[Asterisk-Users] SIP trunk problem

[George Vagenas]

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????
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