[asterisk-users] Queue not sending call to Agent

[Duane Larson]

After a Good Call from a PSTN phone if I do a "sip prune realtime peer
9013XX9XX8" (9013XX9XX8 being the phone number of the Agent/Member) then I
can call the number again and not get the issue.  So this has something to
do with the stuff that is put in my peer table after a call.


Any ideas?

On Tue, Jun 14, 2011 at 5:18 PM, Duane Larson <duane.larson at gmail.com>wrote:

> One more piece to add.  I had mentioned before that I could get a call from
> a PSTN user to work the first time.  So here is all the output of a Good
> call from a PSTN user after I have performed a "RELOAD" on asterisks CLI
>
> http://pastebin.com/9RSvQsmN
>
> And when the caller or agent hangs this call up all calls from the PSTN
> afterward get put in the queue automatically and the agent never gets
> called.
>
>   On Tue, Jun 14, 2011 at 4:37 PM, Duane Larson <duane.larson at gmail.com>wrote:
>
>> 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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>>
>>
>>
>> --
>> --
>> *--*--*--*--*--*
>> Duane
>> *--*--*--*--*--*
>> --
>>
>
>
>
> --
> --
> *--*--*--*--*--*
> Duane
> *--*--*--*--*--*
> --
>



-- 
--
*--*--*--*--*--*
Duane
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