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authorMatt Jordan <mjordan@digium.com>2015-09-05 14:58:41 -0500
committerMatt Jordan <mjordan@digium.com>2015-09-05 15:22:35 -0500
commit78d0b9d97ecf034468e252440217dd4bc371ef71 (patch)
treea2a2818ca849e37e77b62ef227377ff393b9ca2e /CHANGES
parentd238cf33a9bfe86f9ebdb204384df820098c2670 (diff)
channels/pjsip/dialplan_functions: Add an option for extracting the SIP call-id
This patch adds a new option to the CHANNEL function that allows for the extraction of the SIP call-id. It is used in conjunction with the 'pjsip' option, and will return the Call-ID of the INVITE request that established the PJSIP channel. ASTERISK-25352 Change-Id: I278d1f8bcfe3a53c5aa1dadebc14e92b0abd476a
Diffstat (limited to 'CHANGES')
-rw-r--r--CHANGES11
1 files changed, 11 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 ------------
------------------------------------------------------------------------------