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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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421403 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -381,11 +381,13 @@ static struct ast_channel *chan_pjsip_new(struct ast_sip_session *session, int s
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return NULL;
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}
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chan = ast_channel_alloc_with_endpoint(1, state, S_OR(session->id.number.str, ""),
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S_OR(session->id.name.str, ""), session->endpoint->accountcode, "",
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"", assignedids, requestor, 0, session->endpoint->persistent,
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"PJSIP/%s-%08x", ast_sorcery_object_get_id(session->endpoint),
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(unsigned)ast_atomic_fetchadd_int((int *)&chan_idx, +1));
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chan = ast_channel_alloc_with_endpoint(1, state,
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S_COR(session->id.number.valid, session->id.number.str, ""),
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S_COR(session->id.name.valid, session->id.name.str, ""),
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session->endpoint->accountcode, "", "", assignedids, requestor, 0,
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session->endpoint->persistent, "PJSIP/%s-%08x",
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ast_sorcery_object_get_id(session->endpoint),
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(unsigned) ast_atomic_fetchadd_int((int *) &chan_idx, +1));
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if (!chan) {
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ao2_ref(caps, -1);
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return NULL;
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@@ -432,6 +434,9 @@ static struct ast_channel *chan_pjsip_new(struct ast_sip_session *session, int s
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ast_channel_adsicpe_set(chan, AST_ADSI_UNAVAILABLE);
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ast_party_id_copy(&ast_channel_caller(chan)->id, &session->id);
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ast_party_id_copy(&ast_channel_caller(chan)->ani, &session->id);
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ast_channel_context_set(chan, session->endpoint->context);
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ast_channel_exten_set(chan, S_OR(exten, "s"));
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ast_channel_priority_set(chan, 1);
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@@ -1042,7 +1047,6 @@ static int transmit_info_with_vidupdate(void *data)
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static int update_connected_line_information(void *data)
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{
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RAII_VAR(struct ast_sip_session *, session, data, ao2_cleanup);
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struct ast_party_id connected_id;
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if ((ast_channel_state(session->channel) != AST_STATE_UP) && (session->inv_session->role == PJSIP_UAS_ROLE)) {
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int response_code = 0;
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@@ -1062,12 +1066,20 @@ static int update_connected_line_information(void *data)
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}
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} else {
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enum ast_sip_session_refresh_method method = session->endpoint->id.refresh_method;
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struct ast_party_id connected_id;
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if (session->inv_session->invite_tsx && (session->inv_session->options & PJSIP_INV_SUPPORT_UPDATE)) {
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method = AST_SIP_SESSION_REFRESH_METHOD_UPDATE;
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}
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/*
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* We can get away with a shallow copy here because we are
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* not looking at strings.
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*/
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ast_channel_lock(session->channel);
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connected_id = ast_channel_connected_effective_id(session->channel);
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ast_channel_unlock(session->channel);
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if ((session->endpoint->id.send_pai || session->endpoint->id.send_rpid) &&
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(session->endpoint->id.trust_outbound ||
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((connected_id.name.presentation & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED &&
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@@ -1492,33 +1504,26 @@ static int chan_pjsip_digit_end(struct ast_channel *ast, char digit, unsigned in
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static void update_initial_connected_line(struct ast_sip_session *session)
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{
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struct ast_party_connected_line connected;
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struct ast_set_party_connected_line update_connected;
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struct ast_sip_endpoint_id_configuration *id = &session->endpoint->id;
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if (!id->self.number.valid && !id->self.name.valid) {
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/*
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* Use the channel CALLERID() as the initial connected line data.
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* The core or a predial handler may have supplied missing values
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* from the session->endpoint->id.self about who we are calling.
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*/
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ast_channel_lock(session->channel);
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ast_party_id_copy(&session->id, &ast_channel_caller(session->channel)->id);
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ast_channel_unlock(session->channel);
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/* Supply initial connected line information if available. */
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if (!session->id.number.valid && !session->id.name.valid) {
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return;
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}
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/* Supply initial connected line information if available. */
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memset(&update_connected, 0, sizeof(update_connected));
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ast_party_connected_line_init(&connected);
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connected.id.number = id->self.number;
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connected.id.name = id->self.name;
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connected.id.tag = id->self.tag;
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connected.id = session->id;
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connected.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
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if (connected.id.number.valid) {
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update_connected.id.number = 1;
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}
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if (connected.id.name.valid) {
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update_connected.id.name = 1;
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}
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/* Invalidate any earlier private connected id representation */
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ast_set_party_id_all(&update_connected.priv);
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ast_channel_queue_connected_line_update(session->channel, &connected, &update_connected);
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ast_channel_queue_connected_line_update(session->channel, &connected, NULL);
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}
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static int call(void *data)
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@@ -1549,7 +1554,7 @@ static int chan_pjsip_call(struct ast_channel *ast, const char *dest, int timeou
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ao2_ref(channel, +1);
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if (ast_sip_push_task(channel->session->serializer, call, channel)) {
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ast_log(LOG_WARNING, "Error attempting to place outbound call to call '%s'\n", dest);
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ast_log(LOG_WARNING, "Error attempting to place outbound call to '%s'\n", dest);
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ao2_cleanup(channel);
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return -1;
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}
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