diff options
| author | Joshua Colp <jcolp@digium.com> | 2017-05-09 10:25:29 +0000 |
|---|---|---|
| committer | Joshua Colp <jcolp@digium.com> | 2017-05-09 05:38:59 -0500 |
| commit | 3c36c29c81c07ad220860a59349def72afc35d86 (patch) | |
| tree | baa4e18bdd28bb75e54b6436a2989da17eb2f38e /CHANGES | |
| parent | 3dae4279be4f61f16473d3e5ea9d45d1f8462eda (diff) | |
res_hep_rtcp: Provide chan_sip Call-ID for RTCP messages.
This change adds the required logic to allow the SIP
Call-ID to be placed into the HEP RTCP traffic if the
chan_sip module is used. In cases where the option is
enabled but the channel is not either SIP or PJSIP then
the code will fallback to the channel name as done
previously.
Based on the change on Nir's branch at:
team/nirs/hep-chan-sip-support
ASTERISK-26427
Change-Id: I09ffa5f6e2fdfd99ee999650ba4e0a7aad6dc40d
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 14.3.0 to Asterisk 14.4.0 ------------ ------------------------------------------------------------------------------ |