Merged revisions 274316 via svnmerge from

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

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  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

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    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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@274347 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 0b099a496eaed5bb6cb975ab0b5cab333cd02068..e47d96f8271468e290ace9c5723ce05224a4e358 100644 (file)
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -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