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

[Johansson Olle E]

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

-------------- next part --------------
A non-text attachment was scrubbed...
Name: smime.p7s
Type: application/pkcs7-signature
Size: 2207 bytes
Desc: not available
Url : http://lists.digium.com/pipermail/asterisk-dev/attachments/20080805/db54d084/attachment.bin