Goodmorning, the title says it all!
its been like 1 year and half since i managed to configure my FreePBX with some analog telephone lines, now i’m getting rid of those and i’m looking to switch to a VoIP system!
The point is that i have basically 0 knowledge of SIP TRUNKING and i have no clue on what to do/where to start
You already have experience with SIP trunks – that’s how your PSTN gateway is connected now.
Please post a copy of the documentation provided by Colt. Mask any personal details such as account numbers, usernames, passwords and phone numbers, but keep the structure intact. We can then tell your where to enter those fields in FreePBX.
Also, if they supplied any equipment (such as an SBC) at your location, provide details.
Goodmorning Stewart, i’m sorry if i’m replying only after 3 days but i dont wokr on the weekend ^^
Anyway, indeed after i experimented and tested with FreePBX and managed to make the system works for my office, i have some experience, but is really low!
I dont know what an SBC is, but i can tell you that for obv reason, with theyr new internet VoIP line they supplied a new router, nothing special i would says
Teorically i know that i just have to set up a new trunk on the PBX, change the in/outbound routes and the work should be done…
this below is the trunk i use to connect with the Gateway
I am assuming that the box from Colt is an SBC (Session Border Controller) used only for VoIP. If it is also your gateway to the internet (for non-VoIP traffic), please provide details.
The usual setup is to have a second NIC on the PBX, cabled directly to the Colt SBC, set up with a static IP address given by Customer PBX IP. If you don’t have a second NIC but do have a managed or smart switch, you could set up Customer PBX IP on a VLAN and configure the switch to send that VLAN untagged to the Colt box.
If the COLT SBC Signalling Point is not on the same subnet as Customer PBX IP, please explain. If RTP Media IP is on a different subnet, you need to set a static route in the PBX (at the operating system level, not in Asterisk or FreePBX) to that subnet.
In Asterisk SIP Settings, set Local Networks to include the Customer PBX IP subnet and the RTP Media IP subnet, in addition to what’s there now.
I recommend a pjsip trunk with
Authentication: None
Registration: None
SIP Server: (SBC Signalling Point)
SIP Server Port: 5060
To get started, set Outbound CallerID for the trunk to your main number (starting with +39) and set CID Options to Force Trunk CID. Set up a catch-all route to an extension for testing.
After you get it working, you can send different caller IDs and route different DDIs as desired.
If you have trouble, at the Asterisk command prompt, type pjsip set logger on
make (or receive) a failing test call, paste the relevant section of the Asterisk log at pastebin.freepbx.org and post the link here.
Yes, its also for non-VoIP traffic, so what kind of details should i search and report? (Customer PBX IP is the same as the IP of the Colt Router supplied by them)
