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author | Jenkins2 <jenkins2@gerrit.asterisk.org> | 2017-05-11 16:33:55 -0500 |
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committer | Gerrit Code Review <gerrit2@gerrit.digium.api> | 2017-05-11 16:33:55 -0500 |
commit | 6383d9214aaba9a7ace1753ff0f6dfe1f2c6c5fa (patch) | |
tree | 855236668bd0e54f199d61510f3737174a2bda72 /CHANGES | |
parent | 3cfbb8b4818e4d8684232a0d1176a9b1bdcf3ca6 (diff) | |
parent | 10a49ab3627b7e93c01b22376cdf71015422894c (diff) |
Merge "res_hep_rtcp: Provide chan_sip Call-ID for RTCP messages." into 13
Diffstat (limited to 'CHANGES')
-rw-r--r-- | CHANGES | 6 |
1 files changed, 6 insertions, 0 deletions
@@ -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 ---------- ------------------------------------------------------------------------------ |