AT&T Flex SIP Trunking

Hi There,

I am having an issue with getting SIP trunks configured to work with AT&T’s Flex SIP Trunking. They don’t provide much support on their end because FreePBX is unsupported. Basically all I was provided with was 2 IP addresses (12.194.215.173 and 12.194.223.109). They said no authentication occurs and that I will not see a registration through my PBX system. When I run the show peers command through the Asterisk CLI I get unreachable. Has anyone else encountered this issue and if so what configuration settings did you set.

Thanks!

We have AT&T sip trunking. Here is our config
disallow=all
allow=ulaw
context=from-pstn
allowoverlap=yes
realm=asterisk
bindport=5060
srvlookup=no
videosupport=no
sendrpid=no
dtmfmode=rfc2833
canreinvite=yes
directrtpsetup=yes
type=peer
host=XX.XX.XX.XX
qualify=2000
insecure=invite,port

They do have a more detailed guide if you ask for it. Also, this setup has been tested on both 1.8 and 11. Does not work reliably on 1.4. You setup 2 trunks. 1 with the 1st IP and the second with the other IP.

One more thing. The trunk will show as reachable, but if I remember correctly, they had to initiate the first call to me at turn up time before the trunk worked. Outbound calling did not work, until the first successful incoming call was placed.