On 2007-01-19, Craig Matsuura <[EMAIL PROTECTED]> wrote: > Seems the 1.2.0 version was working better than the 1.4.0 - 1.5.1 version of > linphone. I think I used some of this versions and it worked well.
> > When using asterisk as the exchange, the calls appear to route but now > sound. Try to switch re-INVITE functionality on Asterisk (canreinvite=no) and on linphone to select alaw or ulaw format. I had problem with speex, becasue Asterisk tried to convert the stream and it was a lot of bad. -- Petr > On Thursday 18 January 2007 1:01 pm, Craig Matsuura wrote: >> I looking for some of the basic setup for linphonec to connect (use) a >> local asterisk server. >> >> My understanding is I have to setup a proxy server (proxy add) to my >> asterisk server? I'm unsure if I am doing it correctly as I can not call >> the other sip phones in my network. I had success with minisip, but prefer >> to use linphone as it is much smaller. >> >> I can dial directly from one linphone to another, but rather go via the >> asterisk server. >> >> Examples of a working configuration of asterisk and linphonec would be >> greatly appreciated, as well as usage (assuming I using linphonec >> incorrectly). >> >> Thanks, >> Craig >> >> >> _______________________________________________ >> Linphone-users mailing list >> Linphone-users@nongnu.org >> http://lists.nongnu.org/mailman/listinfo/linphone-users _______________________________________________ Linphone-users mailing list Linphone-users@nongnu.org http://lists.nongnu.org/mailman/listinfo/linphone-users