maybe some asterisk guru have idea for some smart script, how to do this
;-)
I found some RFC for better sip conferencing, but currently probably not
implemented in asterisk :'(
High-Level Requirements for Tightly Coupled SIP Conferencing
ftp://ftp.rfc-editor.org/in-notes/rfc4245.txt
nik600 wrote:
Ok thanks, do you think that it isn't possible to do that
automatically from asterisk?
On 12/12/06, Pavel Jezek <[EMAIL PROTECTED]> wrote:
I think, that adhoc conferencing isn't possible in this way, instead you
should use meetme, ie.:
skinny user calls to user A and transfer his to meetme number
skinny user calls to user B and transfer his to meetme number
skinny user calls to meetme number
all three speech in conference...
nik600 wrote:
> Hi, can i set up my asterisk for:
>
> - receive a skinny call in a specific context (yes, i have already
> compiled asteirsk with h323 support)
> - forward the call to a sip user A
> - make the sip user B join the call and create a conference between
> skinny caller, A and B
>
> maky thanks
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