No Audio Changing default TLS SIP port

Hi folks,

I have been testing SIP over TLS and it work great for me in the default port 5061, but when I try to change the port , for example 5666, the extension register well but don’t have audio in the call.

I don’t think is RTP forward issue because it work well in the default port.

FreePBX 15
Asterisk 16.9

Any idea?

New port should be set as follow:
Settings --> Asterisk SIP Settings --> Chan SIP Settings --> Other SIP Settings:

This work perfect to me! Thanks!!!

I’m happy that your problem solved.
I suggest to mark this topic as SOLVED.

Sorry for my ignorance… how can I mark SOLVE?

Already marked solved.


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