SOLVED Difficulty Setting Up Callcentric E911

[Ramsey F]

I'm having trouble getting Callcentric set up as my 911 service. I'm using GV as my main service and it seems to be working well. I even have a rudimentary voicemail transcription I created using the IBM Watson Speech to Text service.

My initial setup was following the tutorial for the Callcentric trunk. Then I followed the Inbound trunk tutorial and the Outbound tutorial using the Vitelity example. Outgoing, I set up both 911 and 933 because 933 is the test number.

When I call the Callcentric DID, in the handset, I hear one ring and then a fast busy signal. The phone connected to the ATA does not ring. When I dial out to 933, I get a message that that person is busy or unavailable. I've done the setup twice with the same result.

Wazo is registered with Callcentric.

For Contexts, I'm using "from-extern" which is already setup with the correct 10 and 11 digit number pattern. And "to-extern" which requires no modifications.

For Callcentric trunk, I set it up just like the tutorial using my own Callcentric identifier.

For Call Management:Incoming, I set up with my DID pointed to my user, just like the tutorial.

For Call Management:Outgoing, I setup a new template using my Callcentric trunk and Exten 911 and 933.

What am I missing?

 

[Ramsey F]

I've been working with Callcentric and they say the following. Can anyone give me some help on how to make this happen? Thanks.

One issue that we do see with the SIP INVITE packet above is the "From" header your PBX is sending; specifically:

From: "Ramsey French" <sip:[email protected]>;tag=as4c5b7aa8

It should be sending your username(1777XXXXXXXX) @ our domain; something like:

From: "Ramsey French" <sip:[email protected]>;tag=as4c5b7aa

 

[atsak]

This is in your trunk setting. Post what you have in the trunk inbound / outbound, but sanitize it (ie replace your passwords and ID)

 

[Ramsey F]

Thanks for responding.

PDF containing SIP trunk, Incoming and Outgoing for Callcentric:

https://www.dropbox.com/s/upxjraefjks8t9y/trunk2.pdf?dl=0

 

[Ramsey F]

When I use Asterisk debug while calling 933, I get the following:

[Feb 14 21:59:50] -- Executing [dial@outcall:7] Dial("SIP/w7zxxxxx-0000001e", "SIP/1777XXXXXXX/933,120,o(933)") in new stack
[Feb 14 21:59:50] == Using SIP RTP CoS mark 5
[Feb 14 21:59:50] -- Called SIP/1777XXXXXXX/933
[Feb 14 21:59:50] WARNING[2181][C-0000001a]: chan_sip.c:23967 handle_response_invite: Received response: "Forbidden" from '"Ramsey French" <sip:[email protected]>;tag=as4cc845a4'

 

[Ramsey F]

Solved by Callcentric:

In Trunk > SIP > Advanced

Set "Rewriting the From field-User:" to "1777XXXXXXX"
Set "Rewriting the From field-Domain:" to "callcentric.com"

911/933 now works.

But for incoming calls, the caller ID is messed up. However, this is only a 911 account, so incoming caller ID doesn't matter.