I'm, the one that is anxiously waiting for this feature....
I've narrowed down options to do this on following:
- Pluto Linphone code that is currently in review by linphone developers and will maybe be integrated into main project code - this seems the easiest way, but could last long....
- Mythphone (I had it installed- it registers with Asterisk, but didn't try any video call yet) - on the other hand is connected to Myth, that is maybe out of future focus for Pluto. But maybe source code without myth GUI would be good start...
- Gnomemeeting (since we have Asterisk - it can connect to Gnomemeeting as H323 client. Recently SIP support has been added to CVS, so it could be good candidate - but I'm not sure of what option there are to remote control that application - it has DBUS component that is meant for this purpose newbielink:http://www.freedesktop.org/Software/dbus [nonactive] ...)
instructions on gnomemeeting and Asterisk :
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
"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
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
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 :-)
CVS for gnomeeting is here :
I'm writting this to list possible options from my point of view and maybe open some discussion - also to attract other users to help with opinions or development.