[asterisk-users] Asterisk 1.4 & Polycom buddy status

Olle E Johansson oej at edvina.net
Fri Jan 26 10:09:00 MST 2007


26 jan 2007 kl. 16.31 skrev James Fromm:

> Olle E Johansson wrote:
>> 24 jan 2007 kl. 18.10 skrev Eric "ManxPower" Wieling:
>>> James Fromm wrote:
>>>
>>>> The behavior we see is that the SIP interface in the queue will  
>>>> sometimes not release from the in-use state.  Connecting to the  
>>>> interface from another SIP device and immediately hanging up  
>>>> will clear the state.
>>>> The phones in question are configured with one line that will  
>>>> except only one call.  The device itself does not think it is in- 
>>>> use because it will accept another call.  Something in the SIP  
>>>> channel driver is not clearing the state when a call is completed.
>>>> There is definitely no correlation between this and Asterisk  
>>>> restarting.  In fact, if a device is 'stuck' on in-use,  
>>>> restarting Asterisk will clear the state.
>>>> I've been working on this for a week now.  It only started for  
>>>> us because I just implemented the call-limit option in the  
>>>> sip.conf in Asterisk for the devices.  See my posts with subject  
>>>> 'Queue and Interface time out'.
>>>
>>> I believe there is/was a bug relating to call-limit.  Buddy Watch  
>>> doesn't work if you use call-limit and if a call from a queue is  
>>> transfered, the call-limit is not released until the original  
>>> call is terminated.  I do not know if these issues have been  
>>> fixed or not.
>> Again, a relation to call transfer. I think the bug is that we  
>> don't handle call-limits properly during a call transfer. That needs
>> to be verified and fixed.
>
> There may be, but transfers are not the cause of the issue I  
> describe. SIP interfaces that are members of a Queue, will  
> erratically not be released from 'in-use' when a call is  
> completed.  I have tested with both caller terminated and agent  
> terminated calls and both will cause this behavior.  It happens on  
> approximately 20% of all calls the queue members receive.  Dialing  
> the SIP device with another device will immediately free the status.
>
> I wonder if this only happens on calls sent to the SIP device by  
> the Queue application.  I will test that today.

If you are using chan_agent as a proxy channel, check if that changes  
things.

/O


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