[Asterisk-Users] SIP app_queue

[Mark Spencer]

Sounds like it's not cancelling right... Post to the bug tracker and be
sure to include SIP debug (and maybe some commentary about what's going on
in each situation).

Mark

On Sat, 2 Aug 2003, Brian West wrote:

> 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
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>