overlap dialing to hear dialtone and other early media. A PROGRESS "inband-information is now available" message is now sent after the SETUP_ACKNOWLEDGE message for non-digital calls. An AST_CONTROL_PROGRESS is now generated for incoming SETUP_ACKNOWLEDGE messages for non-digital calls. * Handling of the AST_CONTROL_CONGESTION in chan_dahdi/sig_pri was inconsistent with the cause codes. * Added better protection from sending out of sequence messages by combining several flags into a single enum value representing call progress level. * Added diagnostic messages for deferred overlap digits handling corner cases. (closes issue #17085) Reported by: shawkris (closes issue #18509) Reported by: wimpy Patches: issue18509_early_media_v1.8_v3.patch uploaded by rmudgett (license 664) Expanded upon issue18509_early_media_v1.8_v3.patch to include analog and SS7 because of backporting requirements. Tested by: wimpy, rmudgett ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-25 17:58:00 +00:00
if (p->call_level < SIG_SS7_CALL_LEVEL_ALERTING) {
p->call_level = SIG_SS7_CALL_LEVEL_ALERTING;
}
sig_ss7_lock_owner(linkset, chanpos);
if (p->owner) {
ast_setstate(p->owner, AST_STATE_RINGING);
ast_channel_unlock(p->owner);
}
sig_ss7_queue_control(linkset, chanpos, AST_CONTROL_RINGING);
}
sig_ss7_unlock_private(p);
}
break;
case ISUP_EVENT_CGB:
chanpos = ss7_find_cic_gripe(linkset, e->cgb.startcic, e->cgb.opc, "CGB");
if (chanpos < 0) {
break;
}
p = linkset->pvts[chanpos];
ss7_block_cics(linkset, e->cgb.startcic, e->cgb.endcic, e->cgb.opc, e->cgb.status, 1);
isup_cgba(linkset->ss7, e->cgb.startcic, e->cgb.endcic, e->cgb.opc, e->cgb.status, e->cgb.type);
break;
case ISUP_EVENT_CGU:
chanpos = ss7_find_cic_gripe(linkset, e->cgu.startcic, e->cgu.opc, "CGU");
if (chanpos < 0) {
break;
}
p = linkset->pvts[chanpos];
ss7_block_cics(linkset, e->cgu.startcic, e->cgu.endcic, e->cgu.opc, e->cgu.status, 0);
isup_cgua(linkset->ss7, e->cgu.startcic, e->cgu.endcic, e->cgu.opc, e->cgu.status, e->cgu.type);
break;
case ISUP_EVENT_UCIC:
chanpos = ss7_find_cic_gripe(linkset, e->ucic.cic, e->ucic.opc, "UCIC");
if (chanpos < 0) {
break;
}
p = linkset->pvts[chanpos];
ast_debug(1, "Unequiped Circuit Id Code on CIC %d\n", e->ucic.cic);
sig_ss7_lock_private(p);
sig_ss7_set_remotelyblocked(p, 1);
sig_ss7_set_inservice(p, 0);
sig_ss7_unlock_private(p);/* doesn't require a SS7 acknowledgement */
break;
case ISUP_EVENT_BLO:
chanpos = ss7_find_cic_gripe(linkset, e->blo.cic, e->blo.opc, "BLO");
if (chanpos < 0) {
break;
}
p = linkset->pvts[chanpos];
ast_debug(1, "Blocking CIC %d\n", e->blo.cic);
sig_ss7_lock_private(p);
sig_ss7_set_remotelyblocked(p, 1);
sig_ss7_unlock_private(p);
isup_bla(linkset->ss7, e->blo.cic, p->dpc);
break;
case ISUP_EVENT_BLA:
chanpos = ss7_find_cic_gripe(linkset, e->bla.cic, e->bla.opc, "BLA");
if (chanpos < 0) {
break;
}
ast_debug(1, "Blocking CIC %d\n", e->bla.cic);
p = linkset->pvts[chanpos];
sig_ss7_lock_private(p);
sig_ss7_set_locallyblocked(p, 1);
sig_ss7_unlock_private(p);
break;
case ISUP_EVENT_UBL:
chanpos = ss7_find_cic_gripe(linkset, e->ubl.cic, e->ubl.opc, "UBL");
if (chanpos < 0) {
break;
}
p = linkset->pvts[chanpos];
ast_debug(1, "Unblocking CIC %d\n", e->ubl.cic);
sig_ss7_lock_private(p);
sig_ss7_set_remotelyblocked(p, 0);
sig_ss7_unlock_private(p);
isup_uba(linkset->ss7, e->ubl.cic, p->dpc);
break;
case ISUP_EVENT_UBA:
chanpos = ss7_find_cic_gripe(linkset, e->uba.cic, e->uba.opc, "UBA");
if (chanpos < 0) {
break;
}
p = linkset->pvts[chanpos];
ast_debug(1, "Unblocking CIC %d\n", e->uba.cic);
sig_ss7_lock_private(p);
sig_ss7_set_locallyblocked(p, 0);
sig_ss7_unlock_private(p);
break;
case ISUP_EVENT_CON:
case ISUP_EVENT_ANM:
if (e->e == ISUP_EVENT_CON) {
chanpos = ss7_find_cic_gripe(linkset, e->con.cic, e->con.opc, "CON");
if (chanpos < 0) {
isup_rel(ss7, e->con.call, -1);
break;
}
} else {
chanpos = ss7_find_cic_gripe(linkset, e->anm.cic, e->anm.opc, "ANM");
if (chanpos < 0) {
isup_rel(ss7, e->anm.call, -1);
break;
}
}
{
p = linkset->pvts[chanpos];
sig_ss7_lock_private(p);
Merged revisions 303771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303771 | rmudgett | 2011-01-25 11:49:20 -0600 (Tue, 25 Jan 2011) | 54 lines Merged revisions 303769 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303769 | rmudgett | 2011-01-25 11:42:42 -0600 (Tue, 25 Jan 2011) | 47 lines Merged revisions 303765 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303765 | rmudgett | 2011-01-25 11:36:50 -0600 (Tue, 25 Jan 2011) | 40 lines Sending out unnecessary PROCEEDING messages breaks overlap dialing. Issue #16789 was a good idea. Unfortunately, it breaks overlap dialing through Asterisk. There is not enough information available at this point to know if dialing is complete. The ast_exists_extension(), ast_matchmore_extension(), and ast_canmatch_extension() calls are not adequate to detect a dial through extension pattern of "_9!". Workaround is to use the dialplan Proceeding() application early in non-dial through extensions. * Effectively revert issue #16789. * Allow outgoing overlap dialing to hear dialtone and other early media. A PROGRESS "inband-information is now available" message is now sent after the SETUP_ACKNOWLEDGE message for non-digital calls. An AST_CONTROL_PROGRESS is now generated for incoming SETUP_ACKNOWLEDGE messages for non-digital calls. * Handling of the AST_CONTROL_CONGESTION in chan_dahdi/sig_pri was inconsistent with the cause codes. * Added better protection from sending out of sequence messages by combining several flags into a single enum value representing call progress level. * Added diagnostic messages for deferred overlap digits handling corner cases. (closes issue #17085) Reported by: shawkris (closes issue #18509) Reported by: wimpy Patches: issue18509_early_media_v1.8_v3.patch uploaded by rmudgett (license 664) Expanded upon issue18509_early_media_v1.8_v3.patch to include analog and SS7 because of backporting requirements. Tested by: wimpy, rmudgett ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-25 17:58:00 +00:00
if (p->call_level < SIG_SS7_CALL_LEVEL_CONNECT) {
p->call_level = SIG_SS7_CALL_LEVEL_CONNECT;
}
sig_ss7_queue_control(linkset, chanpos, AST_CONTROL_ANSWER);
sig_ss7_set_dialing(p, 0);
sig_ss7_open_media(p);
sig_ss7_set_echocanceller(p, 1);
sig_ss7_unlock_private(p);
}
break;
case ISUP_EVENT_RLC:
/* XXX Call ptr should be passed up from libss7! */
chanpos = ss7_find_cic_gripe(linkset, e->rlc.cic, e->rlc.opc, "RLC");
if (chanpos < 0) {
break;
}
{
p = linkset->pvts[chanpos];
sig_ss7_lock_private(p);
if (p->alreadyhungup) {
if (!p->owner) {
p->call_level = SIG_SS7_CALL_LEVEL_IDLE;
}
p->ss7call = NULL;
}
sig_ss7_unlock_private(p);
}
break;
case ISUP_EVENT_FAA:
/*!
* \todo The handling of the SS7 FAA message is not good and I
* don't know enough to handle it correctly.
*/
chanpos = ss7_find_cic_gripe(linkset, e->faa.cic, e->faa.opc, "FAA");
if (chanpos < 0) {
isup_rel(linkset->ss7, e->faa.call, -1);
break;
}
{
/* XXX FAR and FAA used for something dealing with transfers? */
p = linkset->pvts[chanpos];
ast_debug(1, "FAA received on CIC %d\n", e->faa.cic);
sig_ss7_lock_private(p);
if (p->alreadyhungup){
if (!p->owner) {
p->call_level = SIG_SS7_CALL_LEVEL_IDLE;
}
/* XXX We seem to be leaking the isup call structure here. */
p->ss7call = NULL;
ast_log(LOG_NOTICE, "Received FAA and we haven't sent FAR. Ignoring.\n");
}
sig_ss7_unlock_private(p);
}
break;
default:
ast_debug(1, "Unknown event %s\n", ss7_event2str(e->e));
break;
}
}
ast_mutex_unlock(&linkset->lock);
}
return 0;
}
static inline void ss7_rel(struct sig_ss7_linkset *ss7)
{
ast_mutex_unlock(&ss7->lock);
}
static void ss7_grab(struct sig_ss7_chan *pvt, struct sig_ss7_linkset *ss7)
{
/* Grab the lock first */
while (ast_mutex_trylock(&ss7->lock)) {
/* Avoid deadlock */
sig_ss7_deadlock_avoidance_private(pvt);
}
/* Then break the poll */
if (ss7->master != AST_PTHREADT_NULL) {
pthread_kill(ss7->master, SIGURG);
}
}
/*!
* \brief Notify the SS7 layer that the link is in alarm.
* \since 1.8
*
* \param linkset Controlling linkset for the channel.
* \param which Link index of the signaling channel.
*
* \return Nothing
*/
void sig_ss7_link_alarm(struct sig_ss7_linkset *linkset, int which)
{
linkset->linkstate[which] |= (LINKSTATE_DOWN | LINKSTATE_INALARM);
linkset->linkstate[which] &= ~LINKSTATE_UP;
ss7_link_alarm(linkset->ss7, linkset->fds[which]);
}
/*!
* \brief Notify the SS7 layer that the link is no longer in alarm.
* \since 1.8
*
* \param linkset Controlling linkset for the channel.
* \param which Link index of the signaling channel.
*
* \return Nothing
*/
void sig_ss7_link_noalarm(struct sig_ss7_linkset *linkset, int which)
{
linkset->linkstate[which] &= ~(LINKSTATE_INALARM | LINKSTATE_DOWN);
linkset->linkstate[which] |= LINKSTATE_STARTING;
ss7_link_noalarm(linkset->ss7, linkset->fds[which]);
}
/*!
* \brief Setup and add a SS7 link channel.
* \since 1.8
*
* \param linkset Controlling linkset for the channel.
* \param which Link index of the signaling channel.
* \param ss7type Switch type of the linkset
* \param transport Signaling transport of channel.
* \param inalarm Non-zero if the channel is in alarm.
* \param networkindicator User configuration parameter.
* \param pointcode User configuration parameter.
* \param adjpointcode User configuration parameter.
*
* \retval 0 on success.
* \retval -1 on error.
*/
int sig_ss7_add_sigchan(struct sig_ss7_linkset *linkset, int which, int ss7type, int transport, int inalarm, int networkindicator, int pointcode, int adjpointcode)
{
if (!linkset->ss7) {
linkset->type = ss7type;
linkset->ss7 = ss7_new(ss7type);
if (!linkset->ss7) {
ast_log(LOG_ERROR, "Can't create new SS7!\n");
return -1;
}
}
ss7_set_network_ind(linkset->ss7, networkindicator);
ss7_set_pc(linkset->ss7, pointcode);
if (ss7_add_link(linkset->ss7, transport, linkset->fds[which])) {
ast_log(LOG_WARNING, "Could not add SS7 link!\n");
}
if (inalarm) {
linkset->linkstate[which] = LINKSTATE_DOWN | LINKSTATE_INALARM;
ss7_link_alarm(linkset->ss7, linkset->fds[which]);
} else {
linkset->linkstate[which] = LINKSTATE_DOWN;
ss7_link_noalarm(linkset->ss7, linkset->fds[which]);
}
ss7_set_adjpc(linkset->ss7, linkset->fds[which], adjpointcode);
return 0;
}
/*!
* \internal
* \brief Determine if a private channel structure is available.
*
* \param pvt Channel to determine if available.
*
* \return TRUE if the channel is available.
*/
static int sig_ss7_is_chan_available(struct sig_ss7_chan *pvt)
{
if (!pvt->inalarm && !pvt->owner && !pvt->ss7call
&& pvt->call_level == SIG_SS7_CALL_LEVEL_IDLE
&& !pvt->locallyblocked && !pvt->remotelyblocked) {
return 1;
}
return 0;
}
/*!
* \brief Determine if the specified channel is available for an outgoing call.
* \since 1.8
*
* \param p Signaling private structure pointer.
*
* \retval TRUE if the channel is available.
*/
int sig_ss7_available(struct sig_ss7_chan *p)
{
int available;
if (!p->ss7) {
/* Something is wrong here. A SS7 channel without the ss7 pointer? */
return 0;
}
/* Only have to deal with the linkset lock. */
ast_mutex_lock(&p->ss7->lock);
available = sig_ss7_is_chan_available(p);
if (available) {
p->call_level = SIG_SS7_CALL_LEVEL_ALLOCATED;
}
ast_mutex_unlock(&p->ss7->lock);
return available;
}
static unsigned char cid_pres2ss7pres(int cid_pres)
{
return (cid_pres >> 5) & 0x03;
}
static unsigned char cid_pres2ss7screen(int cid_pres)
{
return cid_pres & 0x03;
}
/*!
* \brief Dial out using the specified SS7 channel.
* \since 1.8
*
* \param p Signaling private structure pointer.
* \param ast Asterisk channel structure pointer.
* \param rdest Dialstring.
*
* \retval 0 on success.
* \retval -1 on error.
*/
int sig_ss7_call(struct sig_ss7_chan *p, struct ast_channel *ast, const char *rdest)
{
char ss7_called_nai;
int called_nai_strip;
char ss7_calling_nai;
int calling_nai_strip;
const char *charge_str = NULL;
const char *gen_address = NULL;
const char *gen_digits = NULL;
const char *gen_dig_type = NULL;
const char *gen_dig_scheme = NULL;
const char *gen_name = NULL;
const char *jip_digits = NULL;
const char *lspi_ident = NULL;
const char *rlt_flag = NULL;
const char *call_ref_id = NULL;
const char *call_ref_pc = NULL;
const char *send_far = NULL;
char *c;
char *l;
char dest[256];
ast_copy_string(dest, rdest, sizeof(dest));
c = strchr(dest, '/');
if (c) {
c++;
} else {
c = "";
}
if (strlen(c) < p->stripmsd) {
ast_log(LOG_WARNING, "Number '%s' is shorter than stripmsd (%d)\n", c, p->stripmsd);
return -1;
}
if (!p->hidecallerid) {
l = ast_channel_connected(ast)->id.number.valid ? ast_channel_connected(ast)->id.number.str : NULL;
} else {
l = NULL;
}
ss7_grab(p, p->ss7);
if (p->call_level != SIG_SS7_CALL_LEVEL_ALLOCATED) {
/* Call collision before sending IAM. Abort call. */
ss7_rel(p->ss7);
return -1;
}
p->ss7call = isup_new_call(p->ss7->ss7);
if (!p->ss7call) {
ss7_rel(p->ss7);
ast_log(LOG_ERROR, "Unable to allocate new SS7 call!\n");
return -1;
}
called_nai_strip = 0;
ss7_called_nai = p->ss7->called_nai;
if (ss7_called_nai == SS7_NAI_DYNAMIC) { /* compute dynamically */
if (strncmp(c + p->stripmsd, p->ss7->internationalprefix, strlen(p->ss7->internationalprefix)) == 0) {
called_nai_strip = strlen(p->ss7->internationalprefix);
ss7_called_nai = SS7_NAI_INTERNATIONAL;
} else if (strncmp(c + p->stripmsd, p->ss7->nationalprefix, strlen(p->ss7->nationalprefix)) == 0) {
called_nai_strip = strlen(p->ss7->nationalprefix);
ss7_called_nai = SS7_NAI_NATIONAL;
} else {
ss7_called_nai = SS7_NAI_SUBSCRIBER;
}
}
isup_set_called(p->ss7call, c + p->stripmsd + called_nai_strip, ss7_called_nai, p->ss7->ss7);
calling_nai_strip = 0;
ss7_calling_nai = p->ss7->calling_nai;
if ((l != NULL) && (ss7_calling_nai == SS7_NAI_DYNAMIC)) { /* compute dynamically */
if (strncmp(l, p->ss7->internationalprefix, strlen(p->ss7->internationalprefix)) == 0) {
calling_nai_strip = strlen(p->ss7->internationalprefix);
ss7_calling_nai = SS7_NAI_INTERNATIONAL;
} else if (strncmp(l, p->ss7->nationalprefix, strlen(p->ss7->nationalprefix)) == 0) {
calling_nai_strip = strlen(p->ss7->nationalprefix);
ss7_calling_nai = SS7_NAI_NATIONAL;
} else {
ss7_calling_nai = SS7_NAI_SUBSCRIBER;
}
}
isup_set_calling(p->ss7call, l ? (l + calling_nai_strip) : NULL, ss7_calling_nai,
p->use_callingpres ? cid_pres2ss7pres(ast_channel_connected(ast)->id.number.presentation) : (l ? SS7_PRESENTATION_ALLOWED : SS7_PRESENTATION_RESTRICTED),
p->use_callingpres ? cid_pres2ss7screen(ast_channel_connected(ast)->id.number.presentation) : SS7_SCREENING_USER_PROVIDED);
isup_set_oli(p->ss7call, ast_channel_connected(ast)->ani2);
isup_init_call(p->ss7->ss7, p->ss7call, p->cic, p->dpc);
/* Set the charge number if it is set */
charge_str = pbx_builtin_getvar_helper(ast, "SS7_CHARGE_NUMBER");
if (charge_str)
isup_set_charge(p->ss7call, charge_str, SS7_ANI_CALLING_PARTY_SUB_NUMBER, 0x10);
gen_address = pbx_builtin_getvar_helper(ast, "SS7_GENERIC_ADDRESS");
if (gen_address)
isup_set_gen_address(p->ss7call, gen_address, p->gen_add_nai,p->gen_add_pres_ind, p->gen_add_num_plan,p->gen_add_type); /* need to add some types here for NAI,PRES,TYPE */
gen_digits = pbx_builtin_getvar_helper(ast, "SS7_GENERIC_DIGITS");
gen_dig_type = pbx_builtin_getvar_helper(ast, "SS7_GENERIC_DIGTYPE");
gen_dig_scheme = pbx_builtin_getvar_helper(ast, "SS7_GENERIC_DIGSCHEME");
if (gen_digits)
isup_set_gen_digits(p->ss7call, gen_digits, atoi(gen_dig_type), atoi(gen_dig_scheme));
gen_name = pbx_builtin_getvar_helper(ast, "SS7_GENERIC_NAME");
if (gen_name)
isup_set_generic_name(p->ss7call, gen_name, GEN_NAME_TYPE_CALLING_NAME, GEN_NAME_AVAIL_AVAILABLE, GEN_NAME_PRES_ALLOWED);
jip_digits = pbx_builtin_getvar_helper(ast, "SS7_JIP");
if (jip_digits)
isup_set_jip_digits(p->ss7call, jip_digits);
lspi_ident = pbx_builtin_getvar_helper(ast, "SS7_LSPI_IDENT");
if (lspi_ident)
isup_set_lspi(p->ss7call, lspi_ident, 0x18, 0x7, 0x00);
rlt_flag = pbx_builtin_getvar_helper(ast, "SS7_RLT_ON");
if ((rlt_flag) && ((strncmp("NO", rlt_flag, strlen(rlt_flag))) != 0 )) {
isup_set_lspi(p->ss7call, rlt_flag, 0x18, 0x7, 0x00); /* Setting for Nortel DMS-250/500 */
}
call_ref_id = pbx_builtin_getvar_helper(ast, "SS7_CALLREF_IDENT");
call_ref_pc = pbx_builtin_getvar_helper(ast, "SS7_CALLREF_PC");
if (call_ref_id && call_ref_pc) {
isup_set_callref(p->ss7call, atoi(call_ref_id),
call_ref_pc ? atoi(call_ref_pc) : 0);
}
send_far = pbx_builtin_getvar_helper(ast, "SS7_SEND_FAR");
if ((send_far) && ((strncmp("NO", send_far, strlen(send_far))) != 0 ))
(isup_far(p->ss7->ss7, p->ss7call));
Merged revisions 303771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303771 | rmudgett | 2011-01-25 11:49:20 -0600 (Tue, 25 Jan 2011) | 54 lines Merged revisions 303769 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303769 | rmudgett | 2011-01-25 11:42:42 -0600 (Tue, 25 Jan 2011) | 47 lines Merged revisions 303765 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303765 | rmudgett | 2011-01-25 11:36:50 -0600 (Tue, 25 Jan 2011) | 40 lines Sending out unnecessary PROCEEDING messages breaks overlap dialing. Issue #16789 was a good idea. Unfortunately, it breaks overlap dialing through Asterisk. There is not enough information available at this point to know if dialing is complete. The ast_exists_extension(), ast_matchmore_extension(), and ast_canmatch_extension() calls are not adequate to detect a dial through extension pattern of "_9!". Workaround is to use the dialplan Proceeding() application early in non-dial through extensions. * Effectively revert issue #16789. * Allow outgoing overlap dialing to hear dialtone and other early media. A PROGRESS "inband-information is now available" message is now sent after the SETUP_ACKNOWLEDGE message for non-digital calls. An AST_CONTROL_PROGRESS is now generated for incoming SETUP_ACKNOWLEDGE messages for non-digital calls. * Handling of the AST_CONTROL_CONGESTION in chan_dahdi/sig_pri was inconsistent with the cause codes. * Added better protection from sending out of sequence messages by combining several flags into a single enum value representing call progress level. * Added diagnostic messages for deferred overlap digits handling corner cases. (closes issue #17085) Reported by: shawkris (closes issue #18509) Reported by: wimpy Patches: issue18509_early_media_v1.8_v3.patch uploaded by rmudgett (license 664) Expanded upon issue18509_early_media_v1.8_v3.patch to include analog and SS7 because of backporting requirements. Tested by: wimpy, rmudgett ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-25 17:58:00 +00:00
p->call_level = SIG_SS7_CALL_LEVEL_SETUP;
isup_iam(p->ss7->ss7, p->ss7call);
sig_ss7_set_dialing(p, 1);
ast_setstate(ast, AST_STATE_DIALING);
ss7_rel(p->ss7);
return 0;
}
/*!
* \brief SS7 hangup channel.
* \since 1.8
*
* \param p Signaling private structure pointer.
* \param ast Asterisk channel structure pointer.
*
* \retval 0 on success.
* \retval -1 on error.
*/
int sig_ss7_hangup(struct sig_ss7_chan *p, struct ast_channel *ast)
{
int res = 0;
if (!ast_channel_tech_pvt(ast)) {
ast_log(LOG_WARNING, "Asked to hangup channel not connected\n");
return 0;
}
p->owner = NULL;
sig_ss7_set_dialing(p, 0);
sig_ss7_set_outgoing(p, 0);
p->progress = 0;
p->rlt = 0;
p->exten[0] = '\0';
/* Perform low level hangup if no owner left */
ss7_grab(p, p->ss7);
p->call_level = SIG_SS7_CALL_LEVEL_IDLE;
if (p->ss7call) {
if (!p->alreadyhungup) {
const char *cause = pbx_builtin_getvar_helper(ast,"SS7_CAUSE");
int icause = ast_channel_hangupcause(ast) ? ast_channel_hangupcause(ast) : -1;
if (cause) {
if (atoi(cause)) {
icause = atoi(cause);
}
}
isup_rel(p->ss7->ss7, p->ss7call, icause);
p->alreadyhungup = 1;
}
}
ss7_rel(p->ss7);
return res;
}
/*!
* \brief SS7 answer channel.
* \since 1.8
*
* \param p Signaling private structure pointer.
* \param ast Asterisk channel structure pointer.
*
* \retval 0 on success.
* \retval -1 on error.
*/
int sig_ss7_answer(struct sig_ss7_chan *p, struct ast_channel *ast)
{
int res;
ss7_grab(p, p->ss7);
if (p->call_level < SIG_SS7_CALL_LEVEL_CONNECT) {
p->call_level = SIG_SS7_CALL_LEVEL_CONNECT;
}
sig_ss7_open_media(p);
res = isup_anm(p->ss7->ss7, p->ss7call);
ss7_rel(p->ss7);
return res;
}
/*!
* \brief Fix up a channel: If a channel is consumed, this is called. Basically update any ->owner links.
* \since 1.8
*
* \param oldchan Old channel pointer to replace.
* \param newchan New channel pointer to set.
* \param pchan Signaling private structure pointer.
*
* \return Nothing
*/
void sig_ss7_fixup(struct ast_channel *oldchan, struct ast_channel *newchan, struct sig_ss7_chan *pchan)
{
if (pchan->owner == oldchan) {
pchan->owner = newchan;
}
}
/*!
* \brief SS7 answer channel.
* \since 1.8
*
* \param p Signaling private structure pointer.
* \param chan Asterisk channel structure pointer.
* \param condition AST control frame subtype.
* \param data AST control frame payload contents.
* \param datalen Length of payload contents.
*
* \retval 0 on success.
* \retval -1 on error or indication condition not handled.
*/
int sig_ss7_indicate(struct sig_ss7_chan *p, struct ast_channel *chan, int condition, const void *data, size_t datalen)
{
int res = -1;
switch (condition) {
case AST_CONTROL_BUSY:
if (p->call_level < SIG_SS7_CALL_LEVEL_CONNECT) {
ast_channel_hangupcause_set(chan, AST_CAUSE_USER_BUSY);
ast_softhangup_nolock(chan, AST_SOFTHANGUP_DEV);
res = 0;
break;
}
res = sig_ss7_play_tone(p, SIG_SS7_TONE_BUSY);
break;
case AST_CONTROL_RINGING:
ss7_grab(p, p->ss7);
Merged revisions 303771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303771 | rmudgett | 2011-01-25 11:49:20 -0600 (Tue, 25 Jan 2011) | 54 lines Merged revisions 303769 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303769 | rmudgett | 2011-01-25 11:42:42 -0600 (Tue, 25 Jan 2011) | 47 lines Merged revisions 303765 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303765 | rmudgett | 2011-01-25 11:36:50 -0600 (Tue, 25 Jan 2011) | 40 lines Sending out unnecessary PROCEEDING messages breaks overlap dialing. Issue #16789 was a good idea. Unfortunately, it breaks overlap dialing through Asterisk. There is not enough information available at this point to know if dialing is complete. The ast_exists_extension(), ast_matchmore_extension(), and ast_canmatch_extension() calls are not adequate to detect a dial through extension pattern of "_9!". Workaround is to use the dialplan Proceeding() application early in non-dial through extensions. * Effectively revert issue #16789. * Allow outgoing overlap dialing to hear dialtone and other early media. A PROGRESS "inband-information is now available" message is now sent after the SETUP_ACKNOWLEDGE message for non-digital calls. An AST_CONTROL_PROGRESS is now generated for incoming SETUP_ACKNOWLEDGE messages for non-digital calls. * Handling of the AST_CONTROL_CONGESTION in chan_dahdi/sig_pri was inconsistent with the cause codes. * Added better protection from sending out of sequence messages by combining several flags into a single enum value representing call progress level. * Added diagnostic messages for deferred overlap digits handling corner cases. (closes issue #17085) Reported by: shawkris (closes issue #18509) Reported by: wimpy Patches: issue18509_early_media_v1.8_v3.patch uploaded by rmudgett (license 664) Expanded upon issue18509_early_media_v1.8_v3.patch to include analog and SS7 because of backporting requirements. Tested by: wimpy, rmudgett ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-25 17:58:00 +00:00
if (p->call_level < SIG_SS7_CALL_LEVEL_ALERTING && !p->outgoing) {
p->call_level = SIG_SS7_CALL_LEVEL_ALERTING;
if ((isup_far(p->ss7->ss7, p->ss7call)) != -1) {
p->rlt = 1;
}
/* No need to send CPG if call will be RELEASE */
if (p->rlt != 1) {
isup_cpg(p->ss7->ss7, p->ss7call, CPG_EVENT_ALERTING);
}
}
ss7_rel(p->ss7);
res = sig_ss7_play_tone(p, SIG_SS7_TONE_RINGTONE);
if (ast_channel_state(chan) != AST_STATE_UP && ast_channel_state(chan) != AST_STATE_RING) {
ast_setstate(chan, AST_STATE_RINGING);
}
break;
case AST_CONTROL_PROCEEDING:
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,"Received AST_CONTROL_PROCEEDING on %s\n",ast_channel_name(chan));
ss7_grab(p, p->ss7);
/* This IF sends the FAR for an answered ALEG call */
if (ast_channel_state(chan) == AST_STATE_UP && (p->rlt != 1)){
if ((isup_far(p->ss7->ss7, p->ss7call)) != -1) {
p->rlt = 1;
}
}
Merged revisions 303771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303771 | rmudgett | 2011-01-25 11:49:20 -0600 (Tue, 25 Jan 2011) | 54 lines Merged revisions 303769 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303769 | rmudgett | 2011-01-25 11:42:42 -0600 (Tue, 25 Jan 2011) | 47 lines Merged revisions 303765 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303765 | rmudgett | 2011-01-25 11:36:50 -0600 (Tue, 25 Jan 2011) | 40 lines Sending out unnecessary PROCEEDING messages breaks overlap dialing. Issue #16789 was a good idea. Unfortunately, it breaks overlap dialing through Asterisk. There is not enough information available at this point to know if dialing is complete. The ast_exists_extension(), ast_matchmore_extension(), and ast_canmatch_extension() calls are not adequate to detect a dial through extension pattern of "_9!". Workaround is to use the dialplan Proceeding() application early in non-dial through extensions. * Effectively revert issue #16789. * Allow outgoing overlap dialing to hear dialtone and other early media. A PROGRESS "inband-information is now available" message is now sent after the SETUP_ACKNOWLEDGE message for non-digital calls. An AST_CONTROL_PROGRESS is now generated for incoming SETUP_ACKNOWLEDGE messages for non-digital calls. * Handling of the AST_CONTROL_CONGESTION in chan_dahdi/sig_pri was inconsistent with the cause codes. * Added better protection from sending out of sequence messages by combining several flags into a single enum value representing call progress level. * Added diagnostic messages for deferred overlap digits handling corner cases. (closes issue #17085) Reported by: shawkris (closes issue #18509) Reported by: wimpy Patches: issue18509_early_media_v1.8_v3.patch uploaded by rmudgett (license 664) Expanded upon issue18509_early_media_v1.8_v3.patch to include analog and SS7 because of backporting requirements. Tested by: wimpy, rmudgett ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-25 17:58:00 +00:00
if (p->call_level < SIG_SS7_CALL_LEVEL_PROCEEDING && !p->outgoing) {
p->call_level = SIG_SS7_CALL_LEVEL_PROCEEDING;
isup_acm(p->ss7->ss7, p->ss7call);
}
ss7_rel(p->ss7);
/* don't continue in ast_indicate */
res = 0;
break;
case AST_CONTROL_PROGRESS:
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,"Received AST_CONTROL_PROGRESS on %s\n",ast_channel_name(chan));
ss7_grab(p, p->ss7);
Merged revisions 303771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303771 | rmudgett | 2011-01-25 11:49:20 -0600 (Tue, 25 Jan 2011) | 54 lines Merged revisions 303769 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303769 | rmudgett | 2011-01-25 11:42:42 -0600 (Tue, 25 Jan 2011) | 47 lines Merged revisions 303765 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303765 | rmudgett | 2011-01-25 11:36:50 -0600 (Tue, 25 Jan 2011) | 40 lines Sending out unnecessary PROCEEDING messages breaks overlap dialing. Issue #16789 was a good idea. Unfortunately, it breaks overlap dialing through Asterisk. There is not enough information available at this point to know if dialing is complete. The ast_exists_extension(), ast_matchmore_extension(), and ast_canmatch_extension() calls are not adequate to detect a dial through extension pattern of "_9!". Workaround is to use the dialplan Proceeding() application early in non-dial through extensions. * Effectively revert issue #16789. * Allow outgoing overlap dialing to hear dialtone and other early media. A PROGRESS "inband-information is now available" message is now sent after the SETUP_ACKNOWLEDGE message for non-digital calls. An AST_CONTROL_PROGRESS is now generated for incoming SETUP_ACKNOWLEDGE messages for non-digital calls. * Handling of the AST_CONTROL_CONGESTION in chan_dahdi/sig_pri was inconsistent with the cause codes. * Added better protection from sending out of sequence messages by combining several flags into a single enum value representing call progress level. * Added diagnostic messages for deferred overlap digits handling corner cases. (closes issue #17085) Reported by: shawkris (closes issue #18509) Reported by: wimpy Patches: issue18509_early_media_v1.8_v3.patch uploaded by rmudgett (license 664) Expanded upon issue18509_early_media_v1.8_v3.patch to include analog and SS7 because of backporting requirements. Tested by: wimpy, rmudgett ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-25 17:58:00 +00:00
if (!p->progress && p->call_level < SIG_SS7_CALL_LEVEL_ALERTING && !p->outgoing) {
p->progress = 1;/* No need to send inband-information progress again. */
isup_cpg(p->ss7->ss7, p->ss7call, CPG_EVENT_INBANDINFO);
ss7_rel(p->ss7);
/* enable echo canceler here on SS7 calls */
sig_ss7_set_echocanceller(p, 1);
} else {
ss7_rel(p->ss7);
}
/* don't continue in ast_indicate */
res = 0;
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:
if (p->call_level < SIG_SS7_CALL_LEVEL_CONNECT) {
ast_channel_hangupcause_set(chan, AST_CAUSE_INVALID_NUMBER_FORMAT);
ast_softhangup_nolock(chan, AST_SOFTHANGUP_DEV);
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
res = 0;
break;
}
/* Wait for DTMF digits to complete the dialed number. */
res = 0;
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
break;
case AST_CONTROL_CONGESTION:
if (p->call_level < SIG_SS7_CALL_LEVEL_CONNECT) {
ast_channel_hangupcause_set(chan, AST_CAUSE_CONGESTION);
ast_softhangup_nolock(chan, AST_SOFTHANGUP_DEV);
res = 0;
break;
}
res = sig_ss7_play_tone(p, SIG_SS7_TONE_CONGESTION);
break;
case AST_CONTROL_HOLD:
ast_moh_start(chan, data, p->mohinterpret);
break;
case AST_CONTROL_UNHOLD:
ast_moh_stop(chan);
break;
case AST_CONTROL_SRCUPDATE:
res = 0;
break;
case -1:
res = sig_ss7_play_tone(p, -1);
break;
}
return res;
}
/*!
* \brief SS7 channel request.
* \since 1.8
*
* \param p Signaling private structure pointer.
* \param law Companding law preferred
* \param requestor Asterisk channel requesting a channel to dial (Can be NULL)
* \param transfercapability
*
* \retval ast_channel on success.
* \retval NULL on error.
*/
struct ast_channel *sig_ss7_request(struct sig_ss7_chan *p, enum sig_ss7_law law, const struct ast_channel *requestor, int transfercapability)
{
struct ast_channel *ast;
sig_ss7_set_outgoing(p, 1);
ast = sig_ss7_new_ast_channel(p, AST_STATE_RESERVED, law, transfercapability, p->exten, requestor);
if (!ast) {
sig_ss7_set_outgoing(p, 0);
/* Release the allocated channel. Only have to deal with the linkset lock. */
ast_mutex_lock(&p->ss7->lock);
p->call_level = SIG_SS7_CALL_LEVEL_IDLE;
ast_mutex_unlock(&p->ss7->lock);
}
return ast;
}
/*!
* \brief Delete the sig_ss7 private channel structure.
* \since 1.8
*
* \param doomed sig_ss7 private channel structure to delete.
*
* \return Nothing
*/
void sig_ss7_chan_delete(struct sig_ss7_chan *doomed)
{
ast_free(doomed);
}
#define SIG_SS7_SC_HEADER "%-4s %4s %-4s %-3s %-3s %-10s %-4s %s\n"
#define SIG_SS7_SC_LINE "%4d %4d %-4s %-3s %-3s %-10s %-4s %s"
void sig_ss7_cli_show_channels_header(int fd)
{
ast_cli(fd, SIG_SS7_SC_HEADER, "link", "", "Chan", "Lcl", "Rem", "Call", "SS7", "Channel");
ast_cli(fd, SIG_SS7_SC_HEADER, "set", "Chan", "Idle", "Blk", "Blk", "Level", "Call", "Name");
}
void sig_ss7_cli_show_channels(int fd, struct sig_ss7_linkset *linkset)
{
char line[256];
int idx;
struct sig_ss7_chan *pvt;
ast_mutex_lock(&linkset->lock);
for (idx = 0; idx < linkset->numchans; ++idx) {
if (!linkset->pvts[idx]) {
continue;
}
pvt = linkset->pvts[idx];
sig_ss7_lock_private(pvt);
sig_ss7_lock_owner(linkset, idx);
snprintf(line, sizeof(line), SIG_SS7_SC_LINE,
linkset->span,
pvt->channel,
sig_ss7_is_chan_available(pvt) ? "Yes" : "No",
pvt->locallyblocked ? "Yes" : "No",
pvt->remotelyblocked ? "Yes" : "No",
sig_ss7_call_level2str(pvt->call_level),
pvt->ss7call ? "Yes" : "No",
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
pvt->owner ? ast_channel_name(pvt->owner) : "");
if (pvt->owner) {
ast_channel_unlock(pvt->owner);
}
sig_ss7_unlock_private(pvt);
ast_mutex_unlock(&linkset->lock);
ast_cli(fd, "%s\n", line);
ast_mutex_lock(&linkset->lock);
}
ast_mutex_unlock(&linkset->lock);
}
/*!
* \brief Create a new sig_ss7 private channel structure.
* \since 1.8
*
* \param pvt_data Upper layer private data structure.
* \param callback Callbacks to the upper layer.
* \param ss7 Controlling linkset for the channel.
*
* \retval sig_ss7_chan on success.
* \retval NULL on error.
*/
struct sig_ss7_chan *sig_ss7_chan_new(void *pvt_data, struct sig_ss7_callback *callback, struct sig_ss7_linkset *ss7)
{
struct sig_ss7_chan *pvt;
pvt = ast_calloc(1, sizeof(*pvt));
if (!pvt) {
return pvt;
}
pvt->calls = callback;
pvt->chan_pvt = pvt_data;
pvt->ss7 = ss7;
return pvt;
}
/*!
* \brief Initialize the SS7 linkset control.
* \since 1.8
*
* \param ss7 SS7 linkset control structure.
*
* \return Nothing
*/
void sig_ss7_init_linkset(struct sig_ss7_linkset *ss7)
{
int idx;
memset(ss7, 0, sizeof(*ss7));
ast_mutex_init(&ss7->lock);
ss7->master = AST_PTHREADT_NULL;
for (idx = 0; idx < ARRAY_LEN(ss7->fds); ++idx) {
ss7->fds[idx] = -1;
}
}
/* ------------------------------------------------------------------- */
#endif /* defined(HAVE_SS7) */
/* end sig_ss7.c */