As written many time in this mailing list it depends on what services
you want to provide using such "voice servers". If they are transparent
media relays then rtpproxy or mediaproxy(-ng) can help. Also you can
integrate it with your billing to authorize calls and limit their duration.
Though
On 08/16/2013 12:29 PM, Roberto Fichera wrote:
> On 08/14/2013 09:58 PM, Vitaliy Aleksandrov wrote:
>> On 08/14/2013 07:32 PM, Roberto Fichera wrote:
>>> On 08/14/2013 04:36 PM, Vitaliy Aleksandrov wrote:
If you won't be able to disable SIP ALG on your router you can fill
$avp(received)
Thanks again Daniel
I've been reading deeply the module info and learning a lot. Your easy
solution worked thou, using the subst_hf function instead of the normal
subst the alias is added correctly and then the following bye is also
correctly handled.
The final line I use is:
subst_hf("Conta
Hello,
On 8/29/13 5:24 PM, Helena Garcia-Nieto wrote:
Thanks! Thank you very very much for your help.
I'll add the ngrep next time!
I'm absolutely new at kamailio and I'm doing the Subs on the main
config file (/usr/local/etc/kamailio/kamailio.cfg) on the PSTN call
detection route[PSTN] {:
Hello, I have a question about the load balancing module of kamailio.
As the site http://kb.asipto.com/ say, Kamailio is as a SIP proxy router to
scale Asterisk.
Can I run a kamailio instance as load balancer, and other several instances
as voice server replace of Asterisk?
If I can do that, coul
