We are working on deploying a system with FreePBX 2.10.1.2 & Asterisk 1.8.16.0 using AT&T as the SIP Trunks provider. They are not sure about the compatibility on those versions of FreePBX and Asterisk with their network. AT&T states that they the versions compatible with their system are Asterisk Business Edition C.1.6.2 & Asterisk 1.4.15. However, they are not sure about the versions we are using. Does anyone have any experience with this type of product with that provider. I have Googled this issue with no success, we would appreciate any input.
I am working on this now as well. I was able to make calls just fine, and receive calls (but with no audio). I am playing with the codecs, and let you know what I find.
It’s not your CODEC’s. That’s would generate an immediate fail. You have a NAT issue. More than likely you don’t have the AT&T gateway in the same network as your PBX or you missed the localnet setting in SIP settings.
Thanks for the info Skyking, but it wasn’t NAT. I had the codec settings in the trunk config as well as in the “Asterisk SIP settings” section. As soon as I removed the codes settings from the trunk config, everything works! I will post my config later for Franco177. I have FreePBX behind a Cisco ASA, so I can post that config as well if anyone wants it.
Skyking - you are correct. I did a bunch of things at the same time, so I kinda assumed the codec fixed it since that is the last thing I did to make it work.
Here you go Franco. Hope it helps! Â I took it from my install guide, and works well for us.
General Settings
1. Trunk Name = ATT 2. Outbound CallerID = Whatever you want to display as your main number 3. CID Options = Allow Any CID 4. Maximum Channels= However many SIP lines you bought from AT&T 5. Disable Trunk=Nothing checked 6. Monitor Trunk Failures= Nothing Checked
Dialed Number Manipulation Rules Leave at default settings
IP Configuration:
Select Static IP
Click Auto Configure and check the values it enters ensures that the following are entered (tweak it if you need to):
External IP=The public IP address AT&T is sending the SIP to
Local Neworks=Your local network address E.G. 192.168.1.0/255.255.255.0 (Don’t forget any VPN addresses)
Skyking - I had the following as the first three lines in my SIP trunk config:
disallow=all
allow=ulaw
allow=G729
I had both codecs there as well as in my SIP config section. After I removed the settings from my trunk config, all was well. Now that I think about it, I had to add “NAT=yes” to my trunk settings as well.
I am not familiar with the licensing aspect of the codec. Now that I have selected the G729 codec, my assumption is that it will not use it until it has been purchased and downloaded? I did a quick search and it appears you need to purchase it based on concurrent calls. Given that we have 6 SIP trunks am I correct in that I need 6 licenses of G729?
Not really, it is as you deduced based on concurrent calls, for example if you land g729 call on an IVR and then send it to a g729 phone, you will need two to transcribe the call, if you accept a call from your carrier in g729 and send it to a phone which has a g729 license you will need none. If you have 6 SIP trunks that send g729 and land on an IVR that sends back to 6 phones that use g729, you will need 12 licenses.
It uses less bandwidth. Disadvantage is it costs money and uses more processing. GSM comes close to bandwidth saving and is free. For good quality phonecalls use g711 where you can, like on your LAN, only transcode to a lower bandwidth codec when you need to for bandwidth limitation, but seriously, did you ever think of consulting auntie google? she has all your answers for you already . . .
