[SR-Users] Job: System Engineer VoIP - Germany / HO

[Gerrit Jacobsen via sr-users]

VIER is looking for a System Engineer VoIP

What VIER does

VIER is a global contact center technology provider. VIER develops and runs its 
own technology in several data centers. You will be part of the Real-time Media 
Processing Team which is responsible for the telephony connections between 
carriers, applications and call-center agents.

Your mission

You are responsible for the support of the VoIP systems as well as the 
associated infrastructure
Since you have sovereignty over the VoIP systems, you are also responsible for 
performing system upgrades and configurations.
You have a lot of freedom to design, plan, and implement new VoIP system 
components for internal and external projects.
You manage and implement change requests and are a specialist for special 
topics in the communication area.
You show a high-sense of responsibility and foresight to keep systems running 
24/7
You are prepared to work occasionally also in the evening hours to roll out new 
features

Your skills

You have several years experience in the day-to-day operation of telecom systems
You have worked extensively with opensource VoIP software such as Kamailio / 
OpenSIPS, FreeSWITCH
You can administer Linux systems such as RockyLinux and Debian
You know SIP, RTP, RTCP, TLS, WebRTC protocols and can debug problems with 
Wireshark
You have also worked with MySQL, Git
You are communicative and like to work with team colleagues

Further beneficial skills

Experience with Ansible or Puppet for automation and configuration management.
CheckMK, Homer, Hepic, Redis, CGRates, CDR Tools, Python
Own VoIP software projects
German language skills

Please apply here: 
https://vier.jobs.personio.de/job/1638419?language=en&display=en


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Re: [SR-Users] Connection between Kamailio and redirect server

[Gerrit Jacobsen]

Hi Igor,

> I'm not sure to understand, and even after reading the dispatcher module 
> documentation, if my understanding is correct: in the use-case described, the 
> module is used to load-balance the INVITE towards multiple route servers, 
> right?


yes. You don't necessarily need the dispatcher for your case.

> But then, I'm not sure to understand what is reading the 302 responses and 
> the URI include.

The 302 responses are processed in the failure route by the TM module. You have 
to set the failure route before you send the invite to the redirect server. 
https://www.kamailio.org/docs/modules/devel/modules/tm.html#tm.serial_forking 


Best Gerry


> On 16 Feb 2021, at 09:52, Igor Potjevlesh  wrote:
> 
> Hi Gerry
>  
> Thank you for highlighting this thread. It's useful to start!
> I'm not sure to understand, and even after reading the dispatcher module 
> documentation, if my understanding is correct: in the use-case described, the 
> module is used to load-balance the INVITE towards multiple route servers, 
> right?
> But then, I'm not sure to understand what is reading the 302 responses and 
> the URI include.
>  
> Regards,
>  
> Igor.
>  
> De : sr-users  > De la part de Gerry | Rigatta
> Envoyé : vendredi 12 février 2021 12:51
> À : Kamailio (SER) - Users Mailing List  >
> Objet : Re: [SR-Users] Connection between Kamailio and redirect server
>  
> Hi Igor,
>  
> The following might help
>  
> https://lists.kamailio.org/pipermail/sr-users/2015-August/089388.html 
> 
>  
> Cheers Gerry
>  
>> On 12 Feb 2021, at 11:09, Igor Potjevlesh > > wrote:
>>  
>> Hello !
>>  
>> I will have to connect my Kamailio proxies to a routing server which will 
>> receive INVITE and replies with 302 and a list of a SIP Contact where 
>> (gateways, IP, prefixes or whatever we want) the call must be routed to.
>>  
>> I'm wondering what would be the best approach to handle with Kamalio those 
>> 302 and then proceed with the routing?
>> For the moment, we use to work with carrierroute. But, since the redirect 
>> server will take the routing decision, I believe that carrierroute will no 
>> be useful anymore.
>>  
>> Thank you for your inputs/experience.
>> 
>> Regards,
>>  
>> Igor.
>>  
>>  
>> 
>>     
>> L'absence de virus dans ce courrier électronique a été vérifiée par le 
>> logiciel antivirus Avast. 
>> www.avast.com 
>> 
>> 
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[SR-Users] Push proxy and load balancer for asterisk?

[Gerrit Jacobsen]

Good morning,

I want to realise a push notification proxy to signal incoming calls from 
asterisk to softphones. Is Kamailio the preferred solution for this? I looked 
also at Flexisip and Asterisk scripts and was told by a seasoned professional 
that Kamailio is the preferable solution as its scalable. True?

I would be glad if someone could share his experiences.

Also can one Kamailio instance be used as an asterisk load balancer and also 
push notification proxy ?

I have been a heavy openser user a long time ago. I used first openser as my 
core routing engine but then switched to yate. Please excuse my rather broad 
questions. From my experience with Openser the devil is in the detail. So any 
hints on where this setup may cause problems would be great.

Thanks a lot for your input.

Gerry

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