[HELP] Debugging PJSIP/CHAN-SIP for outbound calls

Hello PBX redditors,

over the last week I have tried in my off time to setup the "easiest" possible configuration I could try. I have one router with RTP ports 30000-31000 routed to the FreePBX/Asterisk Server (nothing else). I have registered 1 Trunk with the german telekom. I have configured 1 Inbound Route that goes to the MicroSIP softphone extension on my PC and I have tried to make the Outbound Route work (dial pattenr X.).

The Inbound Call works and transmitted Audio without Problems.

The Outbound calls alsways "time out" and dont even ring.

I have done this for both chan-SIP and PJSIP with exact same results.

I tried to debug the issue with the asterisk CLI but the messages there sadly dont tell me much, and I hoped some people here might have had similiar issues and solutions, all I found online or tried myself has not yet worked.

Any Information needed to resolve the issue i would gladly supply.

FreePBX 13.0.195.4

Asterisk Version: 13.12.1

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[SOLUTION] edit:

Many people find threads such as this and most of the time solutions are not posted, with the following sentences i want to avoid that.

The actual problem in hindsight was, that the softphone I used (microsip) was able to recieve calls without issue but could not make any outbound calls. In the end the softphone did not even make any contact with the asterisk PBX when trying outbound calls (no logs nothing), which indicates a faulty setting on my part or a capibility issue of the softphone.

I changed to a softphone (3CX for wndows) and managed to set it up so it rings correctly. IMPORTANT HERE was that i had to designate the RTP ports on the softphone aswell because the calls would break upon answering otherwise.

Inbetween I have upgruaded FreePBX and asterisk to Versions 14 and 15 respectivly but I believe this would not have been necessary.

I want to give my thanks to the redditors here for multiple suggestions that even if not all solving the problem at hand gave me points to investigate from further as this might make me able to help others in my position in the future.

thanks

TLDR:

It was not a trunk issue, it was not a dialplan issue,

it was a softphone issue