[OpenSIPS-Users] [asterisk-users] Asterisk is not designed for University with largeuser base?

[Stefan Sayer]

o Yehavi Bourvine [03/19/09 07:08]:
> - Confference is limited to 3 participants. I guess I can do more with 
> external server but didn't
>   manage yet to make it working.
...
> - Same confference and "busy dial" problem.
for conference bridge you could give SEMS (http://iptel.org/sems) a try:
get http://ftp.iptel.org/pub/sems/sems-1.0.1.tar.gz; make ; make install
or on debian add deb http://ftp.iptel.org/pub/sems/debian etch free to 
sources.list and apt-get install sems.

in /etc/sems/sems.conf set
sip_ip=a.b.c.d
sip_port=xxyy
media_ip=a.b.c.d
application=conference
load_plugins=wav;gsm;ilbc;adpcm;speex;l16;sipctrl;session_timer;conference

start sems (sems -f /etc/sems/sems.conf [-D 3 -E]) or /etc/init.d/sems start
and send your call to a.b.c.d:xxyy, INVITE ruri user is the room name.

if you want to have the caller enter the room number in the beginning, set
application=webconference
load_plugins=wav;gsm;ilbc;adpcm;speex;l16;sipctrl;session_timer;webconference

BR
Stefan Sayer
-- 
Stefan Sayer
VoIP Services

stefan.sayer at iptego.com
www.iptego.com

IPTEGO GmbH
Wittenbergplatz 1
10789 Berlin
Germany

Amtsgericht Charlottenburg, HRB 101010
Geschaeftsfuehrer: Alexander Hoffmann