[OpenSIPS-Users] opensips 2.1 call_center queue position

[Jonathan Hunter]

Hi Bogdan,


That seems to be better, certainly from the count incrementing,  however  a couple of issues, I am now seeing this error, and when first call hits opensips it doesnt now route to the Agent who is logged in free and available;


: ERROR:b2b_entities:b2b_send_request: Can not send request [INVITE] for entity type [0] for dlg[0x7fe3f348a0f8]->[B2B.273.29] in terminated state
Dec 23 15:49:22 HPBXProxy1-beta /usr/local/sbin/opensips[12140]: ERROR:b2b_logic:b2bl_bridge: Failed to send INVITE request
Dec 23 15:49:22 HPBXProxy1-beta /usr/local/sbin/opensips[12140]: ERROR:call_center:set_call_leg: bridging failed
Dec 23 15:49:22 HPBXProxy1-beta /usr/local/sbin/opensips[12140]: ERROR:call_center:b2bl_callback_customer: failed to set new destination for call
Dec 23 15:49:22 HPBXProxy1-beta /usr/local/sbin/opensips[12140]: ERROR:b2b_logic:b2b_logic_notify_request: The callback function was unsuccessful




Also the first member in the queue is given value 0, and the second is then given value 1, can it start at 1 or is there a reason behind it?



Thanks


Jon


________________________________
From: Bogdan-Andrei Iancu <bogdan at opensips.org>
Sent: 23 December 2016 14:14
To: Jonathan Hunter; OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position

Hi Jonathan,

A Merry Christmas to you too !

I found a small mistake in the original patch I sent you. Please revert that one and use this new patch (see attachment). Let me know if it does the trick.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

[http://www.opensips-solutions.com/imgs/slideshow/slide3.jpg]<http://www.opensips-solutions.com/>

Home - OpenSIPS Solutions<http://www.opensips-solutions.com/>
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OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities.


On 22.12.2016 13:58, Jonathan Hunter wrote:

Hi Bogdan,


Merry Christmas to you and the mailing list!


I am testing the real queue position, and with one call in progress, and then 2 subsequent calls, both of which hit the hold music loop, I unfortunately dont see the position value increment, it remains at cc_pos=0 for all the calls, this is with just one queue defined to keep things very  simple.


Please let me know what debug or information you require, here are some outputs;



opensipsctl fifo cc_list_calls
Call:: 255.0 Ref=2 State=queued Call Time=2 Flow=Cust1
Call:: 884.0 Ref=2 State=queued Call Time=51 Flow=Cust1
Call:: 344.0 Ref=1 State=toagent Call Time=70 Flow=Cust1 Agent=2000

opensipsctl fifo cc_list_agents
Agent:: 2000 Ref=1 Loged in=YES State=incall

opensipsctl fifo cc_list_flows
Flow:: Cust1 Avg Call Duration=199 Processed Calls=3 Logged Agents=1 Ongoing Calls=3 Ref=3

opensipsctl fifo cc_list_queue
Call:: 0 Waiting for=114 ETW=0
        Flow:: Cust1 Priority=256 Skill=custcare
Call:: 1 Waiting for=65 ETW=199
        Flow:: Cust1 Priority=256 Skill=custcare

Many thanks

Jon




________________________________
From: Bogdan-Andrei Iancu <bogdan at opensips.org><mailto:bogdan at opensips.org>
Sent: 07 November 2016 20:41
To: Jonathan Hunter; OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position

Hi Jonathan,

I was not able to test it with a real queue, so let me know if it really does the job (in terms of reporting the real position in the queue).

Some questions:
1) currently I add that value only when sending to the queue / MOH - do you foresee any need to be added for other announcements like for welcome ?

2) will it be useful to add the ETW (estimate time to wait) ? is it useful ?

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com







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