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"
#include "asterisk/conversions.h"
/*!
* \internal
* \brief Set an ANI2 integer based on OLI data in a From header
*
* This uses the contents of a From header in order to set Originating Line information.
*
* \param rdata The incoming message
* \param ani2 The ANI2 field to set
* \retval 0 Successfully parsed OLI
* \retval non-zero Could not parse OLI
*/
static int set_id_from_oli(pjsip_rx_data *rdata, int *ani2)
{
char oli[AST_CHANNEL_NAME];
pjsip_param *oli1, *oli2, *oli3;
static const pj_str_t oli_str1 = { "isup-oli", 8 };
static const pj_str_t oli_str2 = { "ss7-oli", 7 };
static const pj_str_t oli_str3 = { "oli", 3 };
pjsip_fromto_hdr *from = pjsip_msg_find_hdr(rdata->msg_info.msg,
PJSIP_H_FROM, rdata->msg_info.msg->hdr.next);
if (!from) {
return -1; /* This had better not happen */
}
if ((oli1 = pjsip_param_find(&from->other_param, &oli_str1))) {
ast_copy_pj_str(oli, &oli1->value, sizeof(oli));
} else if ((oli2 = pjsip_param_find(&from->other_param, &oli_str2))) {
ast_copy_pj_str(oli, &oli2->value, sizeof(oli));
} else if ((oli3 = pjsip_param_find(&from->other_param, &oli_str3))) {
ast_copy_pj_str(oli, &oli3->value, sizeof(oli));
} else {
return -1;
}
return ast_str_to_int(oli, ani2);
}
/*!
* \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
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/* 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
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/* 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_connected_line
|| !session->endpoint->id.trust_inbound) {
return;
}
ast_party_id_init(&id);
if (!ast_sip_set_id_connected_line(rdata, &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
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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) {
int ani2;
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
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*/
ast_sip_set_id_from_invite(rdata, &session->id, &session->endpoint->id.self, session->endpoint->id.trust_inbound);
session->ani2 = set_id_from_oli(rdata, &ani2) ? 0 : ani2;
} 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 Create an identity header for an outgoing message
* \param hdr_name The name of the header to create
* \param base
* \param tdata The message to place the header on
* \param id The identification information for the new header
* \return newly-created header
*/
2016-02-24 23:25:09 +00:00
static pjsip_fromto_hdr *create_new_id_hdr(const pj_str_t *hdr_name, pjsip_fromto_hdr *base, pjsip_tx_data *tdata, const struct ast_party_id *id)
{
pjsip_fromto_hdr *id_hdr;
pjsip_name_addr *id_name_addr;
pjsip_sip_uri *id_uri;
id_hdr = pjsip_from_hdr_create(tdata->pool);
id_hdr->type = PJSIP_H_OTHER;
id_hdr->sname = id_hdr->name = *hdr_name;
id_name_addr = pjsip_uri_clone(tdata->pool, base->uri);
id_uri = pjsip_uri_get_uri(id_name_addr->uri);
if (id->name.valid && !ast_strlen_zero(id->name.str)) {
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);
} else {
/*
* We need to clear the remnants of the clone or it'll be left set.
* pj_strdup2 is safe to call with a NULL src and it resets both slen and ptr.
*/
pj_strdup2(tdata->pool, &id_name_addr->display, NULL);
}
if (id->number.valid) {
pj_strdup2(tdata->pool, &id_uri->user, id->number.str);
} else {
/* Similar to name, make sure the number is also cleared when invalid */
pj_strdup2(tdata->pool, &id_uri->user, NULL);
}
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 session The session on which communication is happening
* \param tdata The message to add the header to
* \param id The identification information used to populate the header
*/
2016-02-24 23:25:09 +00:00
static void add_pai_header(const struct ast_sip_session *session, pjsip_tx_data *tdata, const struct ast_party_id *id)
{
static const pj_str_t pj_pai_name = { "P-Asserted-Identity", 19 };
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pjsip_fromto_hdr *base;
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) {
/* If type is OTHER, then the existing header was most likely
* added by the PJSIP_HEADER dial plan function as a simple
* name/value pair. We can't pass this to modify_id_header because
* there are no virtual functions to get the uri. We could parse
* it into a pjsip_fromto_hdr but it isn't worth it since
* modify_id_header is just going to overwrite the name and number
* anyway. We'll just remove it from the header list instead
* and create a new one.
*/
if (old_pai->type == PJSIP_H_OTHER) {
pj_list_erase(old_pai);
} else {
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ast_sip_modify_id_header(tdata->pool, old_pai, id);
add_privacy_header(tdata, id);
return;
}
}
if (tdata->msg->type == PJSIP_REQUEST_MSG) {
base = session->saved_from_hdr ? session->saved_from_hdr : PJSIP_MSG_FROM_HDR(tdata->msg);
} else {
base = PJSIP_MSG_TO_HDR(tdata->msg);
}
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pai_hdr = create_new_id_hdr(&pj_pai_name, base, tdata, id);
if (!pai_hdr) {
return;
}
add_privacy_header(tdata, id);
pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *)pai_hdr);
}
/*!
* \internal
* \brief Add party parameter to a Remote-Party-ID header.
*
* \param tdata The message where the Remote-Party-ID header is
* \param hdr The header on which the parameters are being added
* \param session The session involved
*/
static void add_party_param(pjsip_tx_data *tdata, pjsip_fromto_hdr *hdr, const struct ast_sip_session *session)
{
static const pj_str_t party_str = { "party", 5 };
static const pj_str_t calling_str = { "calling", 7 };
static const pj_str_t called_str = { "called", 6 };
pjsip_param *party;
/* The party value can't change throughout the lifetime, so it is set only once */
party = pjsip_param_find(&hdr->other_param, &party_str);
if (party) {
return;
}
party = PJ_POOL_ALLOC_T(tdata->pool, pjsip_param);
party->name = party_str;
party->value = (session->inv_session->role == PJSIP_ROLE_UAC) ? calling_str : called_str;
pj_list_insert_before(&hdr->other_param, party);
}
/*!
* \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 session The session on which communication is happening
* \param tdata The message to add the header to
* \param id The identification information used to populate the header
*/
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static void add_rpid_header(const struct ast_sip_session *session, pjsip_tx_data *tdata, const struct ast_party_id *id)
{
static const pj_str_t pj_rpid_name = { "Remote-Party-ID", 15 };
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pjsip_fromto_hdr *base;
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) {
/* If type is OTHER, then the existing header was most likely
* added by the PJSIP_HEADER dial plan function as a simple
* name/value pair. We can't pass this to modify_id_header because
* there are no virtual functions to get the uri. We could parse
* it into a pjsip_fromto_hdr but it isn't worth it since
* modify_id_header is just going to overwrite the name and number
* anyway. We'll just remove it from the header list instead
* and create a new one.
*/
if (old_rpid->type == PJSIP_H_OTHER) {
pj_list_erase(old_rpid);
} else {
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ast_sip_modify_id_header(tdata->pool, old_rpid, id);
add_party_param(tdata, old_rpid, session);
add_privacy_params(tdata, old_rpid, id);
return;
}
}
if (tdata->msg->type == PJSIP_REQUEST_MSG) {
base = session->saved_from_hdr ? session->saved_from_hdr : PJSIP_MSG_FROM_HDR(tdata->msg);
} else {
base = PJSIP_MSG_TO_HDR(tdata->msg);
}
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rpid_hdr = create_new_id_hdr(&pj_rpid_name, base, tdata, id);
if (!rpid_hdr) {
return;
}
add_party_param(tdata, rpid_hdr, session);
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 id 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)
{
Stir/Shaken Refactor Why do we need a refactor? The original stir/shaken implementation was started over 3 years ago when little was understood about practical implementation. The result was an implementation that wouldn't actually interoperate with any other stir-shaken implementations. There were also a number of stir-shaken features and RFC requirements that were never implemented such as TNAuthList certificate validation, sending Reason headers in SIP responses when verification failed but we wished to continue the call, and the ability to send Media Key(mky) grants in the Identity header when the call involved DTLS. Finally, there were some performance concerns around outgoing calls and selection of the correct certificate and private key. The configuration was keyed by an arbitrary name which meant that for every outgoing call, we had to scan the entire list of configured TNs to find the correct cert to use. With only a few TNs configured, this wasn't an issue but if you have a thousand, it could be. What's changed? * Configuration objects have been refactored to be clearer about their uses and to fix issues. * The "general" object was renamed to "verification" since it contains parameters specific to the incoming verification process. It also never handled ca_path and crl_path correctly. * A new "attestation" object was added that controls the outgoing attestation process. It sets default certificates, keys, etc. * The "certificate" object was renamed to "tn" and had it's key change to telephone number since outgoing call attestation needs to look up certificates by telephone number. * The "profile" object had more parameters added to it that can override default parameters specified in the "attestation" and "verification" objects. * The "store" object was removed altogther as it was never implemented. * We now use libjwt to create outgoing Identity headers and to parse and validate signatures on incoming Identiy headers. Our previous custom implementation was much of the source of the interoperability issues. * General code cleanup and refactor. * Moved things to better places. * Separated some of the complex functions to smaller ones. * Using context objects rather than passing tons of parameters in function calls. * Removed some complexity and unneeded encapsuation from the config objects. Resolves: #351 Resolves: #46 UserNote: Asterisk's stir-shaken feature has been refactored to correct interoperability, RFC compliance, and performance issues. See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information. UpgradeNote: The stir-shaken refactor is a breaking change but since it's not working now we don't think it matters. The stir_shaken.conf file has changed significantly which means that existing ones WILL need to be changed. The stir_shaken.conf.sample file in configs/samples/ has quite a bit more information. This is also an ABI breaking change since some of the existing objects needed to be changed or removed, and new ones added. Additionally, if res_stir_shaken is enabled in menuselect, you'll need to either have the development package for libjwt v1.15.3 installed or use the --with-libjwt-bundled option with ./configure. (cherry picked from commit e6c7f1aee0e54733709c5e41bf8305b6dc3e3dc3)
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if (!ast_sip_can_present_connected_id(session, id)) {
return;
}
if (session->endpoint->id.send_pai) {
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add_pai_header(session, tdata, id);
}
if (session->endpoint->id.send_rpid) {
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add_rpid_header(session, tdata, id);
}
}
/*!
* \internal
* \brief Session supplement callback for outgoing INVITE requests
*
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* 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
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);
}
/*!
* \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
|| (!session->endpoint->id.send_connected_line
&& session->inv_session
&& session->inv_session->state >= PJSIP_INV_STATE_EARLY)) {
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)
{
ast_module_shutdown_ref(AST_MODULE_SELF);
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,
.requires = "res_pjsip,res_pjsip_session",
);