I have installed FreePBX using Incredible PBX on a cloud server, having successfully tried out FreePBX at home on a Raspberry Pi.
I have one trunk, which is to Voipfone (a UK based service). I got this working on the Raspberry Pi, but on the cloud server I get the “circuits are busy now” message when attempting a trunk call. I have checked the two installations and as far as I can see, the trunk has been set up with exactly the same parameters (including the outgoing caller ID, which I know from reading other posts can be a cause of this problem). The SIP connection is registering OK all the time.
I also tried an incoming call, and this didn’t seem to work either.
Thanks for the reply. The echo test works fine. I have just done a test call and captured the asterisk log. Please go to Asterisk_Log_2024-01-20.txt - Google Drive to view. Sorry but I am not yet exepienced enough to analyse/interpret it.
Cause No. 58 - bearer capability not presently available.
This cause indicates that the user has requested a bearer capability which is implemented by the equipment which generated this cause but which is not available at this time.
make sure you are asking for an acceptable codec and your CallerID is acceptable
Fair comment but actually 200 is a valid number on my own Voipfone account. Tried a full national number to make sure, but still a problem. So I’m presuming the issue is something else.
I am not familiar with Voipfone, so here are some general things to try with your trunk settings:
Set Media Encryption to None. If Voipfone supports encryption and you want to use it, we’ll troubleshoot that later.
On the Codecs tab, enable only alaw and ulaw. If Voipfone supports wideband calling e.g. to/from VoLTE mobiles, we’ll troubleshoot that later.
Try setting From Domain to the same value you have in SIP Server.
Try setting From User to the same value you have in Username.
Possibly, the caller ID format is incorrect. They may require 02030225682, 442030225682, or +442030225682. If they don’t have good documentation, use an incoming call as a guide. However,
you also have an incoming issue, so maybe we should debug that first.
They may require RPID or PAI, try sending both.
For incoming, does anything appear in the Asterisk log when you attempt a call? If not, run sngrep and report what, if anything, appears there. If also nothing there, report router/firewall make/model and any VoIP-related settings you have made.
If you still have trouble, at the Asterisk command prompt type pjsip set logger on
make and/or receive a failing call, paste the relevant section of the Asterisk log at pastebin.com and post the link here. Please don’t use Google Drive or other services that require a login, because some of the most helpful readers of this forum don’t want to be tracked in that way.
Of the changes suggested, I reduced the supported codecs to the two and also set the ‘From User’ to match the user. I don’t know which one did it, but outgoing calls started to work right away.
As for incoming calls, they still did not work at that point, but after a short while started to do so, seemingly of their own accord. I don’t know why this might be, but when I first prototyped the system locally on a Raspberry Pi, I observed exactly the same thing.
Anyway, I’m up and running at least for the time being, so many thanks to everyone for their help.