[Freeswitch-users] No audio on bridged call

[Dragos Oancea]

Check param ext-rtp-ip in the external SIP profile.


On Wed, Mar 10, 2021 at 12:35 AM Mircea Botoca-Huh <mircea.huh at 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.
> _________________________________________________________________________
>
> 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 at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210310/49239774/attachment.html>