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authorJenkins2 <jenkins2@gerrit.asterisk.org>2017-05-11 16:39:54 -0500
committerGerrit Code Review <gerrit2@gerrit.digium.api>2017-05-11 16:39:54 -0500
commit57217e4cc20ff17e8ea736ee19b03b264b166900 (patch)
tree57cf6a2322250aa3eec78528d02a91a14a03db64 /CHANGES
parent542dd7d7950c3d39c98fb80667f053f8cd7bb01f (diff)
parent3c36c29c81c07ad220860a59349def72afc35d86 (diff)
Merge "res_hep_rtcp: Provide chan_sip Call-ID for RTCP messages."
Diffstat (limited to 'CHANGES')
-rw-r--r--CHANGES6
1 files changed, 6 insertions, 0 deletions
diff --git a/CHANGES b/CHANGES
index 7723ab3eb..fcbe04c94 100644
--- a/CHANGES
+++ b/CHANGES
@@ -51,6 +51,12 @@ 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 14.3.0 to Asterisk 14.4.0 ------------
------------------------------------------------------------------------------