[asterisk-biz] asterisk and kamailio realtime integration tutorial

[Daniel-Constantin Mierla]

On 5/14/10 9:18 PM, Daniel-Constantin Mierla wrote:
> Hello,
>
> I thought this might be interesting for some guys here - I wrote a new 
> tutorial about realtime integration of Asterisk and Kamailio 
> (OpenSER), it is available at:
> https://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb 
>
small update about the link -- you can browse on http, no need to bother 
with https and self-signed cert:
http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb

Thanks to Tim.R for emailing me,
Daniel

>
> In the past I wrote and maintained Asterisk and OpenSER (then 
> Kamailio) relatime integration for various versions, but a different 
> approach, published at voip-info.org, like:
> http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+1.4+With+OpenSER+1.3 
>
> http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+1.4+With+Kamailio+1.5.x 
>
>
> There is a new approach with latest tutorial. First, the used database 
> is Asterisk realtime, not kamailio (openser) database like in old 
> tutorials -- this allow to update easily from existing Asterisk 
> installations using realtime, keep your GUI to add/remove extensions 
> and manage the system.
>
> Second, voice call routing is handled by Asterisk, practically the 
> VoIP service goes as you have it configured with Asterisk only (in the 
> old tutorials, call routing between users was handled by kamailio 
> (openser), Asterisk being used only for several media services, such 
> as voicemail, conference, announcement). Therefore you can play audio 
> within each call, have customized ringing tones, process DTMF with 
> calls, etc.
>
> Cheers,
> Daniel
>

-- 
Daniel-Constantin Mierla
Kamailio (OpenSER) Advanced Training
Miami, Fl, USA - June 21-23, 2010
http://www.asipto.com/index.php/kamailio-advanced-training/