[Asterisk-Users] update re. Grandstream + SIP + Echo problems ..

[WipeOut .]

You should be able to simply use either

allow=all

in sip.conf and then change the codec order on the phone..

or if you want a little more control you can put

disallow=all
allow=g729
allow=...
 
in the sip.conf and selectivly allow the codecs.. you will probably still want to set the order on the phone as well..

You may need to make sure on the "allow=g729" syntax.. I typed this from memory so it could be wrong..


> has anyone got G729 and SIP working together?
> some config examples would help :)
> since I need to do this at a client where I don't
> really have internet access, or the will to root
> around mailing lists with the client breathing down
> my neck!
> 
> thsnk
> Dave
> 
> ----- Original Message -----
> From: "WipeOut ." <wipeout at linuxmail.org>
> To: <asterisk-users at lists.digium.com>
> Sent: Thursday, September 04, 2003 7:13 PM
> Subject: Re: [Asterisk-Users] update re. Grandstream + SIP + Echo problems
> ..
> 
> 
> > > now .. i have one slight problem left .. although most of my SIP
> > > phones are on a LAN connection with the asterisk server,
> > > there are two phones which are at a remote office bridged to
> > > my LAN via a 128k point to point ADSL .. these do not seem
> > > to be working well, you do hear speech but the remote person
> > > (dialled over PSTN through an X100P) hears it low and garbled ..
> > > I am assuming it's due to the delays in stuffing 64kbits (of g711)
> > > over a 128k link and was thinking of switching to G729.
> > >
> >
> > Rememeber G.711 is 64Kbps without overhead.. Actual bandwidth required is
> somewhere around 87Kbps.. On a 128Kbps line if anything else uses the line
> you will get clicks and breaks in the transmission..
> > --
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