[OpenSIPS-Users] opensips 2.1 call_center queue position

Wed Nov 2 08:56:43 CET 2016

Hi Bogdan,

Thanks very much for this.

I have just applied patch (installed from sources so when to call_center module directory and ran patch < call_center_pos.patch) then did a recompile.

However when I now route to the call center (cc_handle_call) it generates a core and kills opensips;

!!!!user 2000 has Callqueue set so send to Call Queue Route
Nov  2 07:53:42 HPBXProxy1-beta /sbin/opensips[21141]: NOTICE:core:io_wait_loop_epoll: EPOLLIN(read) event: epollwait() set event EPOLLHUP - connection closed by the remote peer!
Nov  2 07:53:42 HPBXProxy1-beta /sbin/opensips[21141]: CRITICAL:core:receive_fd: EOF on 19
Nov  2 07:53:42 HPBXProxy1-beta /sbin/opensips[21112]: INFO:core:handle_sigs: child process 21119 exited by a signal 11
Nov  2 07:53:42 HPBXProxy1-beta /sbin/opensips[21112]: INFO:core:handle_sigs: core was generated
Nov  2 07:53:42 HPBXProxy1-beta /sbin/opensips[21112]: INFO:core:handle_sigs: terminating due to SIGCHLD

Do you need me to backtrace/debug through to get the issue? Or is problem how I applied patch?

Many thanks

Jon

________________________________
From: Bogdan-Andrei Iancu <bogdan at opensips.org>
Sent: 01 November 2016 21:44
To: Jonathan Hunter; OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position

Hi Jonathan,

Please give it a try to this patch - it is not really tested, but when the call is sent the Queue announcement, it should have a ";cc_pos=xxx" parameter giving the position is the queue (0 being the first to be dispatched to agents).

Let me know if it works.

Regards,

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

Home — OpenSIPS Solutions<http://www.opensips-solutions.com/>
www.opensips-solutions.com
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 28.10.2016 15:59, Jonathan Hunter wrote:

Hi Bogdan,

Great news, really do appreciate that.

Many thanks

Jon

________________________________
From: Bogdan-Andrei Iancu <bogdan at opensips.org><mailto:bogdan at opensips.org>
Sent: 28 October 2016 12:48
To: Jonathan Hunter; OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position

Hi Jonathan,

No, it is no yet available. Give me couple of days and I will make a patch for it.

Best regards,

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

Home — OpenSIPS Solutions<http://www.opensips-solutions.com/>
www.opensips-solutions.com<http://www.opensips-solutions.com>
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 25.10.2016 19:22, Jonathan Hunter wrote:

Hi Bogdan,

Sorry cant recall If I replied to this.

Is cc_pos available now to extract from the module?

Thats the only thing I need then I can implement call center which I think will be much more scale-able than the other approach I am using with FreeSWITCH, I would use that just for announcements.

Any response/help appreciated.

Jon

________________________________
From: Bogdan-Andrei Iancu <bogdan at opensips.org><mailto:bogdan at opensips.org>
Sent: 13 October 2016 10:59
To: Jonathan Hunter; OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position

Hi Jonathan,

No, currently this is not possible. I was trying to envision a solution for your need.

But, checking the code, it is really difficult to add the headers to the INVITEs originated by OpenSIPS (via the B2BUA), as we need some flexibility (different headers to different INVITEs belonging to the same B2B scenario , and even more, we need to traverse couple of internal APIs - to propagate the hdrs from Call center module all the way to TM).

So, a simpler approach may be to add such extra info as URI params to the RURI. Like if you have the RURI "sip:queue at 192.168.1.10:5060"<mailto:sip:queue at 192.168.1.10:5060> for the queue/waiting playback, the RURI in the INVITE to the media server will look like :  sip:queue at 192.168.1.10:5060;cc_eta=40;cc_pos=10<mailto:sip:queue at 192.168.1.10:5060;cc_eta=40;cc_pos=10>  - cc_eta being the estimated time to wait in seconds and cc_pos the position in the queue.

What do you think of this ?

Regards,

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

On 12.10.2016 17:21, Jonathan Hunter wrote:
Hi Bogdan,

Yes being able to grab the queue position would be perfect.

Is that possible?

Thanks

Jon

________________________________
Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position
To: hunterj91 at hotmail.com<mailto:hunterj91 at hotmail.com>; users at lists.opensips.org<mailto:users at lists.opensips.org>
From: bogdan at opensips.org<mailto:bogdan at opensips.org>
Date: Wed, 12 Oct 2016 15:42:43 +0300

Hi Jonathan,

When a call is mapped to a flow / queue (before playing the welcome message), we know the ETA (estimated time to wait) and when is placed in the queue (before playing the queuing) we internally know the position in the queue.

Would it help to have the position in the queue placed into a custome SIP header, when sending the INVITE to the message_queue URL ? or to the welcome message ?

Regards,

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

On 12.10.2016 12:06, Jonathan Hunter wrote:
Hello Bogdan,

Thanks for the response.

In terms of my question, with a number of queuing platforms, they have the capability to tell the caller, what position they are in , and when they are likely to be answered.

I just wondered if this logic was already within the module, or if I would need to use an external code/script to facilitate this function?

As I presume call_center tracks the number of calls currently in a queue ? I would then want to be able to extract that information, and if a caller was for example in 3rd place in a queue, I could inject the relevant audio from freeswitch to tell them their current position?

Does that make sense? :)   Just wanted to know if its something this module can do?

Thanks

Jon

________________________________
Subject: Re: [OpenSIPS-Users] opensips 2.1 call_center queue position
To: users at lists.opensips.org<mailto:users at lists.opensips.org>; hunterj91 at hotmail.com<mailto:hunterj91 at hotmail.com>
From: bogdan at opensips.org<mailto:bogdan at opensips.org>
Date: Wed, 12 Oct 2016 11:23:45 +0300

Hello Jon,

The message_queue is a SIP URI pointing to an audio announcement to play to roll of the waiting/in-queue playback. This needs to be an announcements that never ends (from the perspective of the media server); only the the OpenSIPS Queue may terminate the playback, when it decides to take out the call from waiting and to deliver it to an agent.

As for your question, I'm not sure I understand what you mean by "inject a message with queue position for the caller in question" - could you detail please ?

Regards,

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

On 11.10.2016 13:36, Jonathan Hunter wrote:
Hi guys,

I have implemented an opensips/freeswitch environment, and I wish to add call queues to it, and I like the look of call_center, so just checking this out in comparison to mod_callcenter in FS world.

My main question is if using the call_center module if you can inject a message with queue position for the caller in question, as I cant see that in documentation, I only see message_queue which I assume could be used to report the callers position, but just wondered if anyone has done this and if they could give me some tips as to if possible?

Many thanks

Jon

_______________________________________________
Users mailing list
Users at lists.opensips.org<mailto:Users at lists.opensips.org>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20161102/ad6eb7e8/attachment-0001.htm>