Calling DID does nothing, no log info

freepbx-15
Tags: #<Tag:0x00007f701f0fc2d8>

(Open Pbx) #1

I setup FreePBX using the latest ISO and then signed up for the trial and it auto added the trunk info, routes, and possibly other items. I could not call out or in so after some research I was able to fix outbound calling, but stuck on inbound calls not doing anything. When I watched for outbound I saw a ton of data to help resolve the issue, but with inbound nothing shows up. Port issue perhaps? I edited the “set destination” to be the 1 extension I have setup and working on my phone using a softphone and also tried creating a ring group, but still I see nothing happening what I call the DID i was given. When I call my DID it will say “calling” for quite some time and then just fail so I am thinking there is a missing setting or a need to open a port. I have 1 extension that I setup on sangoma connect and that is where I was able to test calling out. Just never get any incoming and the server shows nothing when trying to call.


(Itzik) #2

Assuming you are referring to SIPStation?

By doing what? What was causing the issue? It might be related

Where did you edit that?

Where are you looking? Asterisk console, full log or sngrep?

Assuming you are using SIPStation, have you read the Wiki? https://wiki.freepbx.org/display/FPG/SIPStation


(Open Pbx) #3

I was able to fix outbound calling - no trunks were chosen for outbound
I edited the “set destination” to be the 1 extension - in inbound
I see nothing happening - everything, gui log view, console, nothing shows any activity when calling in while calling out shows plenty of data


(Itzik) #4

Do you see any activity on sngrep?


(Open Pbx) #5

I will try it and post, otherwise I literally see null on gui log view and console asterisk -rvvvvv while when I make a call I see floods of info that helped solve calling out which was no trunks were listed in the auto-created outbound/int/911 routes when signing up for the trial through sipstation, the inbound route was auto-created as well


(Dave Burgess) #6

Inbound calling and outbound calling are almost completely unrelated.

Your trunk is set up, so SIPStation should be sending your DID calls to that trunk. The trunk will search your Inbound Routes, looking for one that will work. You should set up an “any/any” inbound route (not inbound DID and no inbound CID) and set your destination for that to your extension.

If you’ve done all of that, what errors are you getting?


(Open Pbx) #7

is the incoming sip settings under trunks suppose to be blank? I have sip settings for outgoing but nothing for incoming. It is best to remember I did not add any trunks or the inbound/outbound routes, when I signed up for the trial in the admin GUI during setup of the server it was all auto-setup/added and I wonder if something is missing.


#8

Blank incoming is normal, though if you are using registration, the Register String should be present.

In Asterisk SIP Settings, confirm that External Address and Local Networks are correctly set. If you change these, restart Asterisk.

Does Asterisk show the trunk as registered?

Run sngrep. Does any INVITE appear on an incoming call attempt? If not, post details about your router/firewall.