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authorJoshua Colp <jcolp@digium.com>2017-05-09 10:25:29 +0000
committerJoshua Colp <jcolp@digium.com>2017-05-09 05:38:59 -0500
commit3c36c29c81c07ad220860a59349def72afc35d86 (patch)
treebaa4e18bdd28bb75e54b6436a2989da17eb2f38e /configs
parent3dae4279be4f61f16473d3e5ea9d45d1f8462eda (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 'configs')
-rw-r--r--configs/samples/hep.conf.sample6
1 files changed, 5 insertions, 1 deletions
diff --git a/configs/samples/hep.conf.sample b/configs/samples/hep.conf.sample
index 3d1e74139..32bd8df39 100644
--- a/configs/samples/hep.conf.sample
+++ b/configs/samples/hep.conf.sample
@@ -24,5 +24,9 @@ capture_id = 1234 ; A unique integer identifier for this
; with each packet from this server.
uuid_type = call-id ; Specify the preferred source for the Homer
; correlation UUID. Valid options are:
- ; - 'call-id' for the PJSIP SIP Call-ID
+ ; - 'call-id' for the PJSIP or chan_sip SIP
+ ; Call-ID
; - 'channel' for the Asterisk channel name
+ ; Note: If 'call-id' is specified but the
+ ; channel is not PJSIP or chan_sip then the
+ ; Asterisk channel name will be used instead.