asterisk/res/res_pjsip_caller_id.c

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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2013, Digium, Inc.
*
* Mark Michelson <mmichelson@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.
*/
/*** MODULEINFO
<depend>pjproject</depend>
<depend>res_pjsip</depend>
<depend>res_pjsip_session</depend>
<support_level>core</support_level>
***/
#include "asterisk.h"
#include <pjsip.h>
#include <pjsip_ua.h>
#include "asterisk/res_pjsip.h"
#include "asterisk/res_pjsip_session.h"
#include "asterisk/channel.h"
#include "asterisk/module.h"
#include "asterisk/callerid.h"
/*!
* \internal
* \brief Set an ast_party_id name and number based on an identity header.
* \param hdr From, P-Asserted-Identity, or Remote-Party-ID header on incoming message
* \param[out] id The ID to set data on
*/
static void set_id_from_hdr(pjsip_fromto_hdr *hdr, struct ast_party_id *id)
{
char cid_name[AST_CHANNEL_NAME];
char cid_num[AST_CHANNEL_NAME];
pjsip_sip_uri *uri;
pjsip_name_addr *id_name_addr = (pjsip_name_addr *) hdr->uri;
uri = pjsip_uri_get_uri(id_name_addr);
ast_copy_pj_str(cid_name, &id_name_addr->display, sizeof(cid_name));
ast_copy_pj_str(cid_num, &uri->user, sizeof(cid_num));
ast_free(id->name.str);
id->name.str = ast_strdup(cid_name);
if (!ast_strlen_zero(cid_name)) {
id->name.valid = 1;
}
ast_free(id->number.str);
id->number.str = ast_strdup(cid_num);
if (!ast_strlen_zero(cid_num)) {
id->number.valid = 1;
}
}
/*!
* \internal
* \brief Get a P-Asserted-Identity or Remote-Party-ID header from an incoming message
*
* This function will parse the header as if it were a From header. This allows for us
* to easily manipulate the URI, as well as add, modify, or remove parameters from the
* header
*
* \param rdata The incoming message
* \param header_name The name of the ID header to find
* \retval NULL No ID header present or unable to parse ID header
* \retval non-NULL The parsed ID header
*/
static pjsip_fromto_hdr *get_id_header(pjsip_rx_data *rdata, const pj_str_t *header_name)
{
static const pj_str_t from = { "From", 4 };
pj_str_t header_content;
pjsip_fromto_hdr *parsed_hdr;
pjsip_generic_string_hdr *ident = pjsip_msg_find_hdr_by_name(rdata->msg_info.msg,
header_name, NULL);
int parsed_len;
if (!ident) {
return NULL;
}
pj_strdup_with_null(rdata->tp_info.pool, &header_content, &ident->hvalue);
parsed_hdr = pjsip_parse_hdr(rdata->tp_info.pool, &from, header_content.ptr,
pj_strlen(&header_content), &parsed_len);
if (!parsed_hdr) {
return NULL;
}
return parsed_hdr;
}
/*!
* \internal
* \brief Set an ast_party_id structure based on data in a P-Asserted-Identity header
*
* This makes use of \ref set_id_from_hdr for setting name and number. It uses
* the contents of a Privacy header in order to set presentation information.
*
* \param rdata The incoming message
* \param[out] id The ID to set
* \retval 0 Successfully set the party ID
* \retval non-zero Could not set the party ID
*/
static int set_id_from_pai(pjsip_rx_data *rdata, struct ast_party_id *id)
{
static const pj_str_t pai_str = { "P-Asserted-Identity", 19 };
static const pj_str_t privacy_str = { "Privacy", 7 };
pjsip_fromto_hdr *pai_hdr = get_id_header(rdata, &pai_str);
pjsip_generic_string_hdr *privacy;
if (!pai_hdr) {
return -1;
}
set_id_from_hdr(pai_hdr, id);
if (!id->number.valid) {
return -1;
}
privacy = pjsip_msg_find_hdr_by_name(rdata->msg_info.msg, &privacy_str, NULL);
chan_pjsip/res_pjsip_callerid: Make Party ID handling simpler and consistent. The res_pjsip modules were manually checking both name and number presentation values when there is a function that determines the combined presentation for a party ID struct. The function takes into account if the name or number components are valid while the manual code rarely checked if the data was even valid. * Made use ast_party_id_presentation() rather than manually checking party ID presentation values. * Ensure that set_id_from_pai() and set_id_from_rpid() will not return presentation values other than what is pulled out of the SIP headers. It is best if the code doesn't assume that AST_PRES_ALLOWED and AST_PRES_USER_NUMBER_UNSCREENED are zero. * Fixed copy paste error in add_privacy_params() dealing with RPID privacy. * Pulled the id->number.valid test from add_privacy_header() and add_privacy_params() up into the parent function add_id_headers() to skip adding PAI/RPID headers earlier. * Made update_connected_line_information() not send out connected line updates if the connected line number is invalid. Lower level code would not add the party ID information and thus the sent message would be unnecessary. * Eliminated RAII_VAR usage in send_direct_media_request(). Review: https://reviewboard.asterisk.org/r/4472/ ........ Merged revisions 432892 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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if (privacy && !pj_stricmp2(&privacy->hvalue, "id")) {
id->number.presentation = AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED;
id->name.presentation = AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED;
chan_pjsip/res_pjsip_callerid: Make Party ID handling simpler and consistent. The res_pjsip modules were manually checking both name and number presentation values when there is a function that determines the combined presentation for a party ID struct. The function takes into account if the name or number components are valid while the manual code rarely checked if the data was even valid. * Made use ast_party_id_presentation() rather than manually checking party ID presentation values. * Ensure that set_id_from_pai() and set_id_from_rpid() will not return presentation values other than what is pulled out of the SIP headers. It is best if the code doesn't assume that AST_PRES_ALLOWED and AST_PRES_USER_NUMBER_UNSCREENED are zero. * Fixed copy paste error in add_privacy_params() dealing with RPID privacy. * Pulled the id->number.valid test from add_privacy_header() and add_privacy_params() up into the parent function add_id_headers() to skip adding PAI/RPID headers earlier. * Made update_connected_line_information() not send out connected line updates if the connected line number is invalid. Lower level code would not add the party ID information and thus the sent message would be unnecessary. * Eliminated RAII_VAR usage in send_direct_media_request(). Review: https://reviewboard.asterisk.org/r/4472/ ........ Merged revisions 432892 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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} else {
id->number.presentation = AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED;
id->name.presentation = AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED;
}
return 0;
}
/*!
* \internal
* \brief Set an ast_party_id structure based on data in a Remote-Party-ID header
*
* This makes use of \ref set_id_from_hdr for setting name and number. It uses
* the privacy and screen parameters in order to set presentation information.
*
* \param rdata The incoming message
* \param[out] id The ID to set
* \retval 0 Succesfully set the party ID
* \retval non-zero Could not set the party ID
*/
static int set_id_from_rpid(pjsip_rx_data *rdata, struct ast_party_id *id)
{
static const pj_str_t rpid_str = { "Remote-Party-ID", 15 };
static const pj_str_t privacy_str = { "privacy", 7 };
static const pj_str_t screen_str = { "screen", 6 };
pjsip_fromto_hdr *rpid_hdr = get_id_header(rdata, &rpid_str);
pjsip_param *screen;
pjsip_param *privacy;
if (!rpid_hdr) {
return -1;
}
set_id_from_hdr(rpid_hdr, id);
if (!id->number.valid) {
return -1;
}
privacy = pjsip_param_find(&rpid_hdr->other_param, &privacy_str);
screen = pjsip_param_find(&rpid_hdr->other_param, &screen_str);
if (privacy && !pj_stricmp2(&privacy->value, "full")) {
chan_pjsip/res_pjsip_callerid: Make Party ID handling simpler and consistent. The res_pjsip modules were manually checking both name and number presentation values when there is a function that determines the combined presentation for a party ID struct. The function takes into account if the name or number components are valid while the manual code rarely checked if the data was even valid. * Made use ast_party_id_presentation() rather than manually checking party ID presentation values. * Ensure that set_id_from_pai() and set_id_from_rpid() will not return presentation values other than what is pulled out of the SIP headers. It is best if the code doesn't assume that AST_PRES_ALLOWED and AST_PRES_USER_NUMBER_UNSCREENED are zero. * Fixed copy paste error in add_privacy_params() dealing with RPID privacy. * Pulled the id->number.valid test from add_privacy_header() and add_privacy_params() up into the parent function add_id_headers() to skip adding PAI/RPID headers earlier. * Made update_connected_line_information() not send out connected line updates if the connected line number is invalid. Lower level code would not add the party ID information and thus the sent message would be unnecessary. * Eliminated RAII_VAR usage in send_direct_media_request(). Review: https://reviewboard.asterisk.org/r/4472/ ........ Merged revisions 432892 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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id->number.presentation = AST_PRES_RESTRICTED;
id->name.presentation = AST_PRES_RESTRICTED;
} else {
id->number.presentation = AST_PRES_ALLOWED;
id->name.presentation = AST_PRES_ALLOWED;
}
if (screen && !pj_stricmp2(&screen->value, "yes")) {
id->number.presentation |= AST_PRES_USER_NUMBER_PASSED_SCREEN;
id->name.presentation |= AST_PRES_USER_NUMBER_PASSED_SCREEN;
chan_pjsip/res_pjsip_callerid: Make Party ID handling simpler and consistent. The res_pjsip modules were manually checking both name and number presentation values when there is a function that determines the combined presentation for a party ID struct. The function takes into account if the name or number components are valid while the manual code rarely checked if the data was even valid. * Made use ast_party_id_presentation() rather than manually checking party ID presentation values. * Ensure that set_id_from_pai() and set_id_from_rpid() will not return presentation values other than what is pulled out of the SIP headers. It is best if the code doesn't assume that AST_PRES_ALLOWED and AST_PRES_USER_NUMBER_UNSCREENED are zero. * Fixed copy paste error in add_privacy_params() dealing with RPID privacy. * Pulled the id->number.valid test from add_privacy_header() and add_privacy_params() up into the parent function add_id_headers() to skip adding PAI/RPID headers earlier. * Made update_connected_line_information() not send out connected line updates if the connected line number is invalid. Lower level code would not add the party ID information and thus the sent message would be unnecessary. * Eliminated RAII_VAR usage in send_direct_media_request(). Review: https://reviewboard.asterisk.org/r/4472/ ........ Merged revisions 432892 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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} else {
id->number.presentation |= AST_PRES_USER_NUMBER_UNSCREENED;
id->name.presentation |= AST_PRES_USER_NUMBER_UNSCREENED;
}
return 0;
}
/*!
* \internal
* \brief Set an ast_party_id structure based on data in a From
*
* This makes use of \ref set_id_from_hdr for setting name and number. It uses
* no information from the message in order to set privacy. It relies on endpoint
* configuration for privacy information.
*
* \param rdata The incoming message
* \param[out] id The ID to set
* \retval 0 Succesfully set the party ID
* \retval non-zero Could not set the party ID
*/
static int set_id_from_from(struct pjsip_rx_data *rdata, struct ast_party_id *id)
{
pjsip_fromto_hdr *from = pjsip_msg_find_hdr(rdata->msg_info.msg,
PJSIP_H_FROM, rdata->msg_info.msg->hdr.next);
if (!from) {
/* This had better not happen */
return -1;
}
set_id_from_hdr(from, id);
if (!id->number.valid) {
return -1;
}
return 0;
}
/*!
* \internal
* \brief Determine if a connected line update should be queued
*
* This uses information about the session and the ID that would be queued
* in the connected line update in order to determine if we should queue
* a connected line update.
*
* \param session The session whose channel we wish to queue the connected line update on
* \param id The identification information that would be queued on the connected line update
* \retval 0 We should not queue a connected line update
* \retval non-zero We should queue a connected line update
*/
static int should_queue_connected_line_update(const struct ast_sip_session *session, const struct ast_party_id *id)
{
/* Invalid number means no update */
if (!id->number.valid) {
return 0;
}
/* If the session has never communicated an update or if the
* new ID has a different number than the session, then we
* should queue an update
*/
if (ast_strlen_zero(session->id.number.str) ||
strcmp(session->id.number.str, id->number.str)) {
return 1;
}
/* By making it to this point, it means the number is not enough
* to determine if an update should be sent. Now we look at
* the name
*/
/* If the number couldn't warrant an update and the name is
* invalid, then no update
*/
if (!id->name.valid) {
return 0;
}
/* If the name has changed or we don't have a name set for the
* session, then we should send an update
*/
if (ast_strlen_zero(session->id.name.str) ||
strcmp(session->id.name.str, id->name.str)) {
return 1;
}
/* Neither the name nor the number have changed. No update */
return 0;
}
/*!
* \internal
* \brief Queue a connected line update on a session's channel.
* \param session The session whose channel should have the connected line update queued upon.
* \param id The identification information to place in the connected line update
*/
static void queue_connected_line_update(struct ast_sip_session *session, const struct ast_party_id *id)
{
struct ast_party_connected_line connected;
chan_pjsip: Fix attended transfer connected line name update. A calls B B answers B SIP attended transfers to C C answers, B and C can see each other's connected line information B completes the transfer A has number but no name connected line information about C while C has the full information about A I examined the incoming and outgoing party id information handling of chan_pjsip and found several issues: * Fixed ast_sip_session_create_outgoing() not setting up the configured endpoint id as the new channel's caller id. This is why party A got default connected line information. * Made update_initial_connected_line() use the channel's CALLERID(id) information. The core, app_dial, or predial routine may have filled in or changed the endpoint caller id information. * Fixed chan_pjsip_new() not setting the full party id information available on the caller id and ANI party id. This includes the configured callerid_tag string and other party id fields. * Fixed accessing channel party id information without the channel lock held. * Fixed using the effective connected line id without doing a deep copy outside of holding the channel lock. Shallow copy string pointers can become stale if the channel lock is not held. * Made queue_connected_line_update() also update the channel's CALLERID(id) information. Moving the channel to another bridge would need the information there for the new bridge peer. * Fixed off nominal memory leak in update_incoming_connected_line(). * Added pjsip.conf callerid_tag string to party id information from enabled trust_inbound endpoint in caller_id_incoming_request(). AFS-98 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/3913/ ........ Merged revisions 421400 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421403 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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struct ast_party_caller caller;
chan_pjsip: Fix attended transfer connected line name update. A calls B B answers B SIP attended transfers to C C answers, B and C can see each other's connected line information B completes the transfer A has number but no name connected line information about C while C has the full information about A I examined the incoming and outgoing party id information handling of chan_pjsip and found several issues: * Fixed ast_sip_session_create_outgoing() not setting up the configured endpoint id as the new channel's caller id. This is why party A got default connected line information. * Made update_initial_connected_line() use the channel's CALLERID(id) information. The core, app_dial, or predial routine may have filled in or changed the endpoint caller id information. * Fixed chan_pjsip_new() not setting the full party id information available on the caller id and ANI party id. This includes the configured callerid_tag string and other party id fields. * Fixed accessing channel party id information without the channel lock held. * Fixed using the effective connected line id without doing a deep copy outside of holding the channel lock. Shallow copy string pointers can become stale if the channel lock is not held. * Made queue_connected_line_update() also update the channel's CALLERID(id) information. Moving the channel to another bridge would need the information there for the new bridge peer. * Fixed off nominal memory leak in update_incoming_connected_line(). * Added pjsip.conf callerid_tag string to party id information from enabled trust_inbound endpoint in caller_id_incoming_request(). AFS-98 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/3913/ ........ Merged revisions 421400 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421403 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/* Fill connected line information */
ast_party_connected_line_init(&connected);
chan_pjsip: Fix attended transfer connected line name update. A calls B B answers B SIP attended transfers to C C answers, B and C can see each other's connected line information B completes the transfer A has number but no name connected line information about C while C has the full information about A I examined the incoming and outgoing party id information handling of chan_pjsip and found several issues: * Fixed ast_sip_session_create_outgoing() not setting up the configured endpoint id as the new channel's caller id. This is why party A got default connected line information. * Made update_initial_connected_line() use the channel's CALLERID(id) information. The core, app_dial, or predial routine may have filled in or changed the endpoint caller id information. * Fixed chan_pjsip_new() not setting the full party id information available on the caller id and ANI party id. This includes the configured callerid_tag string and other party id fields. * Fixed accessing channel party id information without the channel lock held. * Fixed using the effective connected line id without doing a deep copy outside of holding the channel lock. Shallow copy string pointers can become stale if the channel lock is not held. * Made queue_connected_line_update() also update the channel's CALLERID(id) information. Moving the channel to another bridge would need the information there for the new bridge peer. * Fixed off nominal memory leak in update_incoming_connected_line(). * Added pjsip.conf callerid_tag string to party id information from enabled trust_inbound endpoint in caller_id_incoming_request(). AFS-98 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/3913/ ........ Merged revisions 421400 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421403 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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connected.id = *id;
connected.id.tag = session->endpoint->id.self.tag;
connected.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
chan_pjsip: Fix attended transfer connected line name update. A calls B B answers B SIP attended transfers to C C answers, B and C can see each other's connected line information B completes the transfer A has number but no name connected line information about C while C has the full information about A I examined the incoming and outgoing party id information handling of chan_pjsip and found several issues: * Fixed ast_sip_session_create_outgoing() not setting up the configured endpoint id as the new channel's caller id. This is why party A got default connected line information. * Made update_initial_connected_line() use the channel's CALLERID(id) information. The core, app_dial, or predial routine may have filled in or changed the endpoint caller id information. * Fixed chan_pjsip_new() not setting the full party id information available on the caller id and ANI party id. This includes the configured callerid_tag string and other party id fields. * Fixed accessing channel party id information without the channel lock held. * Fixed using the effective connected line id without doing a deep copy outside of holding the channel lock. Shallow copy string pointers can become stale if the channel lock is not held. * Made queue_connected_line_update() also update the channel's CALLERID(id) information. Moving the channel to another bridge would need the information there for the new bridge peer. * Fixed off nominal memory leak in update_incoming_connected_line(). * Added pjsip.conf callerid_tag string to party id information from enabled trust_inbound endpoint in caller_id_incoming_request(). AFS-98 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/3913/ ........ Merged revisions 421400 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421403 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-19 16:16:03 +00:00
/* Save to channel driver copy */
ast_party_id_copy(&session->id, &connected.id);
chan_pjsip: Fix attended transfer connected line name update. A calls B B answers B SIP attended transfers to C C answers, B and C can see each other's connected line information B completes the transfer A has number but no name connected line information about C while C has the full information about A I examined the incoming and outgoing party id information handling of chan_pjsip and found several issues: * Fixed ast_sip_session_create_outgoing() not setting up the configured endpoint id as the new channel's caller id. This is why party A got default connected line information. * Made update_initial_connected_line() use the channel's CALLERID(id) information. The core, app_dial, or predial routine may have filled in or changed the endpoint caller id information. * Fixed chan_pjsip_new() not setting the full party id information available on the caller id and ANI party id. This includes the configured callerid_tag string and other party id fields. * Fixed accessing channel party id information without the channel lock held. * Fixed using the effective connected line id without doing a deep copy outside of holding the channel lock. Shallow copy string pointers can become stale if the channel lock is not held. * Made queue_connected_line_update() also update the channel's CALLERID(id) information. Moving the channel to another bridge would need the information there for the new bridge peer. * Fixed off nominal memory leak in update_incoming_connected_line(). * Added pjsip.conf callerid_tag string to party id information from enabled trust_inbound endpoint in caller_id_incoming_request(). AFS-98 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/3913/ ........ Merged revisions 421400 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421403 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-19 16:16:03 +00:00
/* Update our channel CALLERID() */
ast_party_caller_init(&caller);
caller.id = connected.id;
caller.ani = connected.id;
caller.ani2 = ast_channel_caller(session->channel)->ani2;
ast_channel_set_caller_event(session->channel, &caller, NULL);
/* Tell peer about the new connected line information. */
ast_channel_queue_connected_line_update(session->channel, &connected, NULL);
}
/*!
* \internal
* \brief Make updates to connected line information based on an incoming request.
*
* This will get identity information from an incoming request. Once the identification is
* retrieved, we will check if the new information warrants a connected line update and queue
* a connected line update if so.
*
* \param session The session on which we received an incoming request
* \param rdata The incoming request
*/
static void update_incoming_connected_line(struct ast_sip_session *session, pjsip_rx_data *rdata)
{
struct ast_party_id id;
if (!session->endpoint->id.trust_inbound) {
return;
}
ast_party_id_init(&id);
chan_pjsip: Fix attended transfer connected line name update. A calls B B answers B SIP attended transfers to C C answers, B and C can see each other's connected line information B completes the transfer A has number but no name connected line information about C while C has the full information about A I examined the incoming and outgoing party id information handling of chan_pjsip and found several issues: * Fixed ast_sip_session_create_outgoing() not setting up the configured endpoint id as the new channel's caller id. This is why party A got default connected line information. * Made update_initial_connected_line() use the channel's CALLERID(id) information. The core, app_dial, or predial routine may have filled in or changed the endpoint caller id information. * Fixed chan_pjsip_new() not setting the full party id information available on the caller id and ANI party id. This includes the configured callerid_tag string and other party id fields. * Fixed accessing channel party id information without the channel lock held. * Fixed using the effective connected line id without doing a deep copy outside of holding the channel lock. Shallow copy string pointers can become stale if the channel lock is not held. * Made queue_connected_line_update() also update the channel's CALLERID(id) information. Moving the channel to another bridge would need the information there for the new bridge peer. * Fixed off nominal memory leak in update_incoming_connected_line(). * Added pjsip.conf callerid_tag string to party id information from enabled trust_inbound endpoint in caller_id_incoming_request(). AFS-98 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/3913/ ........ Merged revisions 421400 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421403 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-19 16:16:03 +00:00
if (!set_id_from_pai(rdata, &id) || !set_id_from_rpid(rdata, &id)) {
if (should_queue_connected_line_update(session, &id)) {
queue_connected_line_update(session, &id);
}
}
ast_party_id_free(&id);
}
/*!
* \internal
* \brief Session supplement callback on an incoming INVITE request
*
* If we are receiving an initial INVITE, then we will set the session's identity
* based on the INVITE or configured endpoint values. If we are receiving a reinvite,
* then we will potentially queue a connected line update via the \ref update_incoming_connected_line
* function
*
* \param session The session that has received an INVITE
* \param rdata The incoming INVITE
*/
static int caller_id_incoming_request(struct ast_sip_session *session, pjsip_rx_data *rdata)
{
if (!session->channel) {
chan_pjsip: Fix attended transfer connected line name update. A calls B B answers B SIP attended transfers to C C answers, B and C can see each other's connected line information B completes the transfer A has number but no name connected line information about C while C has the full information about A I examined the incoming and outgoing party id information handling of chan_pjsip and found several issues: * Fixed ast_sip_session_create_outgoing() not setting up the configured endpoint id as the new channel's caller id. This is why party A got default connected line information. * Made update_initial_connected_line() use the channel's CALLERID(id) information. The core, app_dial, or predial routine may have filled in or changed the endpoint caller id information. * Fixed chan_pjsip_new() not setting the full party id information available on the caller id and ANI party id. This includes the configured callerid_tag string and other party id fields. * Fixed accessing channel party id information without the channel lock held. * Fixed using the effective connected line id without doing a deep copy outside of holding the channel lock. Shallow copy string pointers can become stale if the channel lock is not held. * Made queue_connected_line_update() also update the channel's CALLERID(id) information. Moving the channel to another bridge would need the information there for the new bridge peer. * Fixed off nominal memory leak in update_incoming_connected_line(). * Added pjsip.conf callerid_tag string to party id information from enabled trust_inbound endpoint in caller_id_incoming_request(). AFS-98 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/3913/ ........ Merged revisions 421400 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421403 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-19 16:16:03 +00:00
/*
* Since we have no channel this must be the initial inbound
* INVITE. Set the session ID directly because the channel
* has not been created yet.
chan_pjsip: Fix attended transfer connected line name update. A calls B B answers B SIP attended transfers to C C answers, B and C can see each other's connected line information B completes the transfer A has number but no name connected line information about C while C has the full information about A I examined the incoming and outgoing party id information handling of chan_pjsip and found several issues: * Fixed ast_sip_session_create_outgoing() not setting up the configured endpoint id as the new channel's caller id. This is why party A got default connected line information. * Made update_initial_connected_line() use the channel's CALLERID(id) information. The core, app_dial, or predial routine may have filled in or changed the endpoint caller id information. * Fixed chan_pjsip_new() not setting the full party id information available on the caller id and ANI party id. This includes the configured callerid_tag string and other party id fields. * Fixed accessing channel party id information without the channel lock held. * Fixed using the effective connected line id without doing a deep copy outside of holding the channel lock. Shallow copy string pointers can become stale if the channel lock is not held. * Made queue_connected_line_update() also update the channel's CALLERID(id) information. Moving the channel to another bridge would need the information there for the new bridge peer. * Fixed off nominal memory leak in update_incoming_connected_line(). * Added pjsip.conf callerid_tag string to party id information from enabled trust_inbound endpoint in caller_id_incoming_request(). AFS-98 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/3913/ ........ Merged revisions 421400 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421403 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-19 16:16:03 +00:00
*/
if (session->endpoint->id.trust_inbound
&& (!set_id_from_pai(rdata, &session->id)
|| !set_id_from_rpid(rdata, &session->id))) {
ast_free(session->id.tag);
session->id.tag = ast_strdup(session->endpoint->id.self.tag);
return 0;
}
ast_party_id_copy(&session->id, &session->endpoint->id.self);
if (!session->endpoint->id.self.number.valid) {
set_id_from_from(rdata, &session->id);
}
} else {
/*
* ReINVITE or UPDATE. Check for changes to the ID and queue
* a connected line update if necessary.
*/
update_incoming_connected_line(session, rdata);
}
return 0;
}
/*!
* \internal
* \brief Session supplement callback on INVITE response
*
* INVITE responses could result in queuing connected line updates.
*
* \param session The session on which communication is happening
* \param rdata The incoming INVITE response
*/
static void caller_id_incoming_response(struct ast_sip_session *session, pjsip_rx_data *rdata)
{
if (!session->channel) {
return;
}
update_incoming_connected_line(session, rdata);
}
/*!
* \internal
* \brief Set name and number information on an identity header.
* \param pool Memory pool to use for string duplication
* \param id_hdr A From, P-Asserted-Identity, or Remote-Party-ID header to modify
* \param id The identity information to apply to the header
*/
static void modify_id_header(pj_pool_t *pool, pjsip_fromto_hdr *id_hdr, const struct ast_party_id *id)
{
pjsip_name_addr *id_name_addr;
pjsip_sip_uri *id_uri;
id_name_addr = (pjsip_name_addr *) id_hdr->uri;
id_uri = pjsip_uri_get_uri(id_name_addr->uri);
if (id->name.valid) {
int name_buf_len = strlen(id->name.str) * 2 + 1;
char *name_buf = ast_alloca(name_buf_len);
ast_escape_quoted(id->name.str, name_buf, name_buf_len);
pj_strdup2(pool, &id_name_addr->display, name_buf);
}
if (id->number.valid) {
pj_strdup2(pool, &id_uri->user, id->number.str);
}
}
/*!
* \internal
* \brief Create an identity header for an outgoing message
* \param hdr_name The name of the header to create
* \param tdata The message to place the header on
* \param id The identification information for the new header
* \return newly-created header
*/
static pjsip_fromto_hdr *create_new_id_hdr(const pj_str_t *hdr_name, pjsip_tx_data *tdata, const struct ast_party_id *id)
{
pjsip_fromto_hdr *id_hdr;
pjsip_fromto_hdr *base;
pjsip_name_addr *id_name_addr;
pjsip_sip_uri *id_uri;
base = tdata->msg->type == PJSIP_REQUEST_MSG ? PJSIP_MSG_FROM_HDR(tdata->msg) :
PJSIP_MSG_TO_HDR(tdata->msg);
id_hdr = pjsip_from_hdr_create(tdata->pool);
id_hdr->type = PJSIP_H_OTHER;
pj_strdup(tdata->pool, &id_hdr->name, hdr_name);
id_hdr->sname.slen = 0;
id_name_addr = pjsip_uri_clone(tdata->pool, base->uri);
id_uri = pjsip_uri_get_uri(id_name_addr->uri);
if (id->name.valid) {
int name_buf_len = strlen(id->name.str) * 2 + 1;
char *name_buf = ast_alloca(name_buf_len);
ast_escape_quoted(id->name.str, name_buf, name_buf_len);
pj_strdup2(tdata->pool, &id_name_addr->display, name_buf);
}
pj_strdup2(tdata->pool, &id_uri->user, id->number.str);
id_hdr->uri = (pjsip_uri *) id_name_addr;
return id_hdr;
}
/*!
* \internal
* \brief Add a Privacy header to an outbound message
*
* When sending a P-Asserted-Identity header, if privacy is requested, then we
* will need to indicate such by adding a Privacy header. Similarly, if no
* privacy is requested, and a Privacy header already exists on the message,
* then the old Privacy header should be removed.
*
* \param tdata The outbound message to add the Privacy header to
* \param id The id information used to determine privacy
*/
static void add_privacy_header(pjsip_tx_data *tdata, const struct ast_party_id *id)
{
static const pj_str_t pj_privacy_name = { "Privacy", 7 };
static const pj_str_t pj_privacy_value = { "id", 2 };
pjsip_hdr *old_privacy;
old_privacy = pjsip_msg_find_hdr_by_name(tdata->msg, &pj_privacy_name, NULL);
chan_pjsip/res_pjsip_callerid: Make Party ID handling simpler and consistent. The res_pjsip modules were manually checking both name and number presentation values when there is a function that determines the combined presentation for a party ID struct. The function takes into account if the name or number components are valid while the manual code rarely checked if the data was even valid. * Made use ast_party_id_presentation() rather than manually checking party ID presentation values. * Ensure that set_id_from_pai() and set_id_from_rpid() will not return presentation values other than what is pulled out of the SIP headers. It is best if the code doesn't assume that AST_PRES_ALLOWED and AST_PRES_USER_NUMBER_UNSCREENED are zero. * Fixed copy paste error in add_privacy_params() dealing with RPID privacy. * Pulled the id->number.valid test from add_privacy_header() and add_privacy_params() up into the parent function add_id_headers() to skip adding PAI/RPID headers earlier. * Made update_connected_line_information() not send out connected line updates if the connected line number is invalid. Lower level code would not add the party ID information and thus the sent message would be unnecessary. * Eliminated RAII_VAR usage in send_direct_media_request(). Review: https://reviewboard.asterisk.org/r/4472/ ........ Merged revisions 432892 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-03-13 16:37:17 +00:00
if ((ast_party_id_presentation(id) & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED) {
if (old_privacy) {
pj_list_erase(old_privacy);
}
} else if (!old_privacy) {
pjsip_generic_string_hdr *privacy_hdr = pjsip_generic_string_hdr_create(
tdata->pool, &pj_privacy_name, &pj_privacy_value);
pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *)privacy_hdr);
}
}
/*!
* \internal
* \brief Add a P-Asserted-Identity header to an outbound message
* \param tdata The message to add the header to
* \param id The identification information used to populate the header
*/
static void add_pai_header(pjsip_tx_data *tdata, const struct ast_party_id *id)
{
static const pj_str_t pj_pai_name = { "P-Asserted-Identity", 19 };
pjsip_fromto_hdr *pai_hdr;
pjsip_fromto_hdr *old_pai;
/* Since inv_session reuses responses, we have to make sure there's not already
* a P-Asserted-Identity present. If there is, we just modify the old one.
*/
old_pai = pjsip_msg_find_hdr_by_name(tdata->msg, &pj_pai_name, NULL);
if (old_pai) {
modify_id_header(tdata->pool, old_pai, id);
add_privacy_header(tdata, id);
return;
}
pai_hdr = create_new_id_hdr(&pj_pai_name, tdata, id);
if (!pai_hdr) {
return;
}
add_privacy_header(tdata, id);
pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *)pai_hdr);
}
/*!
* \internal
* \brief Add privacy and screen parameters to a Remote-Party-ID header.
*
* If privacy is requested, then the privacy and screen parameters need to
* reflect this. Similarly, if no privacy or screening is to be communicated,
* we need to make sure that any previously set values are updated.
*
* \param tdata The message where the Remote-Party-ID header is
* \param hdr The header on which the parameters are being added
* \param id The identification information used to determine privacy
*/
static void add_privacy_params(pjsip_tx_data *tdata, pjsip_fromto_hdr *hdr, const struct ast_party_id *id)
{
static const pj_str_t privacy_str = { "privacy", 7 };
static const pj_str_t screen_str = { "screen", 6 };
static const pj_str_t privacy_full_str = { "full", 4 };
static const pj_str_t privacy_off_str = { "off", 3 };
static const pj_str_t screen_yes_str = { "yes", 3 };
static const pj_str_t screen_no_str = { "no", 2 };
pjsip_param *old_privacy;
pjsip_param *old_screen;
pjsip_param *privacy;
pjsip_param *screen;
chan_pjsip/res_pjsip_callerid: Make Party ID handling simpler and consistent. The res_pjsip modules were manually checking both name and number presentation values when there is a function that determines the combined presentation for a party ID struct. The function takes into account if the name or number components are valid while the manual code rarely checked if the data was even valid. * Made use ast_party_id_presentation() rather than manually checking party ID presentation values. * Ensure that set_id_from_pai() and set_id_from_rpid() will not return presentation values other than what is pulled out of the SIP headers. It is best if the code doesn't assume that AST_PRES_ALLOWED and AST_PRES_USER_NUMBER_UNSCREENED are zero. * Fixed copy paste error in add_privacy_params() dealing with RPID privacy. * Pulled the id->number.valid test from add_privacy_header() and add_privacy_params() up into the parent function add_id_headers() to skip adding PAI/RPID headers earlier. * Made update_connected_line_information() not send out connected line updates if the connected line number is invalid. Lower level code would not add the party ID information and thus the sent message would be unnecessary. * Eliminated RAII_VAR usage in send_direct_media_request(). Review: https://reviewboard.asterisk.org/r/4472/ ........ Merged revisions 432892 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-03-13 16:37:17 +00:00
int presentation;
old_privacy = pjsip_param_find(&hdr->other_param, &privacy_str);
old_screen = pjsip_param_find(&hdr->other_param, &screen_str);
if (!old_privacy) {
privacy = PJ_POOL_ALLOC_T(tdata->pool, pjsip_param);
privacy->name = privacy_str;
pj_list_insert_before(&hdr->other_param, privacy);
} else {
privacy = old_privacy;
}
if (!old_screen) {
screen = PJ_POOL_ALLOC_T(tdata->pool, pjsip_param);
screen->name = screen_str;
pj_list_insert_before(&hdr->other_param, screen);
} else {
screen = old_screen;
}
chan_pjsip/res_pjsip_callerid: Make Party ID handling simpler and consistent. The res_pjsip modules were manually checking both name and number presentation values when there is a function that determines the combined presentation for a party ID struct. The function takes into account if the name or number components are valid while the manual code rarely checked if the data was even valid. * Made use ast_party_id_presentation() rather than manually checking party ID presentation values. * Ensure that set_id_from_pai() and set_id_from_rpid() will not return presentation values other than what is pulled out of the SIP headers. It is best if the code doesn't assume that AST_PRES_ALLOWED and AST_PRES_USER_NUMBER_UNSCREENED are zero. * Fixed copy paste error in add_privacy_params() dealing with RPID privacy. * Pulled the id->number.valid test from add_privacy_header() and add_privacy_params() up into the parent function add_id_headers() to skip adding PAI/RPID headers earlier. * Made update_connected_line_information() not send out connected line updates if the connected line number is invalid. Lower level code would not add the party ID information and thus the sent message would be unnecessary. * Eliminated RAII_VAR usage in send_direct_media_request(). Review: https://reviewboard.asterisk.org/r/4472/ ........ Merged revisions 432892 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-03-13 16:37:17 +00:00
presentation = ast_party_id_presentation(id);
if ((presentation & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED) {
privacy->value = privacy_off_str;
} else {
privacy->value = privacy_full_str;
}
chan_pjsip/res_pjsip_callerid: Make Party ID handling simpler and consistent. The res_pjsip modules were manually checking both name and number presentation values when there is a function that determines the combined presentation for a party ID struct. The function takes into account if the name or number components are valid while the manual code rarely checked if the data was even valid. * Made use ast_party_id_presentation() rather than manually checking party ID presentation values. * Ensure that set_id_from_pai() and set_id_from_rpid() will not return presentation values other than what is pulled out of the SIP headers. It is best if the code doesn't assume that AST_PRES_ALLOWED and AST_PRES_USER_NUMBER_UNSCREENED are zero. * Fixed copy paste error in add_privacy_params() dealing with RPID privacy. * Pulled the id->number.valid test from add_privacy_header() and add_privacy_params() up into the parent function add_id_headers() to skip adding PAI/RPID headers earlier. * Made update_connected_line_information() not send out connected line updates if the connected line number is invalid. Lower level code would not add the party ID information and thus the sent message would be unnecessary. * Eliminated RAII_VAR usage in send_direct_media_request(). Review: https://reviewboard.asterisk.org/r/4472/ ........ Merged revisions 432892 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-03-13 16:37:17 +00:00
if ((presentation & AST_PRES_NUMBER_TYPE) == AST_PRES_USER_NUMBER_PASSED_SCREEN) {
screen->value = screen_yes_str;
} else {
screen->value = screen_no_str;
}
}
/*!
* \internal
* \brief Add a Remote-Party-ID header to an outbound message
* \param tdata The message to add the header to
* \param id The identification information used to populate the header
*/
static void add_rpid_header(pjsip_tx_data *tdata, const struct ast_party_id *id)
{
static const pj_str_t pj_rpid_name = { "Remote-Party-ID", 15 };
pjsip_fromto_hdr *rpid_hdr;
pjsip_fromto_hdr *old_rpid;
/* Since inv_session reuses responses, we have to make sure there's not already
* a P-Asserted-Identity present. If there is, we just modify the old one.
*/
old_rpid = pjsip_msg_find_hdr_by_name(tdata->msg, &pj_rpid_name, NULL);
if (old_rpid) {
modify_id_header(tdata->pool, old_rpid, id);
add_privacy_params(tdata, old_rpid, id);
return;
}
rpid_hdr = create_new_id_hdr(&pj_rpid_name, tdata, id);
if (!rpid_hdr) {
return;
}
add_privacy_params(tdata, rpid_hdr, id);
pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *)rpid_hdr);
}
/*!
* \internal
* \brief Add any appropriate identification headers to an outbound SIP message
*
* This will determine if an outbound message should have identification headers and
* will add the appropriately configured headers
*
* \param session The session on which we will be sending the message
* \param tdata The outbound message
* \param The identity information to place on the message
*/
static void add_id_headers(const struct ast_sip_session *session, pjsip_tx_data *tdata, const struct ast_party_id *id)
{
chan_pjsip/res_pjsip_callerid: Make Party ID handling simpler and consistent. The res_pjsip modules were manually checking both name and number presentation values when there is a function that determines the combined presentation for a party ID struct. The function takes into account if the name or number components are valid while the manual code rarely checked if the data was even valid. * Made use ast_party_id_presentation() rather than manually checking party ID presentation values. * Ensure that set_id_from_pai() and set_id_from_rpid() will not return presentation values other than what is pulled out of the SIP headers. It is best if the code doesn't assume that AST_PRES_ALLOWED and AST_PRES_USER_NUMBER_UNSCREENED are zero. * Fixed copy paste error in add_privacy_params() dealing with RPID privacy. * Pulled the id->number.valid test from add_privacy_header() and add_privacy_params() up into the parent function add_id_headers() to skip adding PAI/RPID headers earlier. * Made update_connected_line_information() not send out connected line updates if the connected line number is invalid. Lower level code would not add the party ID information and thus the sent message would be unnecessary. * Eliminated RAII_VAR usage in send_direct_media_request(). Review: https://reviewboard.asterisk.org/r/4472/ ........ Merged revisions 432892 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-03-13 16:37:17 +00:00
if (!id->number.valid
|| (!session->endpoint->id.trust_outbound
&& (ast_party_id_presentation(id) & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED)) {
return;
}
if (session->endpoint->id.send_pai) {
add_pai_header(tdata, id);
}
if (session->endpoint->id.send_rpid) {
add_rpid_header(tdata, id);
}
}
/*!
* \internal
* \brief Session supplement callback for outgoing INVITE requests
*
* For an initial INVITE request, we may change the From header to appropriately
* reflect the identity information. On all INVITEs (initial and reinvite) we may
* add other identity headers such as P-Asserted-Identity and Remote-Party-ID based
* on configuration and privacy settings
*
* \param session The session on which the INVITE will be sent
* \param tdata The outbound INVITE request
*/
static void caller_id_outgoing_request(struct ast_sip_session *session, pjsip_tx_data *tdata)
{
chan_pjsip: Fix attended transfer connected line name update. A calls B B answers B SIP attended transfers to C C answers, B and C can see each other's connected line information B completes the transfer A has number but no name connected line information about C while C has the full information about A I examined the incoming and outgoing party id information handling of chan_pjsip and found several issues: * Fixed ast_sip_session_create_outgoing() not setting up the configured endpoint id as the new channel's caller id. This is why party A got default connected line information. * Made update_initial_connected_line() use the channel's CALLERID(id) information. The core, app_dial, or predial routine may have filled in or changed the endpoint caller id information. * Fixed chan_pjsip_new() not setting the full party id information available on the caller id and ANI party id. This includes the configured callerid_tag string and other party id fields. * Fixed accessing channel party id information without the channel lock held. * Fixed using the effective connected line id without doing a deep copy outside of holding the channel lock. Shallow copy string pointers can become stale if the channel lock is not held. * Made queue_connected_line_update() also update the channel's CALLERID(id) information. Moving the channel to another bridge would need the information there for the new bridge peer. * Fixed off nominal memory leak in update_incoming_connected_line(). * Added pjsip.conf callerid_tag string to party id information from enabled trust_inbound endpoint in caller_id_incoming_request(). AFS-98 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/3913/ ........ Merged revisions 421400 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421403 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-19 16:16:03 +00:00
struct ast_party_id effective_id;
struct ast_party_id connected_id;
if (!session->channel) {
return;
}
chan_pjsip: Fix attended transfer connected line name update. A calls B B answers B SIP attended transfers to C C answers, B and C can see each other's connected line information B completes the transfer A has number but no name connected line information about C while C has the full information about A I examined the incoming and outgoing party id information handling of chan_pjsip and found several issues: * Fixed ast_sip_session_create_outgoing() not setting up the configured endpoint id as the new channel's caller id. This is why party A got default connected line information. * Made update_initial_connected_line() use the channel's CALLERID(id) information. The core, app_dial, or predial routine may have filled in or changed the endpoint caller id information. * Fixed chan_pjsip_new() not setting the full party id information available on the caller id and ANI party id. This includes the configured callerid_tag string and other party id fields. * Fixed accessing channel party id information without the channel lock held. * Fixed using the effective connected line id without doing a deep copy outside of holding the channel lock. Shallow copy string pointers can become stale if the channel lock is not held. * Made queue_connected_line_update() also update the channel's CALLERID(id) information. Moving the channel to another bridge would need the information there for the new bridge peer. * Fixed off nominal memory leak in update_incoming_connected_line(). * Added pjsip.conf callerid_tag string to party id information from enabled trust_inbound endpoint in caller_id_incoming_request(). AFS-98 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/3913/ ........ Merged revisions 421400 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421403 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-19 16:16:03 +00:00
/* Must do a deep copy unless we hold the channel lock the entire time. */
ast_party_id_init(&connected_id);
ast_channel_lock(session->channel);
effective_id = ast_channel_connected_effective_id(session->channel);
ast_party_id_copy(&connected_id, &effective_id);
ast_channel_unlock(session->channel);
if (session->inv_session->state < PJSIP_INV_STATE_CONFIRMED) {
/* Only change the From header on the initial outbound INVITE. Switching it
* mid-call might confuse some UAs.
*/
pjsip_fromto_hdr *from;
pjsip_dialog *dlg;
from = pjsip_msg_find_hdr(tdata->msg, PJSIP_H_FROM, tdata->msg->hdr.next);
dlg = session->inv_session->dlg;
chan_pjsip/res_pjsip_callerid: Make Party ID handling simpler and consistent. The res_pjsip modules were manually checking both name and number presentation values when there is a function that determines the combined presentation for a party ID struct. The function takes into account if the name or number components are valid while the manual code rarely checked if the data was even valid. * Made use ast_party_id_presentation() rather than manually checking party ID presentation values. * Ensure that set_id_from_pai() and set_id_from_rpid() will not return presentation values other than what is pulled out of the SIP headers. It is best if the code doesn't assume that AST_PRES_ALLOWED and AST_PRES_USER_NUMBER_UNSCREENED are zero. * Fixed copy paste error in add_privacy_params() dealing with RPID privacy. * Pulled the id->number.valid test from add_privacy_header() and add_privacy_params() up into the parent function add_id_headers() to skip adding PAI/RPID headers earlier. * Made update_connected_line_information() not send out connected line updates if the connected line number is invalid. Lower level code would not add the party ID information and thus the sent message would be unnecessary. * Eliminated RAII_VAR usage in send_direct_media_request(). Review: https://reviewboard.asterisk.org/r/4472/ ........ Merged revisions 432892 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-03-13 16:37:17 +00:00
if (ast_strlen_zero(session->endpoint->fromuser)
&& (session->endpoint->id.trust_outbound
|| (ast_party_id_presentation(&connected_id) & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED)) {
modify_id_header(tdata->pool, from, &connected_id);
modify_id_header(dlg->pool, dlg->local.info, &connected_id);
}
ast_sip_add_usereqphone(session->endpoint, tdata->pool, from->uri);
ast_sip_add_usereqphone(session->endpoint, dlg->pool, dlg->local.info->uri);
}
add_id_headers(session, tdata, &connected_id);
chan_pjsip: Fix attended transfer connected line name update. A calls B B answers B SIP attended transfers to C C answers, B and C can see each other's connected line information B completes the transfer A has number but no name connected line information about C while C has the full information about A I examined the incoming and outgoing party id information handling of chan_pjsip and found several issues: * Fixed ast_sip_session_create_outgoing() not setting up the configured endpoint id as the new channel's caller id. This is why party A got default connected line information. * Made update_initial_connected_line() use the channel's CALLERID(id) information. The core, app_dial, or predial routine may have filled in or changed the endpoint caller id information. * Fixed chan_pjsip_new() not setting the full party id information available on the caller id and ANI party id. This includes the configured callerid_tag string and other party id fields. * Fixed accessing channel party id information without the channel lock held. * Fixed using the effective connected line id without doing a deep copy outside of holding the channel lock. Shallow copy string pointers can become stale if the channel lock is not held. * Made queue_connected_line_update() also update the channel's CALLERID(id) information. Moving the channel to another bridge would need the information there for the new bridge peer. * Fixed off nominal memory leak in update_incoming_connected_line(). * Added pjsip.conf callerid_tag string to party id information from enabled trust_inbound endpoint in caller_id_incoming_request(). AFS-98 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/3913/ ........ Merged revisions 421400 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421403 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-19 16:16:03 +00:00
ast_party_id_free(&connected_id);
}
/*!
* \internal
* \brief Session supplement for outgoing INVITE response
*
* This will add P-Asserted-Identity and Remote-Party-ID headers if necessary
*
* \param session The session on which the INVITE response is to be sent
* \param tdata The outbound INVITE response
*/
static void caller_id_outgoing_response(struct ast_sip_session *session, pjsip_tx_data *tdata)
{
chan_pjsip: Fix attended transfer connected line name update. A calls B B answers B SIP attended transfers to C C answers, B and C can see each other's connected line information B completes the transfer A has number but no name connected line information about C while C has the full information about A I examined the incoming and outgoing party id information handling of chan_pjsip and found several issues: * Fixed ast_sip_session_create_outgoing() not setting up the configured endpoint id as the new channel's caller id. This is why party A got default connected line information. * Made update_initial_connected_line() use the channel's CALLERID(id) information. The core, app_dial, or predial routine may have filled in or changed the endpoint caller id information. * Fixed chan_pjsip_new() not setting the full party id information available on the caller id and ANI party id. This includes the configured callerid_tag string and other party id fields. * Fixed accessing channel party id information without the channel lock held. * Fixed using the effective connected line id without doing a deep copy outside of holding the channel lock. Shallow copy string pointers can become stale if the channel lock is not held. * Made queue_connected_line_update() also update the channel's CALLERID(id) information. Moving the channel to another bridge would need the information there for the new bridge peer. * Fixed off nominal memory leak in update_incoming_connected_line(). * Added pjsip.conf callerid_tag string to party id information from enabled trust_inbound endpoint in caller_id_incoming_request(). AFS-98 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/3913/ ........ Merged revisions 421400 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421403 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-19 16:16:03 +00:00
struct ast_party_id effective_id;
struct ast_party_id connected_id;
if (!session->channel) {
return;
}
chan_pjsip: Fix attended transfer connected line name update. A calls B B answers B SIP attended transfers to C C answers, B and C can see each other's connected line information B completes the transfer A has number but no name connected line information about C while C has the full information about A I examined the incoming and outgoing party id information handling of chan_pjsip and found several issues: * Fixed ast_sip_session_create_outgoing() not setting up the configured endpoint id as the new channel's caller id. This is why party A got default connected line information. * Made update_initial_connected_line() use the channel's CALLERID(id) information. The core, app_dial, or predial routine may have filled in or changed the endpoint caller id information. * Fixed chan_pjsip_new() not setting the full party id information available on the caller id and ANI party id. This includes the configured callerid_tag string and other party id fields. * Fixed accessing channel party id information without the channel lock held. * Fixed using the effective connected line id without doing a deep copy outside of holding the channel lock. Shallow copy string pointers can become stale if the channel lock is not held. * Made queue_connected_line_update() also update the channel's CALLERID(id) information. Moving the channel to another bridge would need the information there for the new bridge peer. * Fixed off nominal memory leak in update_incoming_connected_line(). * Added pjsip.conf callerid_tag string to party id information from enabled trust_inbound endpoint in caller_id_incoming_request(). AFS-98 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/3913/ ........ Merged revisions 421400 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421403 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-19 16:16:03 +00:00
/* Must do a deep copy unless we hold the channel lock the entire time. */
ast_party_id_init(&connected_id);
ast_channel_lock(session->channel);
effective_id = ast_channel_connected_effective_id(session->channel);
ast_party_id_copy(&connected_id, &effective_id);
ast_channel_unlock(session->channel);
add_id_headers(session, tdata, &connected_id);
chan_pjsip: Fix attended transfer connected line name update. A calls B B answers B SIP attended transfers to C C answers, B and C can see each other's connected line information B completes the transfer A has number but no name connected line information about C while C has the full information about A I examined the incoming and outgoing party id information handling of chan_pjsip and found several issues: * Fixed ast_sip_session_create_outgoing() not setting up the configured endpoint id as the new channel's caller id. This is why party A got default connected line information. * Made update_initial_connected_line() use the channel's CALLERID(id) information. The core, app_dial, or predial routine may have filled in or changed the endpoint caller id information. * Fixed chan_pjsip_new() not setting the full party id information available on the caller id and ANI party id. This includes the configured callerid_tag string and other party id fields. * Fixed accessing channel party id information without the channel lock held. * Fixed using the effective connected line id without doing a deep copy outside of holding the channel lock. Shallow copy string pointers can become stale if the channel lock is not held. * Made queue_connected_line_update() also update the channel's CALLERID(id) information. Moving the channel to another bridge would need the information there for the new bridge peer. * Fixed off nominal memory leak in update_incoming_connected_line(). * Added pjsip.conf callerid_tag string to party id information from enabled trust_inbound endpoint in caller_id_incoming_request(). AFS-98 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/3913/ ........ Merged revisions 421400 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421403 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-19 16:16:03 +00:00
ast_party_id_free(&connected_id);
}
static struct ast_sip_session_supplement caller_id_supplement = {
.method = "INVITE,UPDATE",
.priority = AST_SIP_SUPPLEMENT_PRIORITY_CHANNEL - 1000,
.incoming_request = caller_id_incoming_request,
.incoming_response = caller_id_incoming_response,
.outgoing_request = caller_id_outgoing_request,
.outgoing_response = caller_id_outgoing_response,
};
static int load_module(void)
{
CHECK_PJSIP_SESSION_MODULE_LOADED();
ast_sip_session_register_supplement(&caller_id_supplement);
return AST_MODULE_LOAD_SUCCESS;
}
static int unload_module(void)
{
ast_sip_session_unregister_supplement(&caller_id_supplement);
return 0;
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP Caller ID Support",
.support_level = AST_MODULE_SUPPORT_CORE,
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_APP_DEPEND,
);