[Asterisk-Users] SIP codecs

[Dave Alan Caruana]

i've installed X-lite, can't get it to actually dial a SIP number,
seems cryptic compared to SJPhone ..

I have a feeling my problems is the codecs within *
though, my question was how could I know which codecs
* supports, and how to add other ones ..

cheers
Dave


----- Original Message ----- 
From: "Dan" <dtoma at fx.ro>
To: <asterisk-users at lists.digium.com>
Sent: Friday, June 06, 2003 2:27 PM
Subject: Re: [Asterisk-Users] SIP codecs


> If you have the package available for download for free from SJLabs, then
> you only have G.711 codec installed on SJPhone.
> If you are a developer, you can register for a G.729 codec from SJLabs.
> 
> BR,
> Dan
> P.S. Have you tried X-Lite? It has G.711ulaw, G.711a law, GSM and iLbc.
> 
> 
> ----- Original Message ----- 
> From: "Dave Alan Caruana" <david at melita.net>
> To: <asterisk-users at lists.digium.com>
> Sent: Friday, June 06, 2003 3:05 PM
> Subject: [Asterisk-Users] SIP codecs
> 
> 
> > i've been having a problem getting two SIP phones
> > to bridge running through asterisk, actually one is
> > a SIP softphone, SJ Phone, and the other is the
> > Go2Call calling gateway.
> > Someone suggested that I don't have the right codecs.
> >
> > How do I find out which codecs are installed, and how
> > can I install further codecs? Any suggestions which
> > would be the right one?
> >
> > I think hte problem is from the Go2Call side, not the
> > SJPhone cos I can dial from SJPHone to SJPhone
> > routing through asterisk with no problems.
> >
> > many cheers
> > Dave
> >
> >
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> 
> 
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