[Asterisk-Users] SIP redirect /New subject/

[SamW]

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:oej at edvina.net] 
Sent: Monday, January 12, 2004 9:51 AM
To: asterisk-users at lists.digium.com
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
Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users