Dialing internal extension fails


(snaplink) #1

I’m running FreePBX version 14.0.13.26 and I just started (after the server has been untouched for 34 days) having an issue dialing some internal extensions.

Example, my internal extension is 128 and calling 139 says the person at extension 139 is unavailable after the tone. It made me think they were on DND and forgot to come out of it but they’re indeed available. I checked endpoint manager and also on the handset itself which is a Yealink T46G. This is happening to three extensions that I know of but most of the others I’ve tested work correctly.

Looking in Call Event Logging using my extension as the source and theirs as the destination I see the call info and here is what I see. I’m guessing the s-CHANUNAVAIL is the issue? I’m not sure where to go from here to troubleshoot or how to fix this issue. Anyone seen this before?


#2

Can you find out if the problem phones can make outbound calls? It could also help to monitor some sip traffic for a phone with the problem to see if they are no issues with registering. Also, you can try rebooting one of them. If they can start receiving calls after a reboot, maybe they need to register more frequently to keep the connection alive. Are there any networking differences with the phones that work and those that don’t? For example, are they all on the same local subnet as the pbx?


(Lorne Gaetz) #3

Call trace may show the issue:
https://wiki.freepbx.org/display/SUP/Providing+Great+Debug#ProvidingGreatDebug-AsteriskLogs-PartII


(snaplink) #4

Thank you for the reply!

I’ve rebooted the yealink and even the freepbx VM itself which didn’t fix it.

Phones are on the same subnet as the freepbx.

I’m checking on the registration time but I’ve never touched those settings for any device.


(snaplink) #5

https://pastebin.com/BjdwLY1d

does this help?


(Lorne Gaetz) #6
[2020-03-17 14:34:39] VERBOSE[18645][C-00000024] pbx.c: Executing [s@macro-dial-one:55] Dial("SIP/128-0000003d", "SIP/139,15,HhTtrb(func-apply-sipheaders^s^1)") in new stack
[2020-03-17 14:34:39] WARNING[18645][C-00000024] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[2020-03-17 14:34:39] VERBOSE[18645][C-00000024] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)

Dial attempt is made, but peer is not registered, or not reachable. This is not a dialplan issue, its probably a signalling failure. You will probably find some history of peers going unreachable with:

grep UNREACHABLE /var/log/asterisk/full*

(snaplink) #7

I’m not every savvy with FreePBX, can you go into a little more detail on what a signalling failure is please?


(snaplink) #8

I did a sip show peers in the GUI, extension 100 is also one that I know that won’t accept a call.

What could this issue be?


(Lorne Gaetz) #9

There is no device registered to extension 100.


(Dave Burgess) #10

The status is “unknown”, so that’s why it won’t take a call.


#11

It could help to see a packet trace of a registration attempt from one of the phones.
https://wiki.sangoma.com/display/PPS/Capture+TCPDumps

I think for this case, you’ll want to do something like:
tcpdump -s0 -w/tmp/capture.pcap -C50 udp and port 5060 and host 129.33.194.122
where the host is the ip of the phone. After starting the capture, reboot the phone. After the phone boots up and a registration is attempted, you can stop the capture and share it here. Just make let us know the extension of the test phone and its ip so we can make sure we’re looking at the right traffic.