On 14 Sep 2007, at 13:45, William Stillwell (Ki4swy) wrote:
> I am trying to determine what would need to be done/modified to
> enable the following:
>
> I have a SIP extension come into my asterisk box, and I then need
> it to call "6-10" remote Sip Stations that are set to Auto-Answer...
>
> (note, my remote sip stations are actually cisco h323 devices, I
> can call them fine from any softphone, or other device, and have
> full-duplex audio, however, i need to be able to conference bring
> all the remote stations automatically.w/Full duplex audio.
>
> Or if someone could direct me to a list that would actually be able
> to answer this question..
Oddly enough I've just done a quick hack like that.
Basically, the dialplan for the incoming call execs System("/usr/
local/bin/mix")
then drops it into a meetme.
Mix is a shell script that puts call files into the asterisk spool
directory.
The spool files dial the autoanswer sip devices and drop them into
the same meetme
Drop me a mail offlist if you need help.
Tim.
_______________________________________________
Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
