Firebox - VoIP and Video Conferencing — WatchGuard Community

VoIP dropping calls (go silent on both ends, but still look connected)

Logan5 — Wed, 03 Aug 2022 15:03:28 +0000

We have an M470 (12.7.2) setup with a VoIP VLAN, using 8x8.com as our VoIP provider and using Polycom vvx310 and vvx311 phones. They have worked great for years. Now we have calls drop (go silent on both ends, but still look connected) randomly throughout the day. The drops do not appear to impact multiple users at a time. There are only about 6 people in the office regularly. There is a an AltaFiber Fiber 1000/250 Gbps connection. I monitor the connection from time to time and do not see any outbound denials from the VoIP VLAN. I do have fairy detailed policies to allow and optimize VoIP traffic and taking this link into account https://support.8x8.com/cloud-phone-service/voice/network-setup-voice/x-series-technical-requirements. As a troubleshooting measure, I have allowed all outbound traffic from the VoIP VLAN. It does not seem to help. Could it be some kind of obscure timeout or global networking setting? Please let me know your thoughts and suggestions. thx.

Linkus client YEASTAR

Aantolkovic — Mon, 26 May 2025 14:04:36 +0000

Dear all,

I have a problem with linkus client. I open port on my watchguard 8111 and client is success connected to PBX but cannot make an call. I try change service port to another but it is the same.. With mikrotik gateway, everything is working.. Do you have that problems?

Thank you!

Video freezes in Teams and Google Meets

D4rkSeven — Tue, 06 May 2025 18:36:15 +0000

I am having problems with Teams and Google Meets calls, whenever the video would freeze and no longer play the following error would appear on the firewall:

2025-05-06 11:16:35 Member1 pxy 0xf32430-36129607 188: 192.168.32.220:59418 -> 74.125.250.245:3478 [A t] {R}: fatal write error (errno=1: Operation not permitted). aborting channel. Debug

2025-05-06 11:17:57 Member1 pxy 0xed4fb0-36131662 2104: 192.168.32.220:53470 -> 74.125.250.245:3478 [A t] {R}: fatal write error (errno=1: Operation not permitted). aborting channel. Debug

2025-05-06 11:27:25 Member1 pxy 0x1ba4ea0-36149841 1319: 192.168.32.220:52500 -> 74.125.250.245:3478 [A t] {R}: fatal write error (errno=1: Operation not permitted). aborting channel. Debug

What could be the cause of this behavior?

The firebox is a M470 with firmware 12.11

Asterisk PBX over Watchguard BOVPN VoIP Issues

Brans — Fri, 17 May 2024 16:03:17 +0000

Hi Guys,

Been having this problem for months and I can't seem to find the issue. Watchguard Support have been helping but TCP Dumps aren't showing the issue.

Setup:

Two sites both running M290's
1GB leased line at both sites, different ISP's
Virtual Interface (BOVPN) between the two.
3 Subnet networks passed over the VI. 1 PC Network, 1 Linux Network, 1 Phone VoIP Network
At Site 1 has a Asterisk PBX SIP Server
At Site 2 60xVoIP phones connect to this PBX over the VPN
SIP-ALG is disabled on both Fireboxes
Policy on both Fireboxes allowing any port to the PBX IP through the tunnel.
TCP MTU Probing set to always enable, on both WG's
DF bit Enabled on the BOVPN, set to Clear on both WG's

The Problem:

Most of the time phones work OK. However more than 20 times a day phones at Site 2 will ring with a internal or external call, the user answers, shows as answered, but no one on the other line. Hangs up, and the phone starts ringing again. Sometimes, if the user picks up the call, waits around 10secs, they can finally hear the other person.

TCP Dumps simply show ICMP phone unreachable when these problems occur.

No issues with the PC or Linux network through the virtual interface.

Phones at Site 1 where the PBX is located, don't have this issue.

Hoping maybe someone out there knows what I might be missing.

UDP Timeout

BazMac — Tue, 06 Feb 2024 19:30:30 +0000

Hi All,

Currently investigating an intermittent issue with Firebox running 12.10.1 with BT Cloud Voice Deskphones.

I have created a custom policy with the required ports for both signalling and RTP. Call work in both directions with 2 way audio. However the client reports that at random times the audio will stop but the call remains connected.

Therefore my feeling is this is some sort of timer. The one I am thinking of is one known as "UDP Timeout" with other vendors. On the Firebox I have the option for "Custom Timeout" which is unticked by default and when ticked I can set to 300s (the UDP Timeout setting requested by BT).

However is this the setting I think I am after? Some posts indicate that this setting only affects TCP whilst other tell me that UDP Timeout is only configurable globally via CLI?

An alternative approach is to configure a TCP-UDP proxy where I can configure the UDP Timeout from 30 to 300s However that doesn't allow me to get specific with ports?

Interested to know your thoughts.

Cheer

API call using Curl to SIP provider

lotty — Mon, 23 Oct 2023 20:35:16 +0000

is there a way to trigger a function call via API/Curl from the firebox to my sip provider (sip provider supports this ) related to failover event?

if so is there any WG documentation you could point to?

Thanks in advance

Need help to configurate the e-phone SIP solution from vodafone.

CML_Carlos_Luis — Sun, 12 Mar 2023 15:58:32 +0000

Can someone help me to configure the e-phone SIP solution from vodafone.
I try to apply a packet filter to the ports, a try to enable the SIP ALG, and no sucess.

https://www.vodafone.pt/content/dam/digital-sites/downloads/aplicacoes/one-net/manual-utilizador-e-phone.pdf

Intermittent Failure of Outgoing SIP Calls

IanW82 — Thu, 08 Sep 2022 09:50:27 +0000

Hi,

M270 Firebox, 12.5.4 fw

I'm wondering if anyone can shed any light on an issue I am having.

I am reconfiguring my Avaya PBX to use SIP for external calls, so far, incoming calls appear to be working without a problem but I am seeing an intermittent issue with external calls.

When calling a mobile, 99% of the time the call completes as expected. When calling a landlane, 99% of the time I have no ringing and no voice when the call connects however the recipient of the call can hear me fine.

I have used traffic monitor and I can see the call connection packets, in the instances when I get no audio, nothing in the logs indicate an issue or data being being blocked.

I have an open support case but I thought I would also ask here incase anyone has experienced a similar issue and managed to resolve it.

Thanks

Looking for someone to help with configuring a Firebox for VoIP

David0 — Wed, 25 May 2022 20:21:02 +0000

I have a client that is getting Intermedia Unite VoIP phone service (hosted PBX?) at their office. This is my first dealing with VoIP in an office setting. They have a T35.

Intermedia has a bunch of pages, including:

https://support.intermedia.com/app/articles/detail/a_id/15507/type/KB

on things that need to be set up on a firewall to ensure quality service.

No Watchguard products are on the recommended list:

https://support.intermedia.com/app/articles/detail/a_id/11411

And they have a page about issues with some firewalls and they talk about the XTMs. Those are old, right (ie their page is out of date?)

https://support.intermedia.com/app/articles/detail/a_id/11404/kw/watchguard firewall

This is way beyond my capabilities / I want to make sure it is done right.

Is this anything that Watchguard would do (gratis or for a fee)?

Or is there anyone more experienced here that would be available for hire to help take care of configuring the T35 and help me know what else I need to deal with (do I need to get the MAC addresses of the phones / set up reservations?, etc.).

Thanks!

Problem with PBX

Jonas — Thu, 05 May 2022 11:41:02 +0000

Hey, we upgraded our Alcatel lucent pbx to use voip. The Firebox is using pppoe to connect to isp. For that I opened some ports given by the isp. I did this by creating a static nat to pbx. Additionally I added the sip-alg proxy. All worked fine until last friday. Now calls ends automatically after about 10s and no voice is transmitted. I checked some logs and recorded a trace. It seems that the firebox is blocking all incoming rtp data. SIP in and outgoing passes. The firebox also transfers outgoing rtp data. It would be nice if someone could help me

Problem with FAX and Firebox (Shamrock Fax Software)

acon59 — Thu, 31 Mar 2022 08:21:37 +0000

The following info:

When you want to send a fax, you always get the message.
"Remote station is not a fax". (Gegenstelle ist kein Fax)
Both remote stations negotiate, one side says it is a fax, sends something back but nothing comes back to the customer.

If you call the customer directly, you only get a connection to the paper fax connected with the Fritzbox, but not to the Lancapi.

GoToMeeting Webcam Problems

Durrant — Wed, 04 Nov 2020 06:40:41 +0000

Hello,

so currently we are trying to run GoToMeeting for Home Office purposes.
Problem here is, despite nearly everything working fine, the Webcam seems to be disabled every time we try to join a meeting.

On the GoToMeeting Website there are some "How to's" for firewalls, with the
bad side effect of putting up an exception for +10k Ips (for Amazon Web Services)

We tried just to put the Domain names into the exception list, didn't solve the Problem,

Here is what we got so far:

We created a new Testrule, with a User that is allowed to Surf HTTPS/HTTP without restrictions, No Webblocker, no Geolocation.
Also we opened the Port 1853/UDP (Video) as well as 8200TCP/UDP (Which seemed to be a must have)

So this Testrule worked just fine and we could enable the Camera, yippieh.
Problem is, we can't let it stay open like that (Full HTTPS/HTTP).

I was hoping that anyone else ran into this kind of problem and might have a solution for that?

EDIT Important notes:
Logitech HD Webcam C525 -> a recommondation from LogMeIn (GoToMeeting)
Sound is always working aswell as Desktop sharing, only video doesn't work.

I've got the feeling that GoToMeeting is buildingup an TLS over SSL tunnel and is not amused that the Proxy inspects the content of the packet (Which it doesn't, the firewall setting in the Testrule is "Allowed", not "Inspect")

SIP-ALG Setting?

ChrisSnape — Tue, 30 Mar 2021 08:28:06 +0000

Hi,

Firewall: Firebox M270
XTM: 12.4.1.B595401

I am seeing the odd issue with our new VoIP/SIP solution in our office. I have been asked by the provider to make sure SIP-ALG is turned off on the firewall. Is there a setting for this anywhere for this? Or is it off by default unless a policy is setup?

Thanks.

Softphones connecting but no voice

Sokratis — Wed, 17 Jul 2019 08:30:34 +0000

We use a Panasonic softpone application for mobile users. In wifi network they work fine. In 4G network (through firewall) the application is connecting but no voice is passing through the firewall. All required ports from Panasonic are NAT to the PBX. No denied packets appear in traffic monitor. What shall we check?

Avaya Cm

Sgimtech — Sun, 05 Apr 2020 04:15:54 +0000

Hi Guyana

Just wAnt to a sked, have you everytime Configure SSL mobile VPN with avaya One x to connect to the US call manager, the setup was, mobile user Will VPN to the HO somewhere its there HO office, there HO office has direct connection to there call manager ni US using routing. WHich meant SSL VPN user virtual Pool Will re route traffic to US call manager.

Is there ang chance you share experience ON this setup?