asterisk users - Aug 2012 - Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration Tutorial

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

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -
http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -
http://asipto.com/u/kpw

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

On Mon, Aug 6, 2012 at 6:48 PM, Daniel-Constantin Mierla
<miconda at gmail.com>wrote:
> 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
>
> --
> Daniel-Constantin Mierla - http://www.asipto.com
> http://twitter.com/#!/miconda -
http://www.linkedin.com/in/**miconda<http://www.linkedin.com/in/miconda>
> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -
> http://asipto.com/u/katu
> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -
> http://asipto.com/u/kpw
>
>
> --
> ______________________________**______________________________**_________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
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> To UNSUBSCRIBE or update options visit:
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On 06/08/12 13:48, Daniel-Constantin Mierla wrote:
>   * 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.
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 at 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