[Asterisk-Users] RE: Asterisk-Users Digest, Vol 19, Issue 47

[Michaël Gaudette]

That was exactly it! Thanks you VERY much!

Mike



----
For the sip setting in sip.conf that setsup your voip provider add:
canreinvite=no

On 2/6/06, Michakl Gaudette <michael.gaudette at virtutel.ca> wrote:
>
> 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