I’m running FreePBX 12.7.6-1904-1.sng7 using the commercial EPM, on a system with 73 extensions. Most of these (about 65) are managed by the EPM; a few unsupported devices are manually configured. The system is fully updated in Yum and Module Admin.
Nearly all devices are Cisco SPA500 series. Recently we added six Grandstream GXP-1628 devices, configured via commercial EPM.
These devices successfully provision (verified in apache log) and have the correct information in the provisioning files for SIP server and ports. I am using PJSIP.
All six devices will only show “Unavailable” in SIP Peers. Attempting to dial one of these extensions from another results in the call going directly to voicemail. They can, however, dial voicemail, receive a password prompt from the server, subsequently interacting with the voicemail subsystem.
Suspecting a NAT issue in the router at the client site, we verified SIP ALG was disabled, and changed NAT settings to the recommended values for a SonicWall per the documentation that the site network admin referenced (I do not have that documentation). The UDP NAT timeout is 30 seconds in the router, so I changed Register Expiration to 1 minute, and ReRegister Before Expiration to 40 seconds in the Basefile Editor for the Grandstream template, this should result in re-registration before the 30 second UDP NAT timeout happens.
However, not even the initial registration is received when the phones are booted. I did reboot the FreePBX server as well, but no change in behavior occured after rebooting.
I’ve also verified that the site IP address is in the “Local” networks group in the Firewall, and is whitelisted in Intrusion Detection.
Any suggestions as to why the phones may not be registering at all?
Look through the /var/log/asterisk/full file and search for the extension number. That will give you the sequence of processing that is succeeding, then failing.
To be honest, this doesn’t really sound like a traditional RTP error. I suspect more of a routing issue, although depending on the router configs and where the phones are relative to your network equipment, it could be a problem with NAT not allowing the phone to stay registered.
It seems as if the router is closing the session. Make sure that your router is not closing the session before the phone re-registers.
Check the SIP ALG on router.
SIP ALG is disabled (which is nearly universally recommended to be disabled, from what I’ve found).
I also suspected the router of closing the session, but the re-reg timeouts should be shorter than the NAT timeouts, which should keep the session open.