A main project in BerkeleyTIP is to put together our own VOIP conference server. A likely candidate is the Asterisk system. Videos: We've had at least one, & it should be listed here. Currently, BTIP uses the Ekiga.net voip server, located in France. AFAIK, we've never had a problem with it. But, it could be unaccessable at some time. Having our own serverer would give: 1) Backup reliability. 2) Customization ability. Customizations: 1) Control panel, to see who is in the voip channel, 2) Multiple channels, for multiple conversations. BTIP Asterisk Server Work Log 8/1/09: More or less initial configuration. Installed asterisk from source including DAHDI software timing module. Enabled IAX and SIP connections. Relevant files are: /etc/asterisk/sip.conf /etc/asterisk/iax.conf /etc/asterisk/extensions.conf Echo test active on extension 600. Conference active on extension 8000. Username and password for both IAX and SIP connections is 'btip'. Created 090422. |