[asterisk-dev] Is there a way to terminate a sip channel when Iknow the CALL ID?

[Fernando Urzedo]

Hi Johansson,

I believe I found a bug already logged for this issue:

http://bugs.digium.com/view.php?id=12101

The symptoms and log are the same I found over here.

Today I have applied the suggested patch for this bug on a fresh 1.4.21.2 installation and so far so good. However, nobody has confirmed that this patch will be officially included in a official asterisk release. Maybe you can update us on this.

Thanks for your reply!
  

-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Johansson Olle E
Sent: terça-feira, 5 de agosto de 2008 12:52
To: Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] Is there a way to terminate a sip channel when Iknow the CALL ID?


4 aug 2008 kl. 18.12 skrev Fernando Urzedo:

>
> Hi,
>
> I use Asterisk 1.4.21.2 and queues. I add/remove/pause SIP peers to 
> this queue using AddQueueMember/RemoveQueueMember/PauseQueueMember.
>
> Today, I noticed that the status of one SIP peer that is receiving 
> calls from this queue became BUSY and, due to this fact, it is not 
> receiving calls anymore, neither from the queue, nor from any other 
> user in this Asterisk box. However, this guy is indeed available to 
> receive calls.
> This peer (using EyeBeam) has already logged off and logged in and its 
> status is still busy for Asterisk.
>
> If I run "SHOW CHANNELS", I cannot see any open channel related to 
> that peer. Howerver, if I run "SIP SHOW CHANNELS", I can see that 
> there is something stuck related to that peer:
>
> Peer              User/ANR     Call ID      Seq (Tx/Rx)  Format
> Hold     Last Message
> XXX.XXX.XXX.XXX   <peer name>  24d957d87a2  00102/00000  0x0 (nothing)
> No  (d)  Tx: ACK
>
> Looks like SIP messaging of the last call this peer answered/placed 
> got stuck and, as a result, Asterisk considers that this peer is 
> currently on a active call (busy).
>
> Please help me with two questions:
>
> - Why does this happen?
>
> - Is the a way (such as SOFT HANGUP command) to free this peer so that 
> is can receive calls again?


You seem to have found a bug. Please open a bug report, and attach all the required information. For this call, I would really like to see "sip history" output.

THanks.

/O