[Asterisk-cvs] asterisk/configs sip.conf.sample,1.45,1.46

[markster at lists.digium.com]

Update of /usr/cvsroot/asterisk/configs
In directory mongoose.digium.com:/tmp/cvs-serv17652/configs

Modified Files:
	sip.conf.sample 
Log Message:
Add user=phone option (bug #2244, thanks oej)


Index: sip.conf.sample
===================================================================
RCS file: /usr/cvsroot/asterisk/configs/sip.conf.sample,v
retrieving revision 1.45
retrieving revision 1.46
diff -u -d -r1.45 -r1.46
--- sip.conf.sample	28 Nov 2004 21:49:07 -0000	1.45
+++ sip.conf.sample	2 Dec 2004 23:29:25 -0000	1.46
@@ -74,6 +74,8 @@
 ;promiscredir = no      	; If yes, allows 302 or REDIR to non-local SIP address
 	                       	; Note that promiscredir when redirects are made to the
        	                	; local system will cause loops since SIP is incapable
+;usereqphone = no		; If yes, ";user=phone" is added to uri that contains
+				; a valid phone number
        	                	; of performing a "hairpin" call.
 ;dtmfmode = rfc2833		; Set default dtmfmode for sending DTMF. Default: rfc2833
 				; Other options: 
@@ -189,6 +191,7 @@
 ;username=yourusername		; Authentication user for outbound proxies
 ;fromuser=yourusername		; Many SIP providers require this!
 ;host=box.provider.com
+;usereqphone=yes		; This provider requires ";user=phone" on URI
 
 ;[grandstream1]
 ;type=friend 			; either "friend" (peer+user), "peer" or "user"