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

Tue Sep 19 09:20:10 UTC 2023

Thanks, let's move the discussion over there.

Regards,

Bogdan-Andrei Iancu

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

On 9/19/23 11:47 AM, Kosmas Palios wrote:
> of course, here it is
>
> https://github.com/OpenSIPS/opensips/issues/3176 
> <https://github.com/OpenSIPS/opensips/issues/3176>
>
> thank you
>
>
>
> Sent from Outlook for iOS <https://aka.ms/o0ukef>
> ------------------------------------------------------------------------
> *From:* Bogdan-Andrei Iancu <bogdan at opensips.org>
> *Sent:* Tuesday, September 19, 2023 11:42:03 AM
> *To:* OpenSIPS users mailling list <users at lists.opensips.org>; Kosmas 
> Palios <kosmas.palios at athenarc.gr>; 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>
> 	
>
> For the second issue, could you point the GH ticket?
>
> Regards,
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>    https://www.opensips-solutions.com  <https://www.opensips-solutions.com>
>    https://www.siphub.com  <https://www.siphub.com>
> On 9/19/23 11:35 AM, Kosmas Palios wrote:
>> 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
>>
>> Sent from Outlook for iOS <https://aka.ms/o0ukef>
>> ------------------------------------------------------------------------
>> *From:* Users <users-bounces at lists.opensips.org> 
>> <mailto:users-bounces at lists.opensips.org> on behalf of Bogdan-Andrei 
>> Iancu <bogdan at opensips.org> <mailto:bogdan at opensips.org>
>> *Sent:* Tuesday, September 19, 2023 11:20:23 AM
>> *To:* OpenSIPS users mailling list <users at lists.opensips.org> 
>> <mailto:users at lists.opensips.org>; Kosmas Palios 
>> <kosmas.palios at gmail.com> <mailto: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 <mailto: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.opensips-solutions.com>
>>    https://www.siphub.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/ 
>>> <https://blog.opensips.org/2021/01/06/the-script-driven-sip-b2bua/>
>>>
>>>
>>> " When comes to the modules using the b2b_logicAPI (providing 
>>> features on top of the B2B engine), the only affected one is the 
>>> call_centermodule. 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.
>>>
>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org  <mailto:Users at lists.opensips.org>
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users  <http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
>>
>>
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>

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