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
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