TUTORIAL HOW TO: AT&T IP Flex Trunk Configuration

Dadburns

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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:


  • 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.
In the end, I just needed to set up two simple trunks for outbound (the other two IPs they assigned aren't really used at all) that look like this in FreePBX;
image1.bmp



Under General Settings make this change:
image2.bmp



And create Inbound Routes for each phone number being ported through the IP Flex trunk:
image3.bmp


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.

 

blanchae

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With "allowing anonymous SIP calls", I hope that they mention putting in a catch-all inbound route to your IVR
 

Dadburns

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You make a good point, and yeah everybody using these trunks should put in such an inbound route, but in fact the "catch-all" inbound route that I already had on this system wouldn't work the way AT&T sends SIP calls over this trunk. I had to put in specific DNIS type inbound routes for calls to come in at all.
The other form of security on this trunk is that the router (AT&T retains management of these included routers) will only accept packets via the SIP ports from addresses of the AT&T servers. This effectively blocks you from using other SIP services over IP Flex. (Annoying)
 

blanchae

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Do you use SIP over the Internet or just through the IP Flex because allow anonymous SIP calls is a global setting. I wonder if it can be configured just for the IP Flex portion? Now I'm wondering what exactly does the "allow anonymous SIP calls" do in the dialplan.

Forgot to thank you for posting the solution. It's quite interesting.
 

Linetux

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Aside from the oddball DID stuff, this is exactly how Bright House networks delivers their SIP service as well (erm, except that they bring in fiber for pretty much limitless channels).
 

Dadburns

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When you use the IP Flex service from AT&T they provide both the T1 circuit and the SIP service. Because the T1 is theirs end-to-end they limit it to only move SIP packets to/from their SIP severs, so no you can't use some other SIP service over the internet service delivered via the IP Flex T1 without some trickery re; the port numbers used. And, you're welcome! I have been developing an in-house phone system using PIAF for the market about eighteen months now and have put out five of them, it's all been really interesting.
 

Dadburns

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Wow! I will look into Bright House, we are looking for VOIP vendors that can provide good service in our market area. Thanks for the tip!
 

Dylan Todd

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I just did a ATT Fiber Flex install for both Data and Voice. Im having some crazy stuff go on though. By looking at your information here i will apply these settings and check back.
 

krzykat

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What is ATT and Brighthouse charging for these T1's ? Also - do they allow you to pile on let's say 100 DID's on a T1?
 

Charles Berghuis

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Is anyone else having trouble viewing the images in this post? I can't see them. Has anyone done a ATT IP Flex trunk recently? Have to install one to this afternoon. Any tips would be greatly appreciated.
 

tombeck

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Hi Charles.
You cannot get any advice from AT&T technician. I could not believe that they configured wrong with ACL
in the Cisco router that they set up at our site, but the technician always tell that our PBX having problem, even we did not change any configuration at our server. I checked and know that the IP we assigned for the PBX was blocked at their router. The technician has spent 02 days for fixing the issue. We did not do anything at our end, and everything came back normally by them. It is so funny. But the service is good quality.
Here is the configuration (attached file):
+ You can use IP address from AT&T or using your own. If you have to use your own, you have to let them know for adding route at their router and for sending out the packets voice at AT&T system.
+ You should create 02 trunks in your Trixbox with 02 IPs address that AT&T gave to you, you have to ask them.
+ I do not mention about outgoing route in this thread, but if you need it, let me know.
Please, let me know if you need anything else.
 

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Charles Berghuis

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Thanks tombeck for your reply. I actually got the trunks working fairly quickly. As dadburns said in the beginning of this thread it's fairly simple. I installed a new Cisco Small Business router for the new fiber connection. I setup the port forwarding from the 2 ATT/SIP ips to my PIAF system in my router. I only put the ATT ips in my trunks, just using the default settings and taking out the password and username fields. The ATT tech was able to pass calls on both of his ips. We were also able to make out going calls with no problems. In testing everything worked fine. But ATT had one hick up, the fax line was in ATT computer to be ported, but it was not part of the order. The tech said it didn't look right on his end. He said we had a slim chance of maybe loosing the fax number. So we decided to abort the port and try it again next Friday. The company is able to use the new sip trunks for out going calls. ATT also didn't have my DID numbers ready for me. So next week I'll see if I can get this finished. Thanks again tombeck for your post. Thanks dadburns for starting this thread.
 

cespbx

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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:


  • 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.
In the end, I just needed to set up two simple trunks for outbound (the other two IPs they assigned aren't really used at all) that look like this in FreePBX;

image1.bmp



Under General Settings make this change:
image2.bmp



And create Inbound Routes for each phone number being ported through the IP Flex trunk:
image3.bmp


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.


Hi

I am also working on to configure ATT sip trunks with asterisk. I was reading your tutorial. Are those border elements (ip address) to assigned to PBX? if this is true then I guess i can dedicate NIC to sip trunks. I have multiple NIC's in my server. AT&T provide two border element IP's .

Also I couldn't see the images in this tutorial. Can some one provide those images or steps.
Thanks in advance
 

cespbx

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Charles

Looks like you recently configured AT&T sip trunks for your PBX. Can you please give me steps to configure AT&T sip trunks on my asterisk server.
 

Charles Berghuis

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Just a couple of tricks with it. Just build a SIP trunk and take out the username and password fields for your outgoing settings. Make sure you set you "HOST= ATT IP" in the outgoing field. For the incoming and registration fields leave blank. Make sure to ask ATT what digits their sending you on the inbound side. On mine they sent me 7 digits on my new numbers and 10 digits on my ported number. If you have ATT change this it takes a day or so to change it. It's better to use what their giving you. Go into the Asterisk Sip Settings and allow anonymous calls. Make sure you have a catch all inbound route that goes to a IVR or ring group. Then build your inbound and outbound routes. Make sure you protect the system with a hardware firewall that only allows incoming SIP connections from the ATT IPs. I used a Cisco Small Business router/firewall RV180. It allowed me to port forward only the SIP traffic from the ATT IPs. As all the GURUs in this forum say "MAKE SURE YOU PROTECT THE SYSTEM". In my case ATT needed the public IP of my router to send calls to. It's fairly a simple setup.
 

cespbx

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Thanks for the reply Charles. The public IP you are referring to is it the Border element IP ? AT&T assigned the following
AT&T Assigned Public Signaling IP Address: 172.x.x.x
AT&T Assigned Public Media IP Address: 172.x.x.x
IP Border Element Assignments: 12.xxx.xxx.xx and 12.xxx.xxx.xx
 

Charles Berghuis

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In my case ATT gave me 5 IPs and they just needed to know witch one to send the calls to. It's just the internet IP of my router. Some ISPs refer to this as a public IP. On my setup ATT gave me 2 IPs that they are sending calls on. When I did mine ATT did work with me to get it working. What version of PIAF are you using? If you installed the Incredible PBX you'll need to go into WebMin and allow the ATT IPs to come in, in the Linux Firewall. Are you doing a cutover or new service? And once again please make sure to protect the PIAF system and only allow the ATT traffic in.
 

cespbx

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Thanks Charles. I am using Elastix it has Asterisk 11.13
 

cespbx

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Thanks Charles and every one else for posting the information.
It worked perfectly and only took an hour to complete the test and turn up.
 

cespbx

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Charles

How did manage the faxes? The existing analog fax machines?
 

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