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authorfile <file@f38db490-d61c-443f-a65b-d21fe96a405b>2009-04-06 16:15:30 +0000
committerfile <file@f38db490-d61c-443f-a65b-d21fe96a405b>2009-04-06 16:15:30 +0000
commitf3f7095a51c721febf03a0457a6df5f71ee0da7e (patch)
treecd16891b76adcf35c4f5f6fceb8c05d5dcdc42ac /doc/tex
parent85bd9cd2bf3f7ae30e5da1fb145708da4a9bcf63 (diff)
Add support for changing the outbound codec on a SIP call using
a dialplan variable. This adds a dialplan variable (SIP_CODEC_OUTBOUND) which controls the codec offered for an outgoing SIP call. This is much like the SIP_CODEC dialplan variable and has the same restrictions. The codec set must be one that is configured for the call. (closes issue #13243) Reported by: samdell3 Patches: 13243.diff uploaded by file (license 11) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@186624 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'doc/tex')
-rw-r--r--doc/tex/channelvariables.tex4
1 files changed, 3 insertions, 1 deletions
diff --git a/doc/tex/channelvariables.tex b/doc/tex/channelvariables.tex
index 4c3195246..4c7b4d5ff 100644
--- a/doc/tex/channelvariables.tex
+++ b/doc/tex/channelvariables.tex
@@ -925,7 +925,9 @@ ${SIPDOMAIN} * SIP destination domain of an inbound call (if appropriate
${SIPFROMDOMAIN} Set SIP domain on outbound calls
${SIPUSERAGENT} * SIP user agent (deprecated)
${SIPURI} * SIP uri
-${SIP_CODEC} Set the SIP codec for a call
+${SIP_CODEC} Set the SIP codec for an inbound call
+${SIP_CODEC_INBOUND} Set the SIP codec for an inbound call
+${SIP_CODEC_OUTBOUND} Set the SIP codec for an outbound call
${SIP_URI_OPTIONS} * additional options to add to the URI for an outgoing call
${RTPAUDIOQOS} RTCP QoS report for the audio of this call
${RTPVIDEOQOS} RTCP QoS report for the video of this call