[Asterisk-Users] One way audio - it doesn't make sense
Michaël Gaudette
michael.gaudette at virtutel.ca
Mon Feb 6 13:05:01 MST 2006
Hi,
I've had a bit of a problem with one way audio, and it happens exactly when
I believe it shouldn't (and works perfectly when I would guess I could have
issues.
Setup:
GrandStream GXP2000-------Linksys Router-----------Internet------Asterisk
box (hosted somewhere, fixed IP, no NAT) ----------- VoIP provider
-------PSTN
When a call comes in from the PSTN, the call goes all the way to my desk
phone (the GXP2000) and it rings. Audio is clear, both ways.
When a call is made from my GXP2000 phone to a PSTN phone (I use my cell and
my home phone as benchmark, they both get the same result) then I get no
audio at all. but ti does rin on the PSTN phone.
I've tried rerouting ALL of the relevant ports on my Linksys router directly
to my VoIP phone (5060 for SIP, 5004 for local RTP on the phone, 10000-20000
as the Asterisk RTP ports)....Nothing works.
What ports am I missing? Could the problem be entirely something else?
Somehow I had the feelings that calls going out (since they originate from
the device behind the NAT) would not be a problem, but calls coming in could
be.
I really would appreciate a hint from somebody who knows better than I do
(i.e. anybody)
Mike
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