diff --git a/CHANGES b/CHANGES index 36aabc7286..52e8f6681f 100644 --- a/CHANGES +++ b/CHANGES @@ -106,6 +106,18 @@ res_musiconhold over the channel-set musicclass. This allows separate hold-music from application (e.g. Queue or Dial) specified music. +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 13.3.0 to Asterisk 13.4.0 ------------ +------------------------------------------------------------------------------ + +chan_pjsip +------------------ + * New 'rpid_immediate' option to control if connected line update information + goes to the caller immediately or waits for another reason to send the + connected line information update. See the online option documentation for + more information. Defaults to 'no' as setting it to 'yes' can result in + many unnecessary messages being sent to the caller. + ------------------------------------------------------------------------------ --- Functionality changes from Asterisk 13.2.0 to Asterisk 13.3.0 ------------ ------------------------------------------------------------------------------ diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c index 11acdffe16..8d9193f933 100644 --- a/channels/chan_pjsip.c +++ b/channels/chan_pjsip.c @@ -1117,7 +1117,8 @@ static int update_connected_line_information(void *data) ast_sip_session_refresh(session, NULL, NULL, NULL, method, generate_new_sdp); } - } else if (session->inv_session->state != PJSIP_INV_STATE_DISCONNECTED + } else if (session->endpoint->id.rpid_immediate + && session->inv_session->state != PJSIP_INV_STATE_DISCONNECTED && is_colp_update_allowed(session)) { int response_code = 0; diff --git a/configs/samples/pjsip.conf.sample b/configs/samples/pjsip.conf.sample index ba8bf751ba..d3bb518f15 100644 --- a/configs/samples/pjsip.conf.sample +++ b/configs/samples/pjsip.conf.sample @@ -637,6 +637,7 @@ ; information to the called user agent (default: "yes") ;send_pai=no ; Send the P Asserted Identity header (default: "no") ;send_rpid=no ; Send the Remote Party ID header (default: "no") +;rpid_immediate=no ; Send connected line updates on unanswered incoming calls immediately. (default: "no") ;timers_min_se=90 ; Minimum session timers expiration period (default: ; "90") ;timers=yes ; Session timers for SIP packets (default: "yes") diff --git a/include/asterisk/res_pjsip.h b/include/asterisk/res_pjsip.h index 57c1a598d8..442ee72f77 100644 --- a/include/asterisk/res_pjsip.h +++ b/include/asterisk/res_pjsip.h @@ -415,6 +415,8 @@ struct ast_sip_endpoint_id_configuration { unsigned int send_pai; /*! Do we send Remote-Party-ID headers to this endpoint? */ unsigned int send_rpid; + /*! Do we send messages for connected line updates for unanswered incoming calls immediately to this endpoint? */ + unsigned int rpid_immediate; /*! Do we add Diversion headers to applicable outgoing requests/responses? */ unsigned int send_diversion; /*! When performing connected line update, which method should be used */ diff --git a/res/res_pjsip.c b/res/res_pjsip.c index 1d6af27a16..aa6a500cdc 100644 --- a/res/res_pjsip.c +++ b/res/res_pjsip.c @@ -199,7 +199,7 @@ This setting allows to choose the DTMF mode for endpoint communication. - DTMF is sent out of band of the main audio stream.This + DTMF is sent out of band of the main audio stream. This supercedes the older RFC-2833 used within the older chan_sip. @@ -316,6 +316,27 @@ Send the Remote-Party-ID header + + Immediately send connected line updates on unanswered incoming calls. + + When enabled, immediately send 180 Ringing + or 183 Progress response messages to the + caller if the connected line information is updated before + the call is answered. This can send a 180 Ringing + response before the call has even reached the far end. The + caller can start hearing ringback before the far end even gets + the call. Many phones tend to grab the first connected line + information and refuse to update the display if it changes. The + first information is not likely to be correct if the call + goes to an endpoint not under the control of this Asterisk + box. + When disabled, a connected line update must wait for + another reason to send a message with the connected line + information to the caller before the call is answered. You can + trigger the sending of the information by using an appropriate + dialplan application such as Ringing. + + Minimum session timers expiration period diff --git a/res/res_pjsip/pjsip_configuration.c b/res/res_pjsip/pjsip_configuration.c index ceb90a008e..5a4741d906 100644 --- a/res/res_pjsip/pjsip_configuration.c +++ b/res/res_pjsip/pjsip_configuration.c @@ -1710,6 +1710,7 @@ int ast_res_pjsip_initialize_configuration(const struct ast_module_info *ast_mod ast_sorcery_object_field_register(sip_sorcery, "endpoint", "trust_id_outbound", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, id.trust_outbound)); ast_sorcery_object_field_register(sip_sorcery, "endpoint", "send_pai", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, id.send_pai)); ast_sorcery_object_field_register(sip_sorcery, "endpoint", "send_rpid", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, id.send_rpid)); + ast_sorcery_object_field_register(sip_sorcery, "endpoint", "rpid_immediate", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, id.rpid_immediate)); ast_sorcery_object_field_register(sip_sorcery, "endpoint", "send_diversion", "yes", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, id.send_diversion)); ast_sorcery_object_field_register(sip_sorcery, "endpoint", "mailboxes", "", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_endpoint, subscription.mwi.mailboxes)); ast_sorcery_object_field_register(sip_sorcery, "endpoint", "aggregate_mwi", "yes", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, subscription.mwi.aggregate));