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 <mico...@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 > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/**mailman/listinfo/asterisk-**users<http://lists.digium.com/mailman/listinfo/asterisk-users> >
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