[SR-Users] Thoughts on setup with Kamailio and FreeSWITCH

[Daniel-Constantin Mierla]

Hello,

On 03/09/15 14:23, Michael Nielsen wrote:
> Hello everyone...
>
> I'm testing various ways of setting up Kamailio and FreeSWITCH (as
> VAS-handler).
>
> I'm struggling a bit on what Kamailio should do and FreeSWITCH should do.
>
> My basic setup is to handle voicemails, internal calls between
> subscribers and outgoing calls to a SIP gateway to GSM.
>
> My idea is to have Kamailio handle all calls and routing between
> subscribers and the SIP gateway and only have FreeSWITCH handle
> voicemail (to save the audio file and e-mail it).
>
> How does that sound?
>
> Or should I route everything to FreeSWITCH via Kamailio and have
> FreeSWITCH route calls between subcsribers and SIP gateway as well?
>
> What are best practises etc...?
I prefer to do everything that can be done with Kamailio, and the rest
with FreeSwitch. The reasons might be obvious ...

Anyhow, if you need to intercept DTMF in the middle of the call, or do
transcoding, then you have to route the calls via freeswitch, even the
routing logic is done by kamailio.

Over the time, I published few tutorials about kamailio+freeswitch, some
of them might be useful for your:

  - https://kb.asipto.com/freeswitch:index

Cheers,
Daniel

-- 
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.com
Kamailio Advanced Training, Sep 28-30, 2015, in Berlin - http://asipto.com/u/kat