On Tue, May 6, 2008 at 10:08 PM, Duane <[EMAIL PROTECTED]> wrote:
> 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]>
> > <[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.
I have the gtalk support in my asterisk 1.4.19.1 ( and started using
gtalk since 1.4.8)
However, I have to admit that the gtalk support in Asterisk is really
limited to client mode basically you have to define each buddy for each
gtalk account, while it is lacking the documentation( if there really
exists) for supporting the gtalk/jingle component mode. The freeswitch is
really shining on the complete supports of gtalk in both client and
component mode, which I likes the component mode most because you can handle
the jabber messages for the whole domain. And gtalk support on freeswitch
can easily be application bridged to conference, where you can have many
many gtalk client joining the conference.
>
> 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.
Just ask yourself how many extra settings you have to complete in sip.conf
and extensions.conf for you to get the BLF/Parking etc working in asterisk
1.4 or 1.2, and other features which have been on the long wish lists of
multi-tenant, or multiple parking lots etc, compare these to the freeswitch
by your own experience, you can come to your own conclusions.
Best regards,
Chris
