FreePBX external phones, sometimes no language/sound/speach


I am new here and would like to thank you in advance for all the information I could and can find here in the forum!
I understand the English language mostly, but I have my problems with writing.
I try to describe my request as objective and simple as possible.

If further information should be needed, I ask for indulgence.
I have only been working with FreePBX for a few days and I do not know a lot of things yet, but to the best of my knowledge I will deliver everything.

My external phone number terminates on a Fritzbox 6490 Cable.
In the FB is a LAN/WAN (IP phone) configured, username is area code + landline number (025xxxxxxxxxx).
The FreePBX (15 0 16 49) is as a VM on a KVM behind the Fritzbox and behind an OPNSense!

WorldWideWeb - Fritzbox - OPNSense (exposed Host in FB) - KVM Host - FreePBX (VM)

In the OPNSense all UDP packets from external to port 5260 are forwarded to the FreePBX port 5260 via port forward.
In the OPNSense all UDP packets are forwarded from external port 10000:10100 to the FreePBX port 10000:10100.
In the OPNSense all TCP packets from external on Port 84 to the FreePBX Port 84 via Port Forward.

The firewall of the FreePBX is disabled.
The LAN/WAN (Ip-phone) of the FB is configured as pjsip Trunk.
Five Sangoma S406 phones are registered in the portal of sangoma (Im not allowed to write a link) including redirection to the static IP of the Fritzbox, Port 84, plus “Phone Provisioning Password”.
Accordingly, five extensions of the type “sip” (Port 5260) are set up in the FreePBX, which are in a shared Ringgroup.
Each extension is bound to a Sangoma phone via EndPointManager.
All telephones use the same template according to EPM.

The telephones are all located in the home office of the employees.
So they each arrive at the public IP of the Fritzbox via their own dynamic, public IP!
The provisioning to the phones seems to work - times and line keys are displayed correctly.

External calls are ALWAYS delivered to all phones via the ringgroup. All external telephones ALWAYS ring.
Internal calls from employee to employee (homeoffice to homeoffice) are ALWAYS delivered to the appropriate phone. The telephone concerned ALWAYS rings.

The problem is the language (sound? speach?)!
Without changing anything in the system, the remote stations can sometimes talk to each other and sometimes not.
It can happen that I successfully make a call now and five minutes later on a new call the participants cannot hear each other anymore.
After an indefinite time everything works again.

In the asterisk logfile you will find the following for example:
[2020-04-20 09:22:45] WARNING [24718] chan_sip.c: Retransmission timeout reached on transmission [email protected] 168 0 39 for seqno 21 (Critical Response)
2020-04-20 09:22:45] WARNING [24718] chan_sip.c: Hanging up call

The 192 168 0 38 has the Sangoma S406 in my private network.

An external STUN/TURN (coturn) unfortunately did not help.
It looks like it is only used by the FreePBX.
Often it is so that calls can be led again after an “Apply Config”. But unfortunately also not always.
In the OPNSense nothing is blocked according to the log. All rules are set accordingly to write logs.

Why do phone calls sometimes work and sometimes not?


the media (udp traffic) will be pair of ports starting with an even one between 10000 and 20000 by default, they are defined in /etc/asterisk/rtp_additional.conf , you can change the range in the GUI or expand your forwarding range to match eachother

Thank you very much for your answer!
The ports are already set up accordingly.

$ grep -v ‘^;’ /etc/asterisk/rtp_additional.conf

sngrep for the next step maybe?

Thank you very much for the reference to the tool.
I’ll see what I can do with it.

Add external phone local lan network in Advanced- SIP setting portion.

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