[Asterisk-Dev] SIP channels not cleared

[Chee Foong]

Hello,

I also found out that asterisk does not clear up the channel if it recieve a
BYE message when the call is not connected. Which means at any stages from
the INVITE to Ringing. For example, if the other end send Ringing to
asterisk, after a short while it send a BYE for some reason to terminate the
call to asterisk, asterisk response an OK. But asterisk just does not clear
the SIP channel, it still show in "sip show channels".

However there is no problem for asterisk to receive a BYE if the call is
answered at the other end.

I have check with the other end's provider. They insists that asterisk
should be able to accept BYE at any stage in the sip message exchange. Is
this really true?

Please help :)
Currently I have 2500++ sip channels hanging and increasing.

Thanks

CCF



-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com]On Behalf Of Olle E.
Johansson
Sent: Wednesday, August 17, 2005 16:50
To: Asterisk Developers Mailing List
Subject: Re: [Asterisk-Dev] SIP channels not cleared


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
_______________________________________________
Asterisk-Dev mailing list
Asterisk-Dev at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-dev
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev