Re: [GnomeMeeting-list] SIP with Gnomemeeting and Phone to PC!

[Conrad Beckert <conrad_b yahoo com>]

Phantastic!! I'll try it once my gatekeeper is running
smoothly ;-)

Conrad

 --- Sam Lown <sam pagmas com> schrieb: > Hi,
> 
> For those interested, there is a way to get a SIP
> user to call any H.323
> device - Asterisk.
> 
> Firstly, I'll explain why this is useful. There is a
> service in the UK
> that gives you a Phone-to-PC gateway, but only via
> SIP. I am actually in
> the process of getting them to test H.323 with me,
> so with a bit of luck
> that will work soon and this e-mail will be
> redundant.... We're going to
> see SIP in GnomeMeeting soon anyway, aren't we? ;-)
> 
> The address is: http://www.speak2world.com, I
> discovered them totally by
> accident on a link in ebay.co.uk of all places!
> Their site basically
> says:
> 
> "Get a FREE UK 0870 [+44870] Number for your
> Internet [SIP] Phone today
> !!!"
> 
> And yes, you can call these numbers internationally.
> It works very well
> indeed with my setup... and it is completely free!
> (the callee pays for
> the cost of the call at national call rates) 
> 
> You might be assuming at this point that you need a
> quicknet card or
> something similar, NO!!! Their service is based on
> an Asterisk server
> and uses GSM!!! FANTASTIC!
> 
> Heres how I managed to set up a SIP to H323 gateway
> using Asterisk (not
> for the weak of heart):
> 
>  - Compile Asterisk with the chan_h323 module -
> you'll need to get and
> compile all the OpenH323 and pwlib stuff
> (www.openh323.org) - see the
> asterisk pages (www.asterisk.org) for more info.
>  - Work out how on earth to use and configure
> Asterisk - RTFM
>  - Add the following lines to the default context of
> the extensions.conf
> file: 
> 
> exten => s,1,Dial(H323/<ip of gnomemeeting
> machine>|30|r)
> exten => s,2,Hangup
> 
> Where r tells the asterisk to route the call, and
> leave the gnomemeeting
> user to answer. (This way the callee only pays when
> you are connected)
> 
> Of course, if correctly set up, it could also
> operate for NAT traversal
> with both H.323 and SIP; asterisk even converts
> compression algorithms,
> so a SIP call using GSM is converted automagically
> to G.711 on my
> ethernet phone.
> 
> I'll try and get round to setting up a web page with
> more details and
> example configs, unless some other site already
> exists!?
> 
> As I said, hopefully they'll add H323 support to
> their servers.
> 
> Hope this helps, or at least inspires someone :-)
> 
> Cheers,
> 
> sam
> 
> 
> 
> 
> BTW: I am in no way associated with speak2world and
> take no
> responsibility if you manage to break your computer
> by following my
> suggestions in this e-mail.
> -- 
> www.samlown.com
> 
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