Chris Chen wrote: > I have been playing freeswitch for a couple of weeks, and have been very > happy with its amazing features such as hardware-free conference, excellent > SIP stack and wonderful googletalk support. Guess what you can just use > googletalk client to join the conferences where SIP, IAX2, H323 or PSTN > extensions can meet together. Start now , add > [EMAIL PROTECTED] > <[EMAIL PROTECTED]>into your googletalk buddy list, > you can talk to many guys joining the Voice > Conference > > - IAX: [EMAIL PROTECTED]/888 > - SIP: [EMAIL PROTECTED] > - H.323: [EMAIL PROTECTED] > - Google Talk/Jingle: [EMAIL PROTECTED] > - PSTN: 1-213-799-1400 > > Although it still needs more work to have all required features working > before its 1.0 release scheduled at may 26, my freeswitch experience so far > has been nice excitement.
I really feel dirty defending Asterisk, however jingle/gtalk support was added in the 1.4 version, you can also do hardware-free conferences by twiddling the zap stuff, which is needed if you run asterisk in a Xen DomU/guest. However to redeem myself I will point out that freeswitch is using a SIP stack that is RFC compliant, where as the SIP stack in Asterisk is rather lacking in several areas. -- Best regards, Duane http://www.freeauth.org - Enterprise Two Factor Authentication http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://e164.org - Global Communication for the 21st Century "In the long run the pessimist may be proved right, but the optimist has a better time on the trip."
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