Toyima, Asterisk is a pain for making conference work because of its dependence on the dahdi kernel driver for timing. I like the FreeSwitch implementation. FreeSwitch has a very active irc at irc.freenode.net.
Dave On Thu, Jan 27, 2011 at 12:21 AM, Toyima Dias <toyim...@gmail.com> wrote: > > Anca, > > What conference system would you recommend for me? Asterisk? SEMS? may be > some licensed software could be ok for me...i would like to heard some > recommendations > > Regards > > 2011/1/26 Anca Vamanu <a...@opensips.org> >> >> Hi Toyima, >> >> On 01/25/2011 06:38 PM, Toyima Dias wrote: >>> >>> Thanks Anca, >>> To make transfers possibles, i would use REFER messages, and for this i >>> would need to use B2BUA module of opensips, rigth? >> >> VoIP phones can handle REFER messages, the problem is with the gateways, and >> this is why sometimes it is required to handle the REFER at the server and >> instead of forwarding it, replace it with connecting the party in a call to >> the new party from the server. >>> >>> What about conferences? For conferences is the same procedure (REFER >>> messages), but always handling a media b2bua, rigth? >> >> For conferences it is not enough to handle the signaling, you also need a >> conference bridge to distribute the media to more parties. >> >> Regards, >> >> -- >> Anca Vamanu >> www.voice-system.ro >> >> >> _______________________________________________ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
