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

[Bartosz Wegrzyn - asterisk]

anyone whos the answer?

thx

> 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
>>
>>
>>
>>
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>
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