asterisk users - Jun 2006 - SPA-2002 call HANGUP. May be a SIP bug.

Hello,
We have problems with Asterisk and Sipura SPA-2002. 
SPA is behind the NAT. Asterisk has nat=yes.

Sometimes call doesn't hangup when user finish the call and hangup the 
headset.

In this case during all conversation SIP packets contains 
Call-ID: bee522ee-8efa7d25@123.123.123.123 
but the final BYE packet from adapter contains 
Call-ID: bee522ee-8efa7d25@10.0.0.2
        Is such scenario correct from SIP protocol point of view?

After receiving such packet Asterisk trying to destroy:
Destroying call 'bee522ee-8efa7d25@10.0.0.2'
But I suppose that such call doesn't exists and call remains active.

Thanks,
Dmitry.

----- Dmytro Mishchenko <dioml@dreamtime.net>
wrote:
> In this case during all conversation SIP packets contains 
> Call-ID: bee522ee-8efa7d25@123.123.123.123 
> but the final BYE packet from adapter contains 
> Call-ID: bee522ee-8efa7d25@10.0.0.2
>         Is such scenario correct from SIP protocol point of view?
No, it is not valid. The Call-ID is used to uniquely identify the SIP dialog,
and must remain the same.

-- 
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.