Random Call Drops with Telekom (DE), NAT, dynamic IP and Yealink endpoints

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fsedarkalex

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Oct 12, 2023
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Hi!

This is my first post and I'm trying to be as detailed as possible, as I am working on this issue for several weeks now (Fun fact: I reinstalled FusionPBX about 5 times)

So... I am using FusionPBX (FPBX from now) for many years now. Some time ago, I hosted it on a vServer on the Internet, as it was only used for a specific number, nothing to do with my real telephony. This worked great.
About a year ago (or two) I migrated it into my home network, back then I was a Vodafone (cable internet) customer and I had a static-dynamic-IP (dynamic per contract but always got the same). The FPBX ran great here, too.
For the ext-sip-ip and ext-rtp-ip (naming might be a bit off, but you knwow what variables I mean) I was using "host:voip.mydomain.tld" which had an A-Record pointing to my public IP.

Then I switched my internet provider to Deutsche Telekom as I will move in a few weeks and at my new place, vodafone is not available.

I left everything as it was, and assigned the current public IP to the said subdomain. (This is technically connected to a dyndns service now)
I deleted the old Vodafone gateways, set the new Telekom gateways up (after some experimenting to find the correct values). I adjusted the dialplans and...
Telephony is working! Great! (I thought). I tested a few scenarios (Phone <-> Phone, Phone -> Mobilephone, Mobilephone -> Fixed Line Number -> Phone etc) and always got good results. Two-Way Audio, immediate responses (Pickup, Hangup etc)... My test calls were about a minute long (to test the obvious 32sec NAT issue)

The next day I was working in Home-Office. Guess, you already know what comes next:
I got a call and after about 3 minutes, the call ended unexpectedly. The rest of my day is TL/DR but long story short:
Calls keep dropping. Some after 6 seconds, some after 30, some after a few minutes up to even 30 minutes and a bit more.
The times seem to be random. If I REALLY want to see a pattern, it MIGHT be that some durations re-ocur, like it happens for example quite oftern after about 6 or 15 minutes. (never exact minutes though, it's like 15:03 or so)
But there are so many different times... Maybe this is still random "unluck".

The funny thing is, I can reproduce this by calling from/to my mobile phone, but this ONLY happens, when the call goes Phone <-> Phone. I can keep my mobile connected to MOH for an hour or more without a call drop. When taking the call over or back to my phone, the call drops after some (usually short) time.

What I tried so far...
- Using STUN at the phone -> I get no audio or one way audio on external calls, and not even signalling on internal calls.
- Using "stun:somestunserver" as ext-rtp-ip/ext-sip-ip -> No change
- Using the actual, external IPv4 address at this place: no change
- Using "auto-nat" (UPNP enabled at the firewall) -> no change
- Setting the freeswitch UDP Port range to a fixed (500) port range and redirecting this range to the PBX, re-iterating all the above tries -> no change
- Setting a variable "sip-force-contact" with value "NDLB-connectile-dysfunction" to circumvent the ACK-Via-Issue some phones have
- Setting "nat-options-ping" to "true" on internal and external profile
- Using TCP or UDP (tried both) for all registrations (internal and gateway)

Lately I installed HOMER/Heplify and tracked a call with it.
I can see THE ACK right at the beginning of the call and then it can take seconds, minutes or even tens of minutes and suddenly a BYE comes in from the provider (seems to originate from the far end if I an reading this right)
I will try to create a new capture soon and upload it for further reference, as I might read that wrong.

Any Ideas so far or maybe someone even knows the reason or even better the solution for it?

Thanks in advance to anyone who tries to help!

One more thing: The ppoint that it worked with Vodafone does not have to mean everything was set up correctly. Vodafone is a bit quirky, so maybe they just ignored the probably misbehaving on my end.
 

farshidhakimy

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Feb 24, 2023
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Hello,

I don't have much more experience than you do, but I would like to help.
Do you also sometimes have the issue that your sip trunk isn't registered anymore (Ansage "caller isn't reachable" when someone is trying to call the number)?
And which router are you using? Is it a Speedport, Fritzbox or a different router?
Edit: and what do you mean by "when the call goes Phone <-> Phone"?
 

fsedarkalex

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Oct 12, 2023
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Do you also sometimes have the issue that your sip trunk isn't registered anymore (Ansage "caller isn't reachable" when someone is trying to call the number)?

Hi! No I never noticed this. If so, at least the case would be more clear.

I am using OpnSense as router/firewall.

With Phone to Phone I mean a physical phone calling another phone. Not MOH, Mailbox or something like that
 
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