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Merged revisions 282740 via svnmerge from

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https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r282740 | twilson | 2010-08-18 21:18:50 -0500 (Wed, 18 Aug 2010) | 16 lines
  
  Merged revisions 282730 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
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    r282730 | twilson | 2010-08-18 21:14:28 -0500 (Wed, 18 Aug 2010) | 9 lines
    
    Merged revisions 282729 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
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      r282729 | twilson | 2010-08-18 21:12:55 -0500 (Wed, 18 Aug 2010) | 2 lines
      
      Add some documentation about codec negotiation to sip.conf
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282751 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Terry Wilson
2010-08-19 02:20:42 +00:00
parent 0e5b6069f4
commit 818bedf763
1 changed files with 12 additions and 0 deletions
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12
configs/sip.conf.sample
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@@ -255,6 +255,18 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Message-Account in the MWI notify message
; defaults to "asterisk"
; Codec negotiation
;
; When Asterisk is receiving a call, the codec will initially be set to the
; first codec in the allowed codecs defined for the user receiving the call
; that the caller also indicates that it supports. But, after the caller
; starts sending RTP, Asterisk will switch to using whatever codec the caller
; is sending.
;
; When Asterisk is placing a call, the codec used will be the first codec in
; the allowed codecs that the callee indicates that it supports. Asterisk will
; *not* switch to whatever codec the callee is sending.
;
;preferred_codec_only=yes ; Respond to a SIP invite with the single most preferred codec
; rather than advertising all joint codec capabilities. This
; limits the other side's codec choice to exactly what we prefer.