AT&T IP Flex Reach and 3CX

[jayfromit]

Hello,

My company has a 3CX server we use, with a T1 PRI. Most of the time it's fine however every time it rains it has issues because it's just copper wires. So we have decided to switch to the ATT flexible Reach service.
I'm fairly a newbie with VOIP/SIP. To my understanding how sip technology function is when configuring your PBX server all you need is 1 IP address then USER and PASS.

However the guys at ATT is giving me 4 IPs (Public Media Address, Public Signaling Address, Customer Media Address, and Customer Signaling Address). I have no idea what to do with these IPs. I can't find anywhere in the 3cx settings where it will accept 4 IPs and let alone not allow it to have a USER and PASS because ATT doesn't have any auth.

If someone can lead me to the right path that would be great.
I already googled ATT flex and 3cx and not much help popped up other then how easy it was to set it up. So googled 3cx and asterisk configuration and got this.
http://pbxinaflash.com/community/index.php?threads/how-to-at-t-ip-flex-trunk-configuration.9155/

 

[Pierre_3CX]

Hi there,
Please try the following :
- as a first step, ensure the Settings / Firewall checker passes all green, if not please adjust firewall/router configuration as per admin manual. Ensure your firewall allows SIP traffic from "Public Signaling Address" and RTP traffic from "Public Media Address.
- confirm that the "Customer Signaling Address" and "Customer Media Address" are your pbx public IP if not please adjust at AT&T.
- go in VoIP Providers/Add Provider, name the provider as you wish, Country: US, Provider: AT&T, Next
- SIP server hostname or IP, enter the "Public Signaling Address" provided, Next
- External number: your main number, Maximum simultaneous calls: the number of calls allowed on trunk, Next
- Define a default inbound rule, Next
- Define an outbound rule, Finish
- go in Ports/Trunks status, confirm the provider appears green
- test inbound/outbound call.

 

[JayP]

jayfromit,

I am having the exact same problem. Did follow the posted instructions from 3CX get it working? if not, did you get it working and how?

 

[cobaltit]

I actually worked with 3CX to get AT&T in as a supported VoIP provider so I'm very familiar with the setup. As far as the how though depends on the actual circuit. For all of our older installs AT&T would give you one of two options. Separate ethernet hand-off for the voice only portion which would require a second NIC but we would typically ask them to put it on the same NIC as the data. If you do it that way then you just treat it as a regular SIP trunk with IP authentication. This was when they were using Cisco routers. However, on the most recent installation using their newer (Business In a Box) which is their own router solution they wouldn't let my tech put everything on a single interface. Now whether this was because the guy didn't know how to or they just won't do it I'm not sure. And this was actually a move for the customer which was previously using the Cisco with a single interface. But the PBX had two NICs so we just did the standard 2 NIC setup.

Keep in mind that AT&T will put an ACL in by default blocking 5060 from the outside world so if you want remote phones via STUN you'll need to request them to remove this. Sometimes they will make you fill out the form and other times they will just do it.

 

[cobaltit]

Yep, that can be a hurdle. But like I said, they used to let us run VoIP over the single ethernet hand-off prior to the Business In A Box change. I'd talk to the rep and see if that is an option before you invest time into the 2nd NIC.

 

[cobaltit]

No RRAS need. Multiple NICs yes and static route maybe (Depends on where the SIP server IP is). My guy who did the business in a box install doesn't remember anything and I don't have remote access to that box to check. But typically you have your internet NIC with a default gateway and no default gateway on your VoIP NIC. The SIP server is either the local device in which case you won't need a static route because the SIP server is the local device or if it is remote then you will setup a static route to that IP or IP block out via the VoIP interface.

 

[cobaltit]

So you don't need RRAS for routes. route add .... -p will do what you need.

I don't think option 1 is possible because I don't believe AT&T will give you access to the BIB router. They want to hand off a public IP to your router. So with the option 2 method, you'll just create a new switch and uncheck the option to share the interface with the management operating system. This will hide the NIC from the host completely (no DHCP or anything). For the VM config, no DNS needed on the 2nd interface. You are talking directly to the BIB router so the interface just needs an IP on the same subnet as the IP AT&T gave you which is the 2nd interface on the BIB.