Check param ext-rtp-ip in the external SIP profile. On Wed, Mar 10, 2021 at 12:35 AM Mircea Botoca-Huh <mircea.huh@gmail.com> wrote:
Hello,
I have freeswitch version 1.10.5 -release-17-25569c1631 64bit installed.
I have one SIP provider, no nat.
I use a simple dialplan which bridge the received call from SIP provider to a mobile number using the same provider.
The call goes through but I have no ringback tone and if I answer there is no audio.
If I send the call to one internal extension, it is working fine.
The only difference I can see in the logs is that, on the call with no audio, I do not have in logs „Correct ip/port confirmed”.
Any suggestion is greatly appreciated.
Thank you.
Best regards,
Mircea. _________________________________________________________________________
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