No audio using pjsip

Asterisk Asterisk SIP
1

I have a working IncrediblePBX install, but migrated to Fedora 34 on a Raspberry Pi 4 4GB, with Asterisk 18.2.0 and FreePBX 15.0.17.34.

I switched to pjsip for my extensions and trunks. They seem to register fine, (although I don’t know since I can’t get the info to show up in any logs). My Fanvil X5 phone now refuses to connect using chan_sip (although it did fine on the previous install), but can with pjsip. I’m using Callcentric and Anveo retail as trunks. Calls seem to egress and ingress (although I can’t tell since the logs show nothing, unlike in my previous install), but there is no call audio at all.

The FreePBX GUI times out regularly when applying the config, with a jQuery error.

I have no idea where to even start to diagnose this issue since logging seems to be minimal. Can someone help? I’m running out of hair to pull out.

2

There is no Asterisk version 18.20 (18.4.0 is the latest) and we do not support FreePBX on this forum.

Generally, if nothing is getting logged, you need to turn up logging until it is. REGISTERs will show on pjsip set logger on, if they are being sent or received at all.

You will need to take this up on the FreePBX forum, although it sounds like you have network quality problems.

3

Thanks for your reply. How do I log chan_sip registrations?

4

“sip set logger on” is the corresponding command for chan_sip.

5 
ast*CLI> sip set logger on
No such command 'sip set logger on' (type 'core show help sip set' for other possible commands)
ast*CLI> core show help sip set
No such command 'sip set'.
ast*CLI> help
No such command 'help' (type 'core show help help' for other possible commands)
ast*CLI> core show help help
No such command 'help'.

Now I’m completely bald with patches of bleeding scalp.

6

pjsip set logger on, instead of “sip set…”

7

The OP specifically asked for chan_sip:

8

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