Thank you for your answer. I have ext-rtp-ip set to correct IP address. Only the forwarded call is not working. The incoming to extension has audio. BR, Mircea mie., 10 mar. 2021, 15:05 Dragos Oancea <dragos@freeswitch.org> a scris:
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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The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
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