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?
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)8116-9161
--
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