Merged revisions 274316 via svnmerge from

https://origsvn.digium.com/svn/asterisk/trunk

................
  r274316 | jpeeler | 2010-07-06 17:23:35 -0500 (Tue, 06 Jul 2010) | 14 lines
  
  Merged revisions 274283 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r274283 | jpeeler | 2010-07-06 17:15:21 -0500 (Tue, 06 Jul 2010) | 7 lines
    
    Correct sip.conf.sample comments for prematuremedia option.
    
    (closes issue #17513)
    Reported by: festr
    Patches: 
          patch uploaded by festr (license 443)
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@274347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.6.2
Jeff Peeler 16 years ago
parent 1f42454a44
commit 0257ac0bf3

@ -215,12 +215,14 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;relaxdtmf=yes ; Relax dtmf handling
;trustrpid = no ; If Remote-Party-ID should be trusted
;sendrpid = yes ; If Remote-Party-ID should be sent
;prematuremedia=no ; Some ISDN links send empty media frames before
; the call is in ringing or progress state. The SIP
; channel will then send 183 indicating early media
; which will be empty - thus users get no ring signal.
; Setting this to "no" will stop any media before we have
; call progress. Default is "yes".
;prematuremedia=no ; Some ISDN links send empty media frames before
; the call is in ringing or progress state. The SIP
; channel will then send 183 indicating early media
; which will be empty - thus users get no ring signal.
; Setting this to "yes" will stop any media before we have
; call progress (meaning the SIP channel will not send 183 Session
; Progress for early media). Default is "yes". Also make sure that
; the SIP peer is configured with progressinband=never.
;progressinband=never ; If we should generate in-band ringing always
; use 'never' to never use in-band signalling, even in cases

Loading…
Cancel
Save