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.

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