Problem : How to configure 2 or more user to call a single conference extension
We have Asterisk 16.9.0 installed in a CentOS 7 server at our office’s local network. The only configurations we have made on this server are those we are making mention here.
–>What we want to do:
We want to configure such that, our Mobile network callers, should call our DID number and it should lead them to a conference through a single extension, which carries our DID number as the number to be called for the conference. We have configured 2 kinds of users in confbridge.conf: Admin and guest user with PINs. We want that the mobile network callers should all call the DID but they should enter the PIN of the guest user which we will give them and the admin will enter the PIN of the admin user account in confbridge.conf we will give them.
–>What we have been able to do:
We have been able to configure our system such that, our mobile network callers, call the DID and they are asked to put the PIN of the conference. But according to what we have been able to do, only the admin PIN functions, because it is it that is in our Dial plan.
Our configurations are below:
-----------User accounts------------ [wv_admin_user_DID] type=user pin=6666 marked=yes admin=yes music_on_hold_when_empty=yes announce_user_count=yes announce_only_user=yes announce_join_leave=yes [wv_guest_user_DID] type=user pin=4444 wait_marked=yes end_marked=yes music_on_hold_when_empty=yes announce_user_count=yes announce_user_count_all=yes announce_only_user=yes announce_join_leave=yes ---------- ConfBridge Bridge------------ [wv_bridge_DID] type=bridge max_members=50 video_mode = follow_talker We didn’t do anything the at level of MENU
[general] allowguest=no srvlookup=no udpbindaddr=0.0.0.0 tcpenable=no canreinvite = no dtmfmode=auto register => sipusername:email@example.com:5060/+237222518004 [sipusername] type=peer context=trunkinbound secret= passwd host= xxx.xx.xxx.xx nat=force_rport,comedia fromdomain= xxx.xx.xxx.xx fromuser=sipusername username=sipusername insecure=invite disallow=all allow=ilbc allow=g726 allow=gsm allow=g723 allow=ulaw dtmfmode=inband quality=yes  user=201 type=friend host=dynamic disallow=all allow=ulaw secret=201 context=trunkinbound nat=comedia
exten => ourusername,1,Dial(SIP/ourusername) exten => 201,1,Dial(SIP/201,20) exten => 201,2,Answer() exten => 201,3,Playback() exten => 201,4,Hangup() exten => +237222518004,1,Progress() exten => +237222518004,2,Wait(1) exten => +237222518004,3,ConfBridge(1,wv_bridge_DID,wv_admin_user_DID)
This above extensions.conf what we put which only considers the admin PIN. We agree it is normal because only it, is mentioned in the dial plan.
In attempt to advance to our need, we changed our extensions.conf as follow:
[trunkinbound] exten => +237222518004,1,NoOP() same => n,Answer() same => n,Set(CONFBRIDGE(wv_guest_user_DID,wv_admin_user_DID)=yes) same => n,ConfBridge(1,wv_bridge_DID) same => n,Hangup() exten => ourusername,1,Dial(SIP/ourusername) exten => 201,1,Dial(SIP/201,20) exten => 201,2,Answer() exten => 201,3,Playback() exten => 201,4,Hangup()
Thank you for your quick response. We appreciate.