[Asterisk-cvs] asterisk/configs sip.conf.sample,1.49,1.50

[markster at lists.digium.com]

Update of /usr/cvsroot/asterisk/configs
In directory mongoose.digium.com:/tmp/cvs-serv1365/configs

Modified Files:
	sip.conf.sample 
Log Message:
Fix example to reflect that "never" is the default value for progressinband (bug #3129)


Index: sip.conf.sample
===================================================================
RCS file: /usr/cvsroot/asterisk/configs/sip.conf.sample,v
retrieving revision 1.49
retrieving revision 1.50
diff -u -d -r1.49 -r1.50
--- sip.conf.sample	28 Dec 2004 21:20:18 -0000	1.49
+++ sip.conf.sample	31 Dec 2004 15:38:14 -0000	1.50
@@ -61,7 +61,7 @@
 ;rtpholdtimeout=300		; Terminate call if 300 seconds of no RTP activity
 				; when we're on hold (must be > rtptimeout)
 ;trustrpid = no			; If Remote-Party-ID should be trusted
-;progressinband=no		; If we should generate in-band ringing always
+;progressinband=never		; If we should generate in-band ringing always
 				; use 'never' to never use in-band signalling, even in cases
 				; where some buggy devices might not render it
 ;useragent=Asterisk PBX		; Allows you to change the user agent string