diff options
| -rw-r--r-- | CHANGES | 12 | ||||
| -rw-r--r-- | channels/chan_pjsip.c | 3 | ||||
| -rw-r--r-- | configs/samples/pjsip.conf.sample | 1 | ||||
| -rw-r--r-- | include/asterisk/res_pjsip.h | 2 | ||||
| -rw-r--r-- | res/res_pjsip.c | 23 | ||||
| -rw-r--r-- | res/res_pjsip/pjsip_configuration.c | 1 |
6 files changed, 40 insertions, 2 deletions
@@ -107,6 +107,18 @@ res_musiconhold 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 11acdffe1..8d9193f93 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 ba8bf751b..d3bb518f1 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 57c1a598d..442ee72f7 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 1d6af27a1..aa6a500cd 100644 --- a/res/res_pjsip.c +++ b/res/res_pjsip.c @@ -199,7 +199,7 @@ <para>This setting allows to choose the DTMF mode for endpoint communication.</para> <enumlist> <enum name="rfc4733"> - <para>DTMF is sent out of band of the main audio stream.This + <para>DTMF is sent out of band of the main audio stream. This supercedes the older <emphasis>RFC-2833</emphasis> used within the older <literal>chan_sip</literal>.</para> </enum> @@ -316,6 +316,27 @@ <configOption name="send_rpid" default="no"> <synopsis>Send the Remote-Party-ID header</synopsis> </configOption> + <configOption name="rpid_immediate" default="no"> + <synopsis>Immediately send connected line updates on unanswered incoming calls.</synopsis> + <description> + <para>When enabled, immediately send <emphasis>180 Ringing</emphasis> + or <emphasis>183 Progress</emphasis> response messages to the + caller if the connected line information is updated before + the call is answered. This can send a <emphasis>180 Ringing</emphasis> + 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.</para> + <para>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 <emphasis>Ringing</emphasis>.</para> + </description> + </configOption> <configOption name="timers_min_se" default="90"> <synopsis>Minimum session timers expiration period</synopsis> <description><para> diff --git a/res/res_pjsip/pjsip_configuration.c b/res/res_pjsip/pjsip_configuration.c index ceb90a008..5a4741d90 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)); |