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authorJenkins2 <jenkins2@gerrit.asterisk.org>2017-05-11 16:33:55 -0500
committerGerrit Code Review <gerrit2@gerrit.digium.api>2017-05-11 16:33:55 -0500
commit6383d9214aaba9a7ace1753ff0f6dfe1f2c6c5fa (patch)
tree855236668bd0e54f199d61510f3737174a2bda72 /CHANGES
parent3cfbb8b4818e4d8684232a0d1176a9b1bdcf3ca6 (diff)
parent10a49ab3627b7e93c01b22376cdf71015422894c (diff)
Merge "res_hep_rtcp: Provide chan_sip Call-ID for RTCP messages." into 13
Diffstat (limited to 'CHANGES')
-rw-r--r--CHANGES6
1 files changed, 6 insertions, 0 deletions
diff --git a/CHANGES b/CHANGES
index c26b0485b..7ae196c9f 100644
--- a/CHANGES
+++ b/CHANGES
@@ -42,6 +42,12 @@ res_pjsip_config_wizard
endpoint/outbound_proxy, aor/outbound_proxy and registration/outbound_proxy
parameters.
+res_hep_rtcp
+------------------
+ * If the 'call-id' value is specified for the uuid_type option and a
+ chan_sip channel is used the resulting HEP traffic will now contain the
+ SIP Call-ID instead of the Asterisk channel name.
+
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.14.0 to Asterisk 13.15.0 ----------
------------------------------------------------------------------------------