asterisk users - Feb 2006 - One way audio - it doesn't make sense

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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> 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)
Pehaps you have set your device to use an outgoing codec which is not 
supported out of the box by asterisk, such as g.729? ulaw or gsm should 
work. Check your codec config in your sip.conf as well. For debugging 
purposes, you should use ulaw everywhere (assuming your ISP supports it).

Also, are you having any messages on the asterisk command line? Log onto 
your server, type in:

asterisk -r
set verbose 10000000000
set debug 10000000000

And let us know what you're seeing on the CLI.

Cheers,
Jean-Michel.

-- 
Jean-Michel Hiver - http://ykoz.net/
D?couvrez la R?union des Technologies IP & Telecom
TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

> > 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)
> 
> Pehaps you have set your device to use an outgoing codec which is not 
> supported out of the box by asterisk, such as g.729? ulaw or 
> gsm should 
> work. Check your codec config in your sip.conf as well. For debugging 
> purposes, you should use ulaw everywhere (assuming your ISP 
> supports it).
I tried, the only allowed codec in my sip.conf file is GSM, as supported by
my provider.  

My CLI doesn`t show anything special with debug turned on full. Just the
typical:

-- SIP/provider-0154 is making progress passing it to SIP/myid
-- SIP/provider-0154 is ringing
-- SIP/provider-0154 answered SIP/myid
-- Attempting native bridge of SIP/myid and SIP/provider-0154

Mike

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

On 2/6/06, Micha?l Gaudette <michael.gaudette@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
>
>
>
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