I’ve got two extensions on the same phone on a test server, one pjsip and one chansip (I know). Everything was working fine until a few hours ago when the pjsip extension quit working. Checked that it wasnt banned, tried turning off the firewall, and for the heck of it I tried restarting asterisk with no luck. Rebooted the phone and then the chan sip extension fell offline too.
Rebooted the pbx, checked asterisk SIP settings, and changed asterisk from version 16 to 18.
In the full log I see a bunch of these (the pjsip extension)
chan_sip.c: Peer ‘1002’ is now UNREACHABLE! Last qualify: 0
even though it never registers.
I’m assuming this is some stupid setting in the PBX but I’ve had enough problems with Polycoms not updating settings when pulling config (or in this case not pulling a config at all). The phone is a VVX 310.
Have you check or try on device configuration to according to Polycom recommendation?
Enter a line identification address that the phone uses to register with the server. The address may include a user name, the host of the phone’s SIP URI, or the H.323 ID/extension. For example, if the phone’s line is [email protected], enter 1002 as the SIP where polycom.com is the server. Or, you can enter [email protected]. Any address entered will be displayed as the phone’s line if the display name and label are not specified.
I dont know if this was the only problem but I found that the Outbound Proxy setting was blank. I entered it manually and the phone worked again so I exported the local config and found the lines in the config that are not in the basefile. Might try adding those. I also need to look at the auto provision settings too, or whatever those are called in a Polycom.