Hi, this is strange. Ran asterisk-version-switch on FreePBX 188.8.131.52 to go to Asterisk 16. After it completes, tried to run:
*CLI> sip show peers
No such command ‘sip show peers’ (type ‘core show help sip show’ for other possible commands)
*CLI> module show like sip
Module Description Use Count Status Support Level
0 modules loaded
*CLI> pjsip show endpoints
No such command ‘pjsip show endpoints’ (type ‘core show help pjsip show’ for other possible commands)
Tried asterisk-version-switch and went back to 13. The same thing as above. No sip, no pjsip
We use customized logrotate.d/asterisk file which gets changed automagically. The way to fix that is chattr +i. This caveat is, setting the flag will also prevent any asterisk update because when asterisk itself is updated, it moves asterisk to asterisk.rpmnew and puts in a default asterisk.
Actually, another FreePBX 14 server (184.108.40.206) we ran asterisk-version-switch on and chose asterisk16. Does almost same thing. In CLI> commands work however while calls come in and go out, there is nothing displayed in CLI> . Have sip debug at 10 and pjsip logger on. Also tried rtp debug on and nothing in CLI>.
Have other PBX’s with a version of Asterisk16 which work just fine. Just one pbx is not able to run 16. No clue yet as to why.
@danardf Just save yourself the trouble and stop here. You won’t get straight answers. We dealt with this in IRC yesterday. What is missing here is the fact this was first presented as a one-way audio issue and then the claim was no RTP could be see flowing through the PBX. Followed by how nothing is showing up in the CLI but calls are working.
So first we double checked that the OP was in the right PBX because the OP has a history of doing something like this and then saying “Opps, I was in the wrong PBX” while doing troubleshooting. Then while multiple people were trying to get more information and offer up suggestions the OP just said “I’m switching them back to 13. Oh it worked. I’m done” and left it at that.
ZERO troubleshooting or efforts to do any level of troubleshooting outside of turning on some basic Asterisk debugs. No tcpdump, no sngrep, nothing to prove that traffic was hitting the PBX. No ‘fwconsole chown’ or ‘fwconsole restart’ just to make sure ownership and permissions were correct or things were restarted properly and in proper order.
Also no confirmation that this system isn’t the same system that this thread was started about and that switching to Asterisk 16 was messed up due to files being chattr +i on the system.
Oh and as well as a history of doing things in the wrong PBX, the OP also has a history of making modifications to a PBX and then forgetting about them and after hours of working on something remember “Oh yeah I did this” and then the problem is fixed.
Franck, no there are no PCI cards whatsoever. As mentioned, not sure why this one FreePBX Server exhibits this behavior. We have several other FreePBX severs running Asterisk 16.x.x and they don’t have any issue at all.
In as far as any custom, the only customization is in logrotate.d and that would not cause this issue.
Hi Franck, one other thing should be mentioned. When this particular FreePBX was running Asterisk 16, I called the DID on that PBX and dialed the owners extension. When the call was sent to the follow-me TN (cell phone number) there was one-way audio issue. The owner could not hear me. Repeated the call 6x and all calls were the same, having the one-way audio issue as well as nothing showing in CLI> . Did have rtp set debug on, sip set debug on, core set verbose 10.
I can’t see how there is an issue with audio when asterisk 13 is running and no one complains about audio. In this one server, install 16, dial company TN, dial extension, follow-me kicks in, the one-way audio is present.