I have converted one of the unlimitel trunks to SIP and the trunk is now operating.
Do you have any idea of how to test if passing the call to a remote SIP Softphone client is working? Is there a test or something I can do to verify this is happening now? Or does it just "happen" automatically? Regards, Chuck -----Original Message----- From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 08, 2006 2:05 PM To: asterisk@uc.org Subject: Re: [on-asterisk] Passing to Softphone Extension from Asterisk On Tuesday 08 August 2006 13:23, Chuck Mariotti wrote: > I am using unlimitel (IAX2 Trunk). ... > Extensions are configured as SIP extensions and we are currently using > X-Lite softphones. > internet, to her softphone). How would I make it so that the extension > talks straight to unlimitel and not through Asterisk? Not going to happen. You have IAX2 on one side and SIP on the other; these two cannot talk to each other without something inbetween which can translate. This is much the same as one leg of a call using ulaw and the other leg using g729; the two will never be able to communicate directly unless they have a common codec. Similarly, your two endpoints will never communicate directly unless they have a common transport. Either use SIP with Unlimitel or get an Asterisk box at the remote office. -A. --------------------------------------------------------------------- To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]