Dadburns
Guru
- Joined
- Feb 25, 2010
- Messages
- 17
- Reaction score
- 0
I recently upgraded a customer using PIAF 1.2.9 to their newly purchased AT&T IP Flex trunk. I'm sure most people posting here are somewhat familiar with the IP Flex service from AT&T, but just in case let me describe it anyway:
It's simply a synchronous T1 circuit bundled with a SIP service. AT&T will provide this with a Cisco router on the customer end specially configured to convert that SIP trunk into either standard analog CO lines or PRI outputs, but if you ask reeeaallll nice, they will let you connect directly via SIP which was my preference.
As usual working with AT&T was painfull. I worked the better part of 6 hours to bring this trunk up and most of that was holding for the next available tech.
The AT&T techs were not much help in general, the only information any of them gave me until the last guy was; “we’re sending 7 digits” and “your Border Elements are nn.nnn.nn.nnn”.
The "Border Elements" are just the IP addresses of the AT&T SIP server, when I asked for things like: Username, secret or peer information they didn't know what I was talking about.
The last guy I worked with did speak SIP and sent me this Word Document:
http://www.i3techgroup.com/AsteriskConfiguration_Guide.doc
But by the time I got hold of him I had already figured it out and had calls going in/out and just needed him for testing and the final cut over, here are the things I learned:
Under General Settings make this change:
And create Inbound Routes for each phone number being ported through the IP Flex trunk:
It was ridiculously easy… once I figured all this out without any real help from AT&T.
Now, in all fairness to AT&T I have to say; that the call quality is fantastic, so far the reliability is great and the built-in QOS makes the internet service in excess of SIP demands a great value. I am recommending this service to our other PIAF customers.
It's simply a synchronous T1 circuit bundled with a SIP service. AT&T will provide this with a Cisco router on the customer end specially configured to convert that SIP trunk into either standard analog CO lines or PRI outputs, but if you ask reeeaallll nice, they will let you connect directly via SIP which was my preference.
As usual working with AT&T was painfull. I worked the better part of 6 hours to bring this trunk up and most of that was holding for the next available tech.
The AT&T techs were not much help in general, the only information any of them gave me until the last guy was; “we’re sending 7 digits” and “your Border Elements are nn.nnn.nn.nnn”.
The "Border Elements" are just the IP addresses of the AT&T SIP server, when I asked for things like: Username, secret or peer information they didn't know what I was talking about.
The last guy I worked with did speak SIP and sent me this Word Document:
http://www.i3techgroup.com/AsteriskConfiguration_Guide.doc
But by the time I got hold of him I had already figured it out and had calls going in/out and just needed him for testing and the final cut over, here are the things I learned:
- AT&T uses no internal security on the SIP communication within their IP Flex service. There is no user name, password or account authentication of any kind. Instead they assign four Class A IP addresses (that they call “Border Elements”) to every account (they must have IP addresses to burn!) and use IP security on the ports instead. I guess that works in that they control the T1 all the way through.
- AT&T does not use a standard SIP header. Asterisk does not recognize incoming calls by a SIP identity so you can’t receive the calls by creating an inbound trunk with that identity, instead you have to configure Asterisk to “accept anonymous SIP calls”…. But that’s not enough… Remember the whole “We’re sending 7 digits” thing? Well, in fact you have to create inbound routes with those 7 digits as the DNIS for calls to come in (a generic “all calls” route won’t work) because they build those 7 digits into the SIP header.
Under General Settings make this change:
And create Inbound Routes for each phone number being ported through the IP Flex trunk:
It was ridiculously easy… once I figured all this out without any real help from AT&T.
Now, in all fairness to AT&T I have to say; that the call quality is fantastic, so far the reliability is great and the built-in QOS makes the internet service in excess of SIP demands a great value. I am recommending this service to our other PIAF customers.