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-rw-r--r--CHANGES11
-rw-r--r--channels/pjsip/dialplan_functions.c5
2 files changed, 16 insertions, 0 deletions
diff --git a/CHANGES b/CHANGES
index ec5c8a9ca..c24919517 100644
--- a/CHANGES
+++ b/CHANGES
@@ -9,6 +9,17 @@
==============================================================================
------------------------------------------------------------------------------
+--- 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 ------------
------------------------------------------------------------------------------
diff --git a/channels/pjsip/dialplan_functions.c b/channels/pjsip/dialplan_functions.c
index 9c5245648..eb6def60f 100644
--- a/channels/pjsip/dialplan_functions.c
+++ b/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;