[asterisk-users] Asterisk Jingle<->SIP GW Question

[Ali Jawad]

 Dear All

I am using gtalk features with my own XMPP server "OpenFire"
I have setup gtalk.conf and jabber.conf on asterisk and now I can make calls
from clients registered on my XMPP server to SIP devices by calling the xmpp
accounts registered as clients on asterisk.

So far so good. So if I want to call sip:1000 I call the xmpp account that
is bound to that account in extensions.conf. However what do I have to do to
make this work with PSTN numbers. I can just setup an entry + extensions for
each pstn number I want to call.

I know that I can parse the incoming number and send it to the PSTN with
sip, however with jingle the number must be online already since jingle is
presence based. So I must have a registered client for each number I want to
call in the following format

XMPP <-----------> SIP
1000     To Call   1000 //sip extension
1001     To Call   15461315461 //pstn num
1002     To Call   46456543213 //cell phone num

So in essence I need to have one entry in jabber.conf per number, is there
something dynamic that can be done ?

Thanks
<asterisk-users at lists.digium.com>
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