Integrating OpenSIPs with Asterisk

Hello All,

I think integrating opensips with asterisk makes a voip solution much more productive and gainful. But it seems there is no specific documentation on how to do the integration. I wish there should be a documentation on getting it done. It will be very useful for even non-techies to go through the documentation and make up the system.

Also I believe there may be many who has got that setup already running. But creating a documentation will be useful for many people.

Regards
Vino

[quote=“vinodc”]Hello All,

I think integrating opensips with asterisk makes a voip solution much more productive and gainful. But it seems there is no specific documentation on how to do the integration. I wish there should be a documentation on getting it done. It will be very useful for even non-techies to go through the documentation and make up the system.

Also I believe there may be many who has got that setup already running. But creating a documentation will be useful for many people.

Regards
Vino[/quote]

opensips.org/index.php?n=Res … umentation

What is the difference between OpenSIPS and Kamailio ?

Hi Vinodc,

we are also facing the same problem. We found a howto how to integrate OpenSer with Asterisk and tried to modify the OpenSer code to our needs to make it work with OpenSips but unfortunely with no success.

Hier you can find the mentioned Howto, maybe you find a clever way to modify it in order to make work with asterisk.

voip-info.org/wiki/view/Real … th+OpenSER

if you achieve any success please let me know

best regards

OpenSIPS and Kamailio are both forks of the OpenSER project. There was some internal fighting and the two main guys decided it was best to go their own ways. I believe Kamailio has the old OpenSER SVN repository… while OpenSIPs ended up with the www.openser.org domain name.

I have Kamailio integrated into my network of asterisk servers. What exactly are you trying to accomplish?

[quote=“g2010”]…

I have Kamailio integrated into my network of asterisk servers. What exactly are you trying to accomplish?[/quote]

Just to learn something new. :smiley:

Well, I may have a fairly unconventional deployment of asterisk/kamailio.

My Kamailio (OpenSER) server serves four primary purposes in my topology:

1 - It is my inbound and outbound SIP gateway. All SIP messaging comes through or goes out my Kamailio server(s). This allows me to lock down my individual asterisk servers from the outside world on the SIP ports, meaning I can’t be subjected to REGISTER flood attacks or the similar.

2 - It is a load balancer for calls from the PSTN into my network. I use the dispatcher module to distribute calls amongst our many IVR/voice processing servers.

3 - It is an LCR engine for my asterisk servers to terminate calls to the PSTN. I have multiple PSTN gateway providers and Kamailio is quite good at helping me route to the provider with the lowest rate for the dialed number.

4 - It is my call accounting system. It keeps track of all calls coming in or going out of my network. The endpoints, duration, etc.

I don’t do any registration, voicemail, or presence with my deployment… but I have heard of many people who have been successful with that stuff.

I can understand that multiple asterisk boxes can run as PSTN gateways behind your Kamailio server for LCR and Load balancing.

What kind of authentication you have in your Kamailio. Is there anything that we can do accounting for the accounts in the Kamailio.

Thanks
Vino

I’m having trouble doing this also:

Here’s what I have so far:

2x Asterisk (1.6) machines with interconnection via dundi
On one of the boxes i’ve installed openSIPS 1.5.1

I’m to the point where my phones will register with openSIPS, but I cannot get opensips to talk to asterisk.

Any help would be greatly appreciated.