Difference between revisions of "Conference calls"

From TD-er's Wiki
Jump to navigationJump to search
 
Line 1: Line 1:
 
To configure this feature in Asterisk, edit the meetme.conf file to map one or more extension numbers to conference rooms (as shown in the example below, which creates three conference rooms numbered 9000, 9001 and 9002).  
 
To configure this feature in Asterisk, edit the meetme.conf file to map one or more extension numbers to conference rooms (as shown in the example below, which creates three conference rooms numbered 9000, 9001 and 9002).  
 +
 +
<tt>meetme.conf</tt>
 
  ;
 
  ;
 
  ; Configuration file for MeetMe simple conference rooms
 
  ; Configuration file for MeetMe simple conference rooms

Revision as of 23:51, 31 January 2007

To configure this feature in Asterisk, edit the meetme.conf file to map one or more extension numbers to conference rooms (as shown in the example below, which creates three conference rooms numbered 9000, 9001 and 9002).

meetme.conf

;
; Configuration file for MeetMe simple conference rooms
; for Asterisk of course.
;
[rooms]
;
; Usage is conf => confno,pincode,adminpin
;
conf => 9000
conf => 9002,123456
conf => 9003,123456,654321

The conference bridge can be accessed in one of several ways depending on the type of telephone system you have, whether it supports voice over IP, and how easily it can be expanded:

  • Internal T1/PRI - here you connect an internal T1 or ISDN Primary Rate circuit from the PBX directly to the conference bridge. This is a good option if your PBX is expandable, but does not support VoIP. This way you don’t have to order a local loop T1 from the local phone company, which will cost you several hundred dollars per month.
  • Connect direct to public telephone network - order one or more local loop T1/PRI circuits from your phone company, connect them directly to the conference bridge. This is a good option if your PBX is not expandable. The downside is you have to pay for more local telephone service.
  • Connect to PBX via Voice over IP. If your PBX or local telephone provider supports SIP, H.323 or IAX voice over IP service, you can route calls to Asterisk via your LAN/WAN. This is the best option if it’s available as you can eliminate the need for T1 interface cards in the Asterisk box, as well as expansion cards for your PBX.



N.B. This is a part of the text in this document.

Links