I’m using FreePBX 13.0.182 (PBS firmware 10.13.66-12). Currently using flowroute trunk without issue dialing in and out. It’s actually quite wonderful , but…
Need to switch to TW SIP - (sorry, please don’t ask why the switch, it’s complicated). They’ve installed their ESBC (enterprise session boarder control) device and it works fine for them in a test. The device is on our LAN as a static IP - I’m able to ping it and confirm it’s there. From there the ESCB connects to their fiber box and goes out to the internet. They don’t have any setup documents for FreePBX, but have provided some sheet for elastix as guidance - can’t attach as I’m a new user here…
I’ve tried various setups based upon this sheet to get this to work but nothing is working. From what I gather my trunk setting needs to point to their host on the LAN which it does, and it uses the DID number as the fromuser. I’ve replicated the Trunk SIP Inbound and Outbound accordingly. I disable the flowroute trunk, call out and receive a all circuits busy from Freepbx. Inbound calls receive a message “due to network difficulties you call can not be completed as dialed at this time” from TW side.
I’m wondering if anyone else has this type of setup working? If so would you provide any info on how you got this working? My thought is I’m not using the correct DID number to get through the ESCB but I won’t know until tomorrow when my support contact is available.
Hi - Thanks for your reply! Here are the settings I tried.
In all cases 1XXXXXXXXXX = DID number from TW
Trunk setup -
Trunk Name - 1XXXXXXXXXX
Outbound CID <1XXXXXXXXXX>
CID - Allow Any CID
Sip Outgoing Trunk Name: 1XXXXXXXXXX-out
Peer Details -
host=192.168.1.XX <-- points to ESBC IP address
fromuser=1XXXXXXXXXX
type=peer
Sip Incoming
USER Context - 1XXXXXXXXXX
User details & register string - empty
For inbound routes I have a DID setup for 1XXXXXXXXXX
DID number - 1XXXXXXXXXX
CallerID Number - 1XXXXXXXXXX
Set Destination - Extensions - 150 Front desk
For Outbound routes
Route Name - 1XXXXXXXXXX-out
Route CID <1XXXXXXXXXX>
Trunk Sequence for Matched Routes : 1XXXXXXXXXX (no other routes selected)
I have since found out that the number I used is not a number in one of 4 “trunk groups” they assigned (found that out this morning). So I will try again tonight when I have access to the system with one of these 4 numbers. I will post my results positive or negative either way to aid others.
This is the text of what I was provided from TW in my first post which I couldn’t upload…
How to add a SIP Trunk from Time Warner (note it was for Elastix)
Go to “Trunks” and add a SIP Trunk
Name the SIP Trunk (I use the complete 11 digit number), also enter that number into
the CID field as well. 3.Name the trunk and the incoming context. (I use the phone number and put “-out” for
oubound)
4.Define the “Peer Details” as seen in my screenshot (same as I posted above)
5.Keep both the “Incoming User Details” and the “Registration String” empty
6.Make sure that you save and then hit apply
7.Check to make sure you can call in and out
XORCOM: - needs an additional value in the outgoing peerdetails: context=fromtrunk
I also tried this I found on the net to no avail -
IIRC, you only want to supply a CallerID number here to match inbound calls from numbers with that CallerID. So if you want all calls that come from a specific number to go to a specific destination, you’d enter that here… e.g. send all calls from a telemarketer with phone number 8001234567 to a VM box. So get rid of the numbers in CallerID and see what happens.
Just following up on this thread. In the end the setup problem was on TWC (now Spectrum) end. They had the wrong port setup on their SBC. Newer version of FPBX uses 5061 while their setup was 5060. Once that was discovered hours later it worked flawlessly.
I have 12 spectrum sips. do I set up the same info 12 times, just once and they handle 12 conversations, or what? I have it set up once, and inbound / outbound is working, but not sure if I can take 12 conversations simultaneously.
, but…