From 8bd82ebc0d6a57cc71f6e5207dfb0925bf506449 Mon Sep 17 00:00:00 2001 From: Jason Parker Date: Wed, 20 Sep 2006 17:39:59 +0000 Subject: 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: https://origsvn.digium.com/svn/asterisk/trunk@43344 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- configs/sip.conf.sample | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'configs/sip.conf.sample') 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 -- cgit v1.2.3