[asterisk-users] Asterisk in Production ?

[Benoit Plessis]

Steve Totaro a écrit :
> On Tue, May 6, 2008 at 9:18 AM, Benoit Plessis <benoit at plessis.info> wrote:
>   
>> lordfuknowsyou a écrit :
>>
>>     
>>> Vinícius Fontes wrote:
>>>       
>>  >
>>  > I use 1.4.18 with no problems. We have quite a few users(125 total
>>  > between branches), but the call volume at the most has been around 15
>>  > active calls at a time.
>>  >
>>  Any IAX2 phone or mostly SIP ?
>>  Do you use Call Queues ?
>>
>>  We have less user than that, less concurrent call but quite a few
>>  crash/deadlocks
>>
>>     
>
> Try SIP only if you can and report back.  I think you will confirm
> what is pretty much a silent consensus (even among Digium Devs).
>
> Thanks,
> Steve Totaro
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>   

I've tried SIP only but i already got one 'stuck' Queue member:
   Members:
      Local/136 at queues with penalty 10 (dynamic) (In use) has taken 1 
calls (last was 45 secs ago)
      Local/888 at queues with penalty 20 (dynamic) (Not in use) has taken 
no calls yet
   Callers:
      1. Zap/10-1 (wait: 0:18, prio: 0)

[May  6 17:48:35] NOTICE[2047]: app_queue.c:2152 wait_for_answer: No one 
is answering queue 'support' (1/0/0)
asterix*CLI> core show channels
Channel              Location             State   
Application(Data)            
SIP/rtournier-081ef2 (None)               Up      Bridged 
Call(Local/136 at queues-

but the other end of the bridged call is long gone