From: "Girish Gopinath" <[EMAIL PROTECTED]> Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] H323 - SIP Interoperability Date: Thu, 01 Apr 2004 22:46:10 +0530
Hello,
From: pesb <[EMAIL PROTECTED]> Subject: [Asterisk-Users] H323 - SIP Interoperability Date: Thu, 1 Apr 2004 12:37:17 -0300
<snip>So, I would like to call SIP/4 phone by dialing 014. Something like this:
exten => 01X,1,Dial(SIP/X) ; This is not working
How can I do that?
Try this: exten => _01X,1,Dial(SIP/${EXTEN:2}) That should do it.
Another question: How can I make the RTP data flow go directly from one IP phone to the other? Rigth now, all the RTP data flow goes through the SIP proxy.
set canreinvite=yes for sip users in sip.conf
Regards, Girish
_________________________________________________________________
Easiest Money Transfer to India. Send Money To 6000 Indian Towns. http://go.msnserver.com/IN/42198.asp Easiest Way To Send Money Home!
_______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
_________________________________________________________________
News, views and gossip. Hot downloads �n previews. http://www.msn.co.in/Cinema/ Get it all at MSN Cinema!
_______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
