OK… I’m a bit of a noob at all this but I’ll describe what I have setup first…
Setup a Ubunto 14.04 server on MS Azure Cloud platform.
Used the following guide to install Asterisk 11.7 and Freepbx 12.0.3…
://wiki.ubuntu.com/installingFreepbx
In Freepbx Sip settings I reduced the number of RTP ports (10000-10006), I also set NAT to ‘no’ and entered the static IP(VIP) of the Azure server.
On Azure I added endpoints for UDP 10000-10006, TCP 5269, UDP 5060.
I have 2 Generic SIP extensions programmed in through Freepbx and 2 mobile handsets each running CSipSimple Sip Client. I am not using inbound/outbound trunks at this stage…I am just trying to get 2 handsets to ‘talk’ to each other using the Asterisk/Freepbx platform.
When I run the SIP client on both handsets they register successfully. I can call from one handset to the other and ‘answer’ the call on the receiving handset but I can’t get sound on either handset.
Everything I have read so far suggests that 99% of these ‘sound’ issues are related to NAT. I’m not using NAT! I have tried with NAT on and off, have also tried using DynamicIP instead of StaticIP, rebooting the server and SIP clients after each change but nothing seems to make a difference.
There must be something simple that I’m missing but, as a noob I’m really not sure where to look next.
All help greatly appreciated.
Mike
PS - I’m also new to Ubuntu so if I need to do something on the CLI please be quite specific otherwise I’ll get completely lost! Thanks. 