I don't believe Unlimitel supports reinvite's yet. If it does though,
the 'reinvite' option needs to be set to 'yes' for both the trunk and
the client in sip.conf

If you look at the network traffic while on the asterisk box, you
shouldn't see much of anything if you do a tcpdump something like this
as root:
# tcpdump -i <your external network interface, likely eth0> not port
5060 and host <ip address of client>


On 8/9/06, Chuck Mariotti <[EMAIL PROTECTED]> wrote:
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: [email protected]
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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