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authorqwell <qwell@f38db490-d61c-443f-a65b-d21fe96a405b>2006-09-20 17:39:59 +0000
committerqwell <qwell@f38db490-d61c-443f-a65b-d21fe96a405b>2006-09-20 17:39:59 +0000
commit62752745f50399224e0d7c9eec77a3879d584683 (patch)
treeb7cb886010c20f183b7a7f50f1aaf2aa5ce1bf59 /configs/sip.conf.sample
parent3c35e5e958ccff5a307542e84ae3916a3dc2ae06 (diff)
Add documentation on rtp packetization.
Technically, ooh323 doesn't support it yet, but there is a patch that should be committed very soon. Issue #7989, patch by DEA, slightly modified. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43344 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'configs/sip.conf.sample')
-rw-r--r--configs/sip.conf.sample2
1 files changed, 1 insertions, 1 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index f931780b9..07389e16b 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -70,7 +70,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; defaults to "asterisk"
;disallow=all ; First disallow all codecs
;allow=ulaw ; Allow codecs in order of preference
-;allow=ilbc ;
+;allow=ilbc ; see doc/rtp-packetization for framing options
;
; This option specifies a preference for which music on hold class this channel
; should listen to when put on hold if the music class has not been set on the