Help with remote Polycom VVX350 extension

I have PBXAct UC 25 behind a FQDN. Other phones have no issues connecting remotely and they are not unregistering at all. All sorts of phones I have provisioned and use remotely, i.e. Ciscos, Softphones, Grandstreams, etc. They’re setup with EPM.

I setup a VVX350 with EPM, it downloaded the config file, the buttons and everything are configured properly and when I make config file change it updates it correctly. HOWEVER the problem is that after a few hours, the phone cannot receive incoming calls from any extension anymore (the caller gets busy tone). The VVX350 can still CALL OUT without any problems. Just incoming calls give busy tone.
I’m guessing it’s some sort of registration timeout issue. Again the phone is setup fully using EPM (external, TFTP with FQDN) and I haven’t edited the basefile for it at all (it’s the default EPM provision settings for remote phone). FYI other phones on this same network are used without issue; just this VVX350 which cannot accept calls after a while.

UDPATE - the vvx350’s status of accepting calls is unstable I noticed; it just rang when someone called. Then after a few minutes the callers get busy tone when trying to call it again…

Kindly advise what to do… probably need to add a few things to the template’s basefile… Thanks

Anyone have any ideas what to try?

Is the PBX remote to the phones?

Have you narrowed the problem down to one phone, or is it possible that the problem is at one location? My wild guess is that the qualify packets sent by Asterisk are not getting a 200 OK back. If this is so, there will be reachable and unreachable log lines in the asterisk full log.

PBX is remote
Other phones (i.e. cisco, grandstream, softphones, etc.) have no problem connecting to PBX from this remote location and no issues at all

It is the polycoms that have unstable incoming calls
I narrow down to the phone

This sounds like a NAT problem. It can make outbound calls no problem but if it has no incoming audio or the PBX cant send calls/options to it means nothing is reaching it.

Bet this scenario works: Stops accepting inbound calls. Make an outbound. Next inbound call works. If this scenario works then 100% NAT.

Can’t be that. Router both locations is identical (the same exact router) and many other phones on the remote location work without issue.
The phone can always make outbound call however sometimes inbound call to it just gives busy tone, after a few hours of it sitting as registered.

Stops accepting inbound calls. Make an outbound. Next inbound call works. If this scenario works then 100% NAT.

No that scenario isn’t happening. It just doesn’t want to accept calls randomly after sitting as registered for a few hrs, and making an outbound call doesn’t fix it

You need to show the extension state when it is going busy. Need to see some debugs. There are numerous reasons why this could happen and most are related to NAT.

We need to see actual debugs/troubleshooting output in order to determine what is actually happening.

If you want to take a potshot at this issue, try changing the phone’ s registration expiry from 3600 to 120. In the VVX reg-advanced.cfg file, you would change
reg.1.server.1.expires=3600"
to
reg.1.server.1.expires=120"
However, this parameter may be called something else in EPM.

If that doesn’t help, you really need to find out what is going wrong. For starters, what appears in the Asterisk log (if anything) when registration is lost? What does Asterisk show for the peer status? On a failing incoming call, is there a timeout (Asterisk sends INVITE but gets no reply), or does it already know that the extension is unreachable?

I waited long enough for the issue to come back;

Attached is log file
what is happening is that the problematic extensions on the remote network are becoming reachable then unreachable
I noticed it’s also happening with other devices now. And call quality audio is very bad (digitized and silence pauses etc.) FYI the router which the PBX sits behind has UDP ports 10001-20000 forwarded to it.

(the only way I could upload the text log file is by zipping it to .tgz which is attached)

Thank you

logs.tgz (6.6 KB)

You will have to have your network figure out where extension 1135 is at, perhaps 10.65.10.3 or perhaps 192.168.2.202, it is unlikely to be at both places.

1135 is registered to a softphone

Something bad has happened, yes now I believe it’s probably NAT, because:

  • all phones on the remote network are having the unreachable/reachable reliability problems now
  • horrible voice quality if the call actually gets established
  • now most of the time the phones don’t even ring; just go to VM or busy tone or silence/extreme delay when placing call

Please identify what PUBLIC_IP_OF_REMOTE_LOCATION and REMOTE_IP_2 are, you need to fix your network, that is nothing that Asterisk/FreePBX can fix

Art you sure that extension 1135 and 1123 only exists in one place?

1135 is softphone app (groundwire) on an iphone; it is possible that it was on CELL connection and not wifi at the time of calls which is why it was named REMOTE_IP_2.

Just discovered something - I rebooted the router that the PBX is on and all of a sudden all the problems are gone. I am now sure it was router problem.

What routers/firewalls does anyone recommend that works reliably with FreePBX?

The problem was the router on the PBX’s network; it was causing these problems. Recommend you don’t use Asus SMB routers on network of PBX as their routers have caused countless problems with SIP devices i.e. Grandstream UCM, FreePBX, Talkswitch and almost any other system imaginable that I’ve tested on Asus router. They are known to build up huge log files which choke the network therefore deteriorating SIP calls quality/registrations/NAT etc.

This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.