[Asterisk-Users] SIP app_queue

[Brian West]

I have figured out that its a problem in app_queue, could be the
interaction between chan_sip and app_queue. Or the ATA is on crack.

in chan_sip if I change

case 501: /* Not Implemented */
if (owner)
	ast_queue_control(p->owner, AST_CONTROL_CONGESTION, 0);
break;

to:

case 501: /* Not Implemented */
if (owner)
	ast_queue_control(p->owner, AST_CONTROL_BUSY, 0);
break;

The problem only happens on one channel of the ATA.  The orginal channel
that was rang and not answered is stuck in circuit-busy.  If I take the
phone off hook and put it back while its not ringing it will ring the
next time that extension comes around.  So is it a bug with the ATA? Or
app_queue or the interaction of app_queue and chan_sip

ATA Version: v2.15 ata18x (Build 020927a)


bkw



On Sat, 2 Aug 2003, Brian West wrote:

> I noticed a few issues with app_queue just wanted to know if its sip
> related or ata186 related:
>
> Ext 111 and Ext 112 are dynamically loged into the queue via
> AddQueueMember.
>
> Call hits queue with fewestcalls routing.
>
> Rings ext 111 if 111 doesn't answer.  It rings ext 112.  If for some
> reason ext 112 doesn't answer it rings back to 111.  Again at this point
> ext 111 isn't answered it rings 112 again.  Now here is where the trouble
> starts.  If 112 isn't answered this round it goes back to 111.
>
> 111 returns circuit-busy promptly goes to 112 returns circuit-busy .. and
> at this point both phones are ringing.  You can pick them up but it won't
> un-queue the caller.  If you take both phones off hook and back on the
> problem starts from square one.  Granted someone should be there to answer
> the phone but if for any reason they don't I would like some options:
>
> 1. either loop till someone answers it or
> 2. un-queue to voicemail.
>
> Also these ext 111 and 112 are on the same ATA186 that might be an issue
> with SIP but wanted to see if anyone else had this problem.
>
> Thanks,
> Brian
>
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