[Asterisk-Users] Asterisk as a bridge between voip clients and POTS confrence bridge

[Bartosz Wegrzyn - asterisk]

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 at 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



> 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.
>
> Thanks
>
>
>
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>