Asterisk as a bridge between voip users and pots conference


#1

Hello,

I am thinking about this,

----POTS–CONFERENCE-BRIDGE
|
|
|
PSTN
|
|
ASTERISK
|
INTERNET
|
|
VOIP USERS

Users registers with asterisk, they join the confrence and later (or maybe
at the begining) asterisk automatically (or maybe manually) calls the POTS
conference bridge using the PSTN network.
This would allow all VOIP users to interact with users on the pstn
conference side. Any ideas how this could be done if possible.

I was able to create simple solution
VOIP users call exten 500 which is
[meetme]
exten => 500,1,Playback,thereare
exten => 500,2,MeetmeCount,500
exten => 500,3,Playback,callersin
exten => 500,4,Meetme,500|pMs|1234
exten => 500,5,Playback,goodbye
exten => 500,6,Hangup

later somebody calls extension 501 which moves 1-test
to /var/spool/asterisk/outgoing/

1-test looks like this
Channel: Sip/number@context (you put whatever you want)
Callerid: 1
MaxRetries: 1
RetryTime: 60
WaitTime: 30
Context: common
Extension: 500
Priority: 1

exten => 501,1,System(/bin/cp /etc/asterisk/1-test
/var/spool/asterisk/outgoing/
)
exten => 501,2,Hangup

The problem with this solution is that the 501 needs to be dialed
separately. Any ideas how to enable 501 in conference call.

Thx


#2

By seperate you mean that some one has to dial 501 when there are people in the confrence ? when do you want the call to be placed ? at a certain time ? after a certain amount of users in the room ?


#3

Allow somehow one of the callers , ability to dial the 501 extension during the conference if possible.