[asterisk-users] Queue not sending call to Agent

[Duane Larson]

Ok.  Something isn't right.  With a user that is local to my SIP user
database calls the queue phone number everything works without issue.  It is
when a remote user (like someone from the PSTN) calls the queue phone number
that the caller gets put into the queue and the agent/member doesn't receive
the call.  I have captured debugs from OpenSIPS and Asterisk and I can't
really see any difference.  I also executed the commands you told me where I
could.  Here are the debugs

Good call from local SIP user to Queue
LocalUser -> OpenSIPSProxy -> Asterisk (then asterisk calls the
agent/member) -> OpenSIPSProxy -> Agent
http://pastebin.com/Fa9y3CXQ



Bad call from PSTN Caller to Queue
PSTN Gatway -> OpenSIPSB2BUA -> OpenSIPSProxy -> Asterisk (then asterisk
doesn't call Agent/Member for some reason)
http://pastebin.com/VBA9nGAs


Thanks for looking at this.  Currently this happens every time.  Any call
from a local user gets put in queue and agent is called right away, but any
call from PSTN user gets put in queue and agent isn't called but the agent
shows as

Asterisk18*CLI> queue show
irock.com has 1 calls (max unlimited) in 'ringall' strategy (6s holdtime,
131s talktime), W:0, C:12, A:15, SL:0.0% within 0s
   Members:
      SIP/9013XX9XX8 with penalty 10 (dynamic) (Busy) has taken 12 calls
(last was 1991 secs ago)
   Callers:
      1. SIP/9013XX9XX8-0000002d (wait: 0:02, prio: 0)

When it is a good call and I do "queue show" I see this
Asterisk18*CLI> queue show
irock.com has 0 calls (max unlimited) in 'ringall' strategy (5s holdtime,
131s talktime), W:0, C:12, A:16, SL:0.0% within 0s
   Members:
      SIP/9013XX9XX8 with penalty 10 (dynamic) (Busy) has taken 12 calls
(last was 2079 secs ago)
   No Callers

*How come with the Bad Call the Agent/Member shows up in a "queue show" as
being a Member and a Caller???*



On Mon, Jun 13, 2011 at 11:58 PM, Satish Barot <satish4asterisk at gmail.com>wrote:

>
> I am not sure but seems like Agent channel not being released from
> Asterisk.
>
> Next time when this happens, try 'core show channels' to check whether
> Agent channel is released or not.
>
> [SATISH]
>
>
> On Mon, Jun 13, 2011 at 9:12 PM, Duane Larson <duane.larson at gmail.com>wrote:
>
>> Yesterday I rebooted the server and it seems to be working again.  Not
>> sure what the reboot might have changed.  Hopefully it doesn't happen again
>> but I can't be sure.  To answer your question I have the sip.conf in my
>> mysql database and in MySQL I have callcounter set to yes.  I don't have a
>> column of 'qualify' in my database for the sip users.  For my config I am
>> using OpenSIPS as the register and proxy.  Asterisk is only used for
>> voicemail and ACD/Hunt groups.
>>
>>
>> On Mon, Jun 13, 2011 at 12:53 AM, Satish Barot <satish4asterisk at gmail.com
>> > wrote:
>>
>>>
>>> Provide the entry for Agent SIP/9013XX9XX8 along with parameters
>>> 'callcounter' and 'qualify' from sip.conf.
>>>
>>> Also provide CLI outputs of 'core show channels',sip show peers' and
>>> 'queue show' when...
>>>
>>> (1)First caller enters the Queue
>>> (2)First caller gets connected with Agent
>>> (3)First caller gets disconnected from Agent
>>> (4)Second caller enters the Queue
>>>
>>> You may have sequences changed for step no 3 and 4 in your scenario.
>>>
>>>
>>> [SATISH]
>>
>>
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-- 
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Duane
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