Hi Carlos
Le 07/08/2020 à 06:33, Carlos Chavez a écrit :> I am having a strange problem with a new provider. We already
> have a couple SIP trunks working fine. We are trying a new provider
> but we are having one way audio problems with outgoing calls. Incoming
> calls do have two way audio, only outgoing calls have this problem. I
> do not see anything odd with a packet capture and using PJSIP history
> to check. The provider says that on outgoing calls the get random
> characters instead of the media port for RTP.
>
> We are using Asterisk 16.12.0 with PJSIP. The server is behind
> NAT so we have external_media_address and external_signaling_address
> set to the public IP and all relevant ports are forwarded to the
> Asterisk server. The other SIP trunks work fine, only this new
> provider has a problem and only for outgoing calls.
>
> An rtp set debug on shows only outgoing packets to the media
> address but no incoming packets. Why would there be a difference that
> makes it work on incoming calls but not on outgoing?
We faced this problem and it was a firewall issue on our side. But if
you say that your provider doesn't get the RTP, I understand that they
can't return anything. RTP ports ?
Cheers
--
Daniel