Peer Status Unknown after reboot

We rebooted our VOIP phone system now some of our peers are showing status unknown while other peers came up fine and show status OK. What info can I give you to help me, I’m new with Asterisk. I have version Asterisk (Ver. Thank you

I’m going to guess they haven’t tried to reconnect. The PBX doesn’t control who registers and who doesn’t, except that it will reject phones that aren’t supposed to connect.

In other words, I’m guessing that the phones haven’t noticed they were offline and re-registered. Restarting them should solve the problem.

when we restart the phones they come back up unprovisioned

More details.

Phone type? FreePBX version? Actual error messages? Reason why you’re five years out-of-date (Asterisk 13 is out?

Tell us more, or we’re forced to ignore you because trying to help you is a waste of everyone’s time…

Thank you for replying. We are using Cisco 7960 phones. When I do SIP SHOW PEERS, some of the phones are working fine and say OK under Status while I have other phones that say UNKNOWN under Status. I am new to this company so this is what I walked into as far as the outdated version of Asterisk. I plant to get things upgraded soon as I don’t like working with old software. Let me know what other info would be helpful to you. Thanks again!

“Unknown” under status means that the phones have not re-registered.

“Unprovisioned” is something completely different, and is actually not really valid with the 7960 running in SIP mode. If I recall correctly, 7960s can’t use “provisioning” servers in SIP mode.

When you start the phones up (unplug and plug back in), what happens at the phone? There are three phases to booting the SIP version of the phone.

  1. VLAN detection.
  2. DHCP and BOOTP configuration.
  3. SIP registration.

I’m going to guess that there is something else going on with those phones that has rendered them disconnected from the network. It could be lots of places, including at the DHCP server, or the SIP settings for the individual phones.

Thanks Dave! I think the problem may lay in our switch, I think they are plugged into
the incorrect ports because all the network and SIP configurations on the
individual phones are correct and they are no longer showing UNPROVISIONED.
They have there extensions just can’t connect to any calls inbound or outbound.
I will keep you posted. Thank you so much for your tips.

I also have 1 phone that is stuck on TFTP Timeout and we can’t get past it to get
into settings even after a factory reset. Any ideas?

TFTP is how the phone gets its configuration, so there’s definitely something up with that phone.

Possible errors include bad bootp server address in the DHCP response or a permissions problem in the file system.

There are a lot of moving parts to using these phones in SIP mode, and they all have to work correctly for the system to keep on working.

On the other phones, if they are getting their configs from the TFTP server (or if they are using their last configured config) they will show their extension numbers locally, but may still have a problem registering with the phone server because of a network issue.

There are really too many places that you could be having problems to be able to narrow it down without following the basic network troubleshooting steps. From the sound of it, your assessment of a problem at the switch does seem plausible.

Just an update…we are still working on our issue, we can see all the phones from the switch, so they are making some connection. not sure what else it could be at this point.

is upgrading to the new asterisk a challenge since we have a much older version?

Check the IP address, netmask, and gateway address of the phones from the “Settings” button on the phone. There is also an entry for the registration server that you can look at to make sure the system is in the right place.

Also note that VLAN interaction could render your server/phones invisible from one another.

As far as upgrading goes - there are scripts available to upgrade your system to the latest version, assuming you started out with a FreePBX system in the first place. There’s a Wiki entry for upgrading that might help you.

got everything working, there was an incorrect setting on the extensions the NAT should have been set to No but it was on route

Thanks again for all your help Dave. Glad we are all working now. I have an issue with the display names now, on the phones under SIP Configuration I have the persons name under DisplayName but it still shows the extension number? any ideas?

I don’t actually use SIP on 7960s and 7940s. _ use the Chan-SCCP-B driver and use the phones in Skinny mode.

I know how to set the display names there. SIP: not so much.

Well thanks for your help. My next task is to try and get access to this phone that has the TFTP timeout. Can’t seem to get past it to get into settings even with a factory reset

do you know anything about this error I keep seeing in the Asterisks logfiles
sip_xmit returns error for pkt data

That’s not formatted like a normal Asterisk error, so it’s hard to say.

It’s an Asterisk problem, so obviously you’d be better served asking the question over there.

Of course you could try what I did: Google “sip_xmit returned error for pkt data” and start there.