[Asterisk-Users] SIP trunk problem

[George Vagenas]

Marty,

But with the same 128 bit upstream circuit, directly connecting the SJPhone
the Stun server and using ulaw,  everything is perfect. The problem comes
when i am putting Asterisk in the picture.


On 3/25/06, Martin Joseph <ast at stillnewt.org> wrote:
>
>
> On Mar 24, 2006, at 1:19 PM, George Vagenas wrote:
>
> > Hi all,
> >
> >  I have the following problem, working with a SIP provider, if i setup
> > my SJPhone to register directly to their STUN server and working over
> > a 384/128 ADSL i have a really good quality, but then if i configure
> > Asterisk to register to the same provider over the same 384/128
> > circuit the quality is REALLY BAD. The obvious difference is that
> > using directly the SJPhone i am using STUN, while when i am using
> > Asterisk to connect to my SIP provider and the SJPhone to connect to
> > Asterisk i have the following configuration for Asterisk.
> >
> >
> >  register => user:pass at sip.provider.com
> >
> >  [mysip]
> >  host=sip.provider.com
> >  type=peer
> >  qualify=yes
> >  username=user
> >  secret=pass
> >  nat=yes
> >  disallow=all
> >  allow=ulaw
> >
> >
> >  I am using Asterisk 1.2.3.
> >
> >  I think that i am missing something or misconfigure something because
> > for sure its not matter of the ADSL since in both tests i am doing i
> > am using the same circuit.
> >
> >  Any idea please????
> I don't think using ulaw on a 128K bit upstream circuit is a good
> choice.  I would use g729.
>
> Marty
>
> PS I can't be the stun server if asterisk is working, but quality is
> poor.
>
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