[Asterisk-cvs] asterisk/configs sip.conf.sample,1.42,1.43

[markster at lists.digium.com]

Update of /usr/cvsroot/asterisk/configs
In directory mongoose.digium.com:/tmp/cvs-serv2259/configs

Modified Files:
	sip.conf.sample 
Log Message:
On second thought, add new "never" option for progressinband


Index: sip.conf.sample
===================================================================
RCS file: /usr/cvsroot/asterisk/configs/sip.conf.sample,v
retrieving revision 1.42
retrieving revision 1.43
diff -u -d -r1.42 -r1.43
--- sip.conf.sample	12 Nov 2004 03:57:39 -0000	1.42
+++ sip.conf.sample	14 Nov 2004 15:13:13 -0000	1.43
@@ -62,6 +62,8 @@
 				; 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
+				; 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
 ;nat=no				; NAT settings 
                                 ; yes = Always ignore info and assume NAT