[Asterisk-Users] About Audio Latency from PSTN to SIP

[Andrew Kohlsmith]

Please don't post HTML to the list, and PLEASE TRIM your posts!  Maybe I'm 
getting oversensitive to this lately but the sheer volume of bandwidth wasted 
due to people not taking 30 seconds to trim replies is staggering!  My reply 
is an example of proper reply trimming; only the essential bits from your 
post are retained, and everything else is deleted.

On April 15, 2005 04:12 am, Qiao Yuansong wrote:
> > put
> >
> > disallow=all
> > allow=ulaw
> >
> > in sip.conf, under [general] and comment out all other allow/disallow
> > lines. Restart asterisk and try again.  Something basic is not right.
>
> I tried your suggestion, and it make no use.

So you have 

[some_sip_user]
type=user
disallow=all
allow=ulaw
context=somecontext

in sip.conf for that sip phone?  Can you post the output from the sip phone 
dialing a PSTN number, and then the output from a PSTN incoming call ringing 
the SIP phone?  What version of asterisk?

Perhaps you should check out  
http://www.catb.org/~esr/faqs/smart-questions.html while you're at it.  We 
can't help you if you're not willing to help us.

-A.