[Asterisk-Users] SIP trunk problem

[Martin Joseph]

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.