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" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> 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" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > 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 > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
