I tried to update my version 6 FreePBX system, which was running nicely and I should have left well enough alone, to Version 10. In the process, which seemed to complete OK, the system was hosed. I decided the best course of action was to just start over, so deactivated the old system, installed version 15 from scratch, reactivated and began configuring. Being a novice at this, I did not realize that CHAN_SIP had been replaced with CHAN_PJSIP, and after figuring that out, got phones to register, outbound and incoming dialing working. BUT this morning while working on the system, I realize that I had no audio in either direction, no ring indication on outgoing calls, internal calls rang and could be answered but no audio.
The system uses POTS on a dahdi interface through a Digium card. No changes. After researching all morning, many suggested a firewall issue, so I attempted to look at the PBX’s firewall, which is disabled. When clicking on “intrusion detection” the System Admin menu I get an error that says the system can’t retrieve the Fail2Ban blacklist. There is no firewall tab in any menu.
I don’t use any SIP trunks, only POTS but did enable ports 10000 through 20000 on the incoming system firewall, no change. Not sure how to check the NAT configuration, as there are no entries regarding it in the chan_pjsip tab, only in the chan_sip tab under SIP settings
I’ve re-installed the System Admin module and I do have the Sys Admin license installed. No change in seeing the firewall.
I’m going to change an extension over to the old CHAN_SIP config and see if that fixes anything while waiting for someone a lot brighter than I am to suggest how to fix.
Some missing critical details such as whether there is a NAT device between the phones and the PBX, but my wild guess is that the Local networks and external IP are not set correctly in Settings, Asterisk SIP Settings.
More details. External network IP is static and correct in the SIP settings. Local address also static and correct. I am running a network OS that does not have any NAT functions between the phones and the PBX, I don’t think, but that’s something to explore. It’s the same OS – ClearOS – that has been running all along with the version 6 box. The only change was going from a dynamic IP with Comcast and a static one as of last Friday, and the phones were all working after that change. As I don’t use any SIP trunks and all the phones are in the local LAN, I wouldn’t suspect the IP change would have caused this.
What other information can I supply to help with an answer? I’ll check on the NAT situation within ClearOS. Thank you.
I found another thread that suggested in sip settings chan_pjchan that the allow transported reloaded setting to be “YES” I now have audio. That’s sort of obscure, I think. But all now seems to be working.
The thread further states that this problem was fixed.
"On a new install “Allow Reload” in SIP Settings under the PJSIP tab should be set to Yes.
This module has been published and is now in the ;edge" track;"
Apparently, it hasn’t. This was a new install and this function was set to “no” Setting it to “yes” and updating the config immediately fixed the problem.
Look in the Dashboard. You will see an notification recommending that you keep this setting disabled. When you make config changes to transports and binding in Asterisk SIP Settings, you need to restart asterisk. Easily done at the command line with:
Making that change fixed the audio. I do see the message you note. Does that mean that setting it off and restarting Asterisk will cause it to remain fixed? Or that I should leave it on and restart Asterisk. I don’t understand.
I hate to be dumb, but your instructions are not clear. Should I leave the control on, reset Asterisk, and then turn it off to retain my audio function?. Or turn it off, reset Asterisk and expect my audio to remain in place? I’ve been fooling with this problem for several days now, and this is the first answer that has worked to restore audio.