[svn-commits] jpeeler: branch 1.4 r274283 - /branches/1.4/configs/sip.conf.sample

[SVN commits to the Digium repositories]

Author: jpeeler
Date: Tue Jul  6 17:15:21 2010
New Revision: 274283

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=274283
Log:
Correct sip.conf.sample comments for prematuremedia option.

(closes issue #17513)
Reported by: festr
Patches: 
      patch uploaded by festr (license 443)

Modified:
    branches/1.4/configs/sip.conf.sample

Modified: branches/1.4/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/branches/1.4/configs/sip.conf.sample?view=diff&rev=274283&r1=274282&r2=274283
==============================================================================
--- branches/1.4/configs/sip.conf.sample (original)
+++ branches/1.4/configs/sip.conf.sample Tue Jul  6 17:15:21 2010
@@ -119,12 +119,14 @@
                                  ; use 'never' to never use in-band signalling, even in cases
                                  ; where some buggy devices might not render it
                                  ; Valid values: yes, no, never Default: never
-;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 "no". Also make sure that
+                                 ; the SIP peer is configured with progressinband=never. 
 ;useragent=Asterisk PBX          ; Allows you to change the user agent string
 ;promiscredir = no               ; If yes, allows 302 or REDIR to non-local SIP address
                                  ; Note that promiscredir when redirects are made to the