We have some VVX411 phones and even an SPIP7000 that are on our current PBX that I’m trying to move over to FreePBX after the current system soft-failed on us.
When one of the Polycom phones registers, it immediately becomes unavailable:
[2023-11-27 17:48:53] VERBOSE[13865] res_pjsip_registrar.c: Added contact 'sip:<ext#>@<phoneip>:5060' to AOR '<ext#>' with expiration of 60 seconds
[2023-11-27 17:48:56] VERBOSE[13865] res_pjsip/pjsip_options.c: Contact <ext#>/sip:<ext#>@<phoneip>:5060 is now Unreachable. RTT: 0.000 msec
These are the only phones having this issue; our YeaLink and other devices register just fine.
The phones are on a different subnet across a site-to-site VPN from the PBX, but there’s no NAT involved. We are allowing all SIP and ICMP traffic to the PBX.
The default is much longer, possibly 3600 seconds. Did you set a short expiry because of this problem? If not, there is a SIP ALG in the path that is likely also causing your issue.
At the Asterisk command prompt, type pjsip set logger on
The Asterisk log will now include a SIP trace. You can see whether OPTIONS was sent correctly and whether a corrupted reply was received. Also, capture with Wireshark at the remote end to see whether the VPN connection is dropping or altering any of these packets. If you don’t have a smart switch there to mirror the packets, see
Turns out there were a few things set up that were affecting this; adding the subnet to the “local networks” in SIP settings seemed to resolve the registration issue, and I later found out that someone had set up a provisioning server on one of the networks as well so that was fun.