Hi, i’ve been trying to setup a small FreePBX system and have mostly got it going after many bumps in the road. My issue is that, apparently randomly, incoming calls hit the PBX, i get the INVITE listed in sngrp but the extensions(?) don’t connect or respond… I have two Yealink phones. A lot of t…

[image] mrclunk: User-Agent: FireBrick/2.01.111 I suspect you have a SIP ALG (application level gateway) in your router. These mainly don’t work and should be disabled. I don’t see any logging of Asterisk;'s responses, or the dialplan execution.

There’s no SIP ALG on our network. User-Agent: FireBrick/2.01.111 Is on our SIP trunk providers network i believe. [image] david55: I don’t see any logging of Asterisk;'s responses, or the dialplan execution. How do i see these? thanks!

It also sounds similar like my thread… Without any resolution yet. [image] Packets not being processed by Asterisk on Debian 12 General Help Hi all, After migrating to FPBX17 on Debian 12, we randomly notice that inbound SIP packets (INVITES/REGISTERS/…) suddenly are not …

[image] dotcom: Do you see the call arriving in asterisk -rvvvvvvvvv ? I wasn’t but i’ve now turned debugging on in shell, even tho it was already set to on in the GUI… But typically, now i can see the logs, the calls are connecting ok… The logs showing up the asterisk CLI seem to be exactl…

Possibly my extensions aren’t connected properly despite reporting registered in the Freepbx GUI? Received SIP request (455 bytes) from UDP:192.168.20.6:5060 ---> SUBSCRIBE sip:101@192.168.20.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.20.6:5060;branch=z9hG4bK822737078 From: "Studio1" <sip:101@192.168…

That’s a SUBSCRIBE message for MWI, do you have a voicemail setup for 101?

I don’t have voicemail setup for 101 but a group voicemail for all extensions.

The output you showed is the phone trying to subscribe to MWI for mailbox 101. Turn that off in the phone, those go away. We need to see what happens when a call actually attempts to call the extension.

So here’s a log from a non working call. Now my local SIP ports on my phones are set to the same as the PBX server 5060, that’s correct right? Same as the ‘listen’ ports on FreePBX server. <--- Received SIP response (359 bytes) from UDP:xx.xxx.xx.116:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP xxx.…

We need to see the verbose call log from the console. This doesn’t show what happens within Asterisk. It very well could be rejecting the call because it comes from anonymous.

INVITE received but extensions not responding/connecting overtime

General Help

dotcom (dotcom) February 13, 2025, 5:29pm 4

It also sounds similar like my thread… Without any resolution yet.

Do you see the call arriving in asterisk -rvvvvvvvvv ?