Hello!
I’ve recently started working on a FreePbx project. After installing the latest FreePbx distro on google cloud following the steps on the FreePbx wiki I encountered a main problem.
I’m using pjsip for the trunk I’m using, as well as for the extensions(currently for testing, only one extension available). We want to use the PBX currently only for WebRtc softphones.
The problem is that when I’m using UCP softphone to make or receive calls, there’s no sound at all when calling, and when answering. I have to note though, that when I click on “Hold”, there is music on the other end playing as expected.
I searched many topics related to the issue, and found that might have been a NAT problem. I disabled the responsive firewall and let everything through the GCD firewall, but the problem is still there.
The setup uses a Let’sEncrypt generated certificate. I’m not entirely sure of what’s going on. The codecs I have are alaw and ulaw only.
On the other hand I tried using a cliend like Bria to see if maybe the problem was on the FreePbx UCP.
The client connects to the server just fine, but when I’m trying to make a call, it drops instantly and gives on the asterisk log the following error:
ERROR[22792] res_pjsip_session.c: 5000: Couldn’t negotiate stream 0:audio-0:audio:sendrecv (nothing)
Also I have to mention that I tried to use it on a jsSip project we currently have, and there I get the error that I’m unable to connect to WSS://x.x.x.x.:8089/ws .
In the link bellow you can find the asterisk log with sip debug enabled:
https://pastebin.ubuntu.com/p/JJHKqdWHgb/
Thank you!