[GnomeMeeting-list] Transferring calls to asterisk
- From: don Paolo Benvenuto <paolobenve gmail com>
- To: GnomeMeeting mailing list <gnomemeeting-list gnome org>
- Subject: [GnomeMeeting-list] Transferring calls to asterisk
- Date: Mon, 12 Jun 2006 22:32:43 -0400
I have ekiga registering to a voip provider and receiving external call
through the stun server.
I want to redirect inconditionally all these calls to my asterisk
server, but I can't understand how and what should I configure in
asterisk in order to accept the redirected call.
In asterisk console I can't see nothing when ekiga passes the call.
If I turn asterisk's sip debug, I can see that the call arrives to
asterisk from 0108392222 voip eutelia it (skypho provider) via something
containing my external IP address, and asterisk tries to communicate
with a host on my external IP address, obviously unsuccessfully, and in
ekiga I get a occupied tone.
Anyone could help me? Thank you!
--
Buon Cammino!
don Paolo Benvenuto
Vuoi sapere di pi�quello che succede qui?
leggi il mio diario a http://www.chiesamissionaria.it/diario
Visita l'enciclopedia libera, dove puoi contribuire anche tu:
http://it.wikipedia.org/
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