Kamailio 3.3.x And Asterisk 10.7.0 Realtime Integration Tutorial

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Asterisk Users 2 Comments

Hello,

I released an update to my series of Kamailio and Asterisk Realtime Integration, using the latest stable versions of the two projects, respectively 3.3.1 and 10.7.0. You can find it at:

* http://asipto.com/u/68

The tutorial focuses on how to use Asterisk’s database structure to perform authentication in Kamailio SIP server, along with user location, nat traversal, instant messaging, presence, a.s.o., offloading processing from Asterisk. Asterisk will still handle all the calls, enabling rich telephony such as MoH, transcoding, ring back, IVR, etc.

Reusing as much as possible the Asterisk database makes the architecture presented in the tutorial easy to be applied to existing installations, without losing management interfaces or other admin tools.

Hope it is useful for many folks out there.

Cheers, Daniel

2 thoughts on - Kamailio 3.3.x And Asterisk 10.7.0 Realtime Integration Tutorial

  • Thats a great tutorial with very good conceptual details like SIP messages flow. Thanks Daniel 🙂

  • This is a good tutorial, but can you clarify the scope of what Kamailio will do in this configuration?

    – just scalability and protocol conversion (e.g. UDP with Asterisk, TLS
    with phones)?

    – does it mean Kamailio is also intended to add other services, e.g. presence and IM functionality?

    – any comments on using the Jabber gateway module?

    – is it intended for fully federated SIP, e.g. someone sets this up for example.org, and somebody else in example.com can make a call to user@example.org, routed over the public Internet, using DNS SRV and mutual TLS?

    If it is intended that someone can turn on the mutual TLS mode and use it to federate their Asterisk server, then I’d like to link to it from

    http://www.opentelecoms.org/use-a-sip-proxy-instead-of-asterisk