Anyone using FreePBX and AT&T IP Flexible Reach?
If so is there an easy way to set it up? Unfortunately AT&T just sent me a 33 page PDF from 2008, which is all but useless.
Basically they give you an example of what extensions.conf, rtp.conf, and sip.conf should look like, however none of the fields are matching up (prolly cuz the doc is so old).
Example, in extensions.conf they want you to enter:
SIPTRUNK1=SIP/first_gateway
SIPTRUNK2=SIP/second_gateway
The IPs for first_gateway and second_gateway are then set in sip.conf
What would this match up to in the FreePBX interface?
TIA
Extensions.conf is the dialplan and those are variable declarations. FreePBX takes care of all of this. SIP is SIP. It’s not any different or better because it’s brought to you by AT&T Either (We can sell you and MPLS T1 with all the SIP trunks you want for less).
Setup the networks so the FreePBX box can reach the AT&T sever (I would use a seperate interface and a static IP route) and then configure trunk and routes in FreePBX. Don’t forget to use a FreePBX context and not the one in the AT&T example.
This applies to any SIP provider. Some are just tought than others. AT&T is simple.
Thanks for the info! The numbers from AT&T were included with a T1 line (High-speed isn’t available in the area).
It seems like it should be easy to setup.
I think somethings wrong on AT&T’s end. And I can’t contact them directly, I have to go through a 3rd party support person, who then contacts an “engineer” at AT&T and gets back to me several days later.
I have the peer details from them as follows:
[first_gateway]
type=peer
host=135.25.xx.xx
qualify=2000
So I’m guessing that goes in PEER Details under Outgoing Settings when I’m adding a trunk (with the exception of [first_gateway]. It appears they don’t have incoming settings or a registration string and they have a “call control element” that validates the call orgin based on IP, but I’m not 100% sure on that.
Yeah you don’t need any inbound settings, one peer is all you need
type=friend
host=135.25.x.x
context=from-pstn
insecure=port,invite
disallow=all
allow=ulaw
This will allow all calls from that IP. The insecure tells it to ignore authentication and the friend tells it to inny and outy. You can set qualify to 2000 if AT&T’s switch really takes 2 seconds to respond. In my experience that is quite high. The value is in ms.
Take a look at the Asterisk documentation sample sip.conf it contains an explanation of every operator and what it does.