Thankyou Alexandru for your suggestions.
I'll give it a try tomorrow and will report my progress here.
It seems that i'm not so far from the result!
Bruno
Il giorno mar 11 ago 2015 alle 23:44 Alexandru Covalschi <568...@gmail.com>
ha scritto:
> Also, take a look at kamailio-advanced.cfg, there is
Also, take a look at kamailio-advanced.cfg, there is PSTN GW route already
included. Also you can use LCR for routing calls to different providers, a
simple guide can be found here
http://dopensource.com/least-cost-routing-with-kamailio-v4-1/
2015-08-12 0:41 GMT+03:00 Alexandru Covalschi <568...@g
First of all I'd suggest to use
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
guide in combination with
http://saevolgo.blogspot.com/2011/11/how-to-increasing-voip-services.html
But, assuming your platform is behind NAT, you need:
1st. Use rtpengine instead of rtpproxy
Hello,
i'm on my first try with kamailio. I need to build a SIP balancer that
should keep SIP
registration from VoIP provider and route the calls to the asterisk boxes
where an IVR
will take care to answer.
Here's my network topology:
+---> [asterisk1]
[publi
