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

[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?

Thanks in advance!