[Asterisk-Dev] SIP channels not cleared

[Olle E. Johansson]

Chee Foong Chiew wrote:
> 
> I have made a sip trace on asterisk and seems like
> they all share a same SIP message flow. When asterisk
> send an INVITE to other sip server say B. B will reply
> with  Trying. When B found out that the actual
> destination can not be reached, it sends a BYE to
> asterisk. Asterisk then reply with a 200 OK. Call is
> hangup succesfully but 'sip show channels' still list
> the call record and never go away untill asterisk is
> restart. See below:
> 

The other end is faulty indeed, it should *never* send BYE when we have
no call. To abort an INVITE, you send CANCEL. To deny an INVITE you send
an error message, like 404 Not Found.

Asterisk should be able to handle these errors more gracefully, but from
the small debug output you did send this is a seriously bad and poor SIP
implementation.

Regards,
/Olle