[Asterisk-Users] Extensions to solve three way calling problem

[PHP Mechanic]

> On Wed, 2005-01-05 at 18:38 +1100, PHP Mechanic wrote:
>> >> Have you considered setting up a meetme confrence line for them? :)
>> >
>> >> > analog phone <=> asterisk/tdm11b <=> pstn
>>
>> I have played with it. But the problem I'm having is as follows
>>
>> exten => _1800.,1,Dial(Zap/4/${EXTEN},20,Tr)         ; call some company
>> willing to pay for my test, preferably get someone with an on hold 
>> message
>> ; Now I press #* on the analog phone to transfer them to Meetme
>> exten => *,1,Meetme,2000                                           ; send
>> them to meetme
>> exten => *,2,Flash() 
>> ;
>> flash the pstn line
>
> What makes you think that would flash the PSTN line?

Because the cli reports that it is executing flash on the Zap/4 - the PSTN 
line

>This is your
> problem. When you transfer the PSTN line anywhere and then go to dial
> again, the flash is actually on the current channel. I wouldn't be
> surprised if you hear it in your receiver. I don't know of anyway to
> flash the PSTN line from within asterisk that would do as you want. In
> fact, to enable it would be a security risk as well. Think of the
> possibility of having multiple lines in and then dialing an extension to
> flash the line and messing up and flashing someone else's connection.
>
> Closest thing I could think of is having your PSTN side caller do the
> transfer and redial. If the PSTN caller was allowed to transfer the
> inside person and then dial a special extension that would initiate the
> flash and the dial command. Of course the trouble here is that as soon
> as the flash occurs, the new caller is the one going to be stuck in an
> odd state and the previous PSTN caller is going to be in unrecoverable
> limbo.
>
> Just looks like you will be SOL on utilizing the PSTN 3 way calling.

Yeah, I think you are right.

But what is the point of  threewaycalling and transfer in zapata.conf - what 
do they do?