That's strange , i am able to see the *url* in Martin's reply . On Sun, Mar 23, 2008 at 6:14 PM, Eric Wieling <[EMAIL PROTECTED]> wrote: > The only messages I have EVER seen Digium remove from the mailing list > archives are discussions about this unlicensed codec. > > > Martin wrote: > > Download an appropriate binary from > > [url removed] > > > > and just drop into /usr/lib/asterisk/modules/ > > add allow=g723 to your sip.conf as necessary and restart asterisk... > > Im only not sure how legal is this, you will probably need to obtain > > licenses for all concurent channels... > > Martin > > > > ----- Original Message ----- > > From: "wassim darwish" <[EMAIL PROTECTED]> > > To: <asterisk-users@lists.digium.com> > > Sent: 22. brezna 2008 15:21 > > Subject: [asterisk-users] G723 on asterisk 1.4.1 > > > > > >> Hi: > >> How to install and set up my asterisk server with G723 codec to send and > >> receive calls using it. > >> > >> Thanks in advance; > >> Wassim > > > > _______________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > -- > Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, > T-1, PRI, Frame Relay, Linux, and network design. Based near > Birmingham, AL. Now accepting clients worldwide. > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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