Merge "channels/pjsip/dialplan_functions: Add an option for extracting the SIP call-id" into 13

CHANGES

@@ -8,6 +8,17 @@
 ===
 ==============================================================================
 
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 13.5.0 to Asterisk 13.6.0 ------------
+------------------------------------------------------------------------------
+
+Dialplan Functions
+------------------
+ * The CHANNEL function, when used on a PJSIP channel, now exposes a 'call-id'
+   extraction option when using with the 'pjsip' signalling option. It will
+   return the SIP Call-ID associated with the INVITE request that established
+   the PJSIP channel.
+
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 13.4.0 to Asterisk 13.5.0 ------------
 ------------------------------------------------------------------------------

channels/pjsip/dialplan_functions.c

@@ -318,6 +318,9 @@
 				<literal>type</literal> parameter must be provided. It specifies
 				which signalling parameter to read.</para>
 				<enumlist>
+					<enum name="call-id">
+						<para>The SIP call-id.</para>
+					</enum>
 					<enum name="secure">
 						<para>Whether or not the signalling uses a secure transport.</para>
 						<enumlist>
@@ -594,6 +597,8 @@ static int channel_read_pjsip(struct ast_channel *chan, const char *type, const
 	if (ast_strlen_zero(type)) {
 		ast_log(LOG_WARNING, "You must supply a type field for 'pjsip' information\n");
 		return -1;
+	} else if (!strcmp(type, "call-id")) {
+		snprintf(buf, buflen, "%.*s", (int) pj_strlen(&dlg->call_id->id), pj_strbuf(&dlg->call_id->id));
 	} else if (!strcmp(type, "secure")) {
 #ifdef HAVE_PJSIP_GET_DEST_INFO
 		pjsip_host_info dest;