Daniel-Constantin Mierla postou um link para um tutorial sobre a integração do Asterisk e Kamailio usando realtime:

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:

http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb

In the past I wrote and maintained Asterisk and OpenSER (then Kamailio) realtime 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

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