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/1.4@274283 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 10d401883bcabb60f770de8bb3001cb7a4de5dbd..9a420f76863c9731c20ba88c9ecee156c931188f 100644 (file)
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -119,12 +119,14 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
                                  ; 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