[SR-Users] RTPPROXY issue and sip to sip calling

[Ahmed Rehan]

Dear All

I m trying to setup kamailio and asterisk in load balancing with a2billing
. Currently all of my VMs, one Kamailio and two asterisks are on same
subnet . I have started the RTPproxy like below

./rtpproxy -s udp:127.0.0.1:7722 -l X.X.X.153 -m 10000 -M 50000 -u root
root -F -d INFO LOG_LOCAL0

My question is if all the VMs are on same subnet with same gateway what
should be written in the private IP X.X.X.153/<private - ip>

Secondly i m authenticating and registering the SIP on kamailio using the
A2B DB . all the dialplan for a2b is being run on asterisk . Now if i want
to call SIP peer to Peer like in case of followme case ,

How should i route the calls in Kamailio ? will it be using usr loc module?
if so any help will be appreciated

-- 


Regards
Ahmed Rehan
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20160129/f51f204c/attachment.html>