[Asterisk-cvs] asterisk/configs sip.conf.sample,1.33,1.34

[markster at lists.digium.com]

Update of /usr/cvsroot/asterisk/configs
In directory mongoose.digium.com:/tmp/cvs-serv19854/configs

Modified Files:
	sip.conf.sample 
Log Message:
Add option for in-band progress (bug #1879)


Index: sip.conf.sample
===================================================================
RCS file: /usr/cvsroot/asterisk/configs/sip.conf.sample,v
retrieving revision 1.33
retrieving revision 1.34
diff -u -d -r1.33 -r1.34
--- sip.conf.sample	21 Jun 2004 06:11:56 -0000	1.33
+++ sip.conf.sample	26 Jun 2004 21:17:12 -0000	1.34
@@ -58,6 +58,7 @@
 ;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
 ;useragent=Asterisk PBX		; Allows you to change the user agent string
 ;promiscredir = no      ; If yes, allows 302 or REDIR to non-local SIP address
 ; Asterisk can register as a SIP user agent to a SIP proxy (provider)
@@ -123,6 +124,7 @@
 ; disallow                    disallow
 ; insecure                    insecure
 ; trustrpid                   trustrpid
+; progressinband              progressinband
 ; promiscredir                promiscredir
 ; callerid
 ; accountcode