Have you tried canreinvite=yes in the sip.conf ? This is what gurus suggested to me when I had similar issues. But that did not work for me. May be it might work for you.
- SamW -----Original Message----- From: Olle E. Johansson [mailto:[EMAIL PROTECTED] Sent: Monday, January 12, 2004 9:51 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIP redirect /New subject/ > If a call comes to Server1 by SIP, is it possible to re-direct client to > Server2. In another words, IAX2 part is (taken out), so client > communicates with Server2 by SIP directly during that call. My primary > motivation behind this is to save on resources. How would you configure this? Always redirect some extensions or only if something happens? Please explain a bit more. I don't think it's possible today. Requires some additions. /O _______________________________________________ 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 _______________________________________________ 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
