[OpenSIPS-Users] Call center issue in 3.1; call center not working after migration to 3.2

Kosmas Palios kosmas.palios at athenarc.gr
Tue Sep 19 08:35:22 UTC 2023


This is indeed the solution, as we figured out by ourselves and as documented in the tm documentation!

Thank you for your kind reply here, as well as at the issue I posted sometime ago i github (my username is kosniaz).

Any input to the second issue (raised on github as well about 10 days ago) would be greatly appreciated!

Thank you very much one more time for your insights.

Have a great day!

Kosmas

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________________________________
From: Users <users-bounces at lists.opensips.org> on behalf of Bogdan-Andrei Iancu <bogdan at opensips.org>
Sent: Tuesday, September 19, 2023 11:20:23 AM
To: OpenSIPS users mailling list <users at lists.opensips.org>; Kosmas Palios <kosmas.palios at gmail.com>
Subject: Re: [OpenSIPS-Users] Call center issue in 3.1; call center not working after migration to 3.2

Δεν λαμβάνετε συχνά μηνύματα ηλεκτρονικού ταχυδρομείου από bogdan at opensips.org. Μάθετε γιατί είναι σημαντικό<https://aka.ms/LearnAboutSenderIdentification>
Hi Kosmas,

For Issue 1, try placing a t_newtran(); before calling the cc function in the script - this will prevent the issues due to retransmissions.

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
  https://www.siphub.com

On 7/4/23 1:15 PM, Kosmas Palios wrote:

Hello community,


Our team has managed to setup a SIP Trunk to forward calls to a number of SIP clients, in opensips 3.1.16. We are using the call center module, and it works fine for low traffic. We would like to ask two separate but related questions.


ISSUE 1


We are using the call center module to forward calls to 100 SIP agents, and it works well if traffic is relatively low (about 25 incoming calls per minute). However, when traffic is higher, i.e. up to 60 incoming calls per minute, we see calls getting rejected because of cc_handle_call() failing with error message:


DBG:b2b_entities:server_new: It is a retransmission, drop

ERROR:b2b_logic:b2b_process_scenario_init: failed to create new b2b server instance


Unfortunately, every time this happens, an agent's status gets stuck to "incall" forever, even though no cc_calls row includes him. So that agent is lost.


We are running in UDP mode, using 6 UDP workers. I’m attaching the configuration file as opensips_3_1_16.cfg

I can share the whole setup if needed.


ISSUE 2


We decided to migrate to 3.2 after seeing the bugfix to b2b_clients leak. When we got to migrating the call center, we read this blogpost: https://blog.opensips.org/2021/01/06/the-script-driven-sip-b2bua/


" When comes to the modules using the b2b_logic API (providing features on top of the B2B engine), the only affected one is the call_center module. The change is minor – the xml file controlling the call queuing logic was removed, as not needed any more. Otherwise, in terms of usage, it is exactly the same."


However, when we removed the lines:


modparam("b2b_logic_xml","script_scenario", "/etc/opensips/scenario_callcenter.xml")

modparam("call_center", "b2b_scenario", "call center")


the call center started behaving weird: it created another invite to the sip trunk, instead of creating the invite to the agent (the call id was good, but the to uri was wrong). I can give detailed logs on this, but I wouldn't want to make this email any bigger than it already is. I’m also attaching the configuration file as opensips_3_2_13.cfg


To sum up, our questions are:


1. Any ideas on what the problem is with creating a new server instance for high numbers of calls?

2. What's the recommended way to migrate the call center to version 3.2 ? Can we find an example script-driven call center somewhere?

Thank you in advance for your help!

Best regards,

Kosmas



P.S.: about our team: we are a small team from Athens, Greece integrating voice assistants on various platforms. Unfortunately we missed the latest Opensips summit held last September.




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