asterisk users - May 2005 - Uncommon callback

Hello!

I got an interesting task to make with asterisk:

pstn--- * ---sip--- * pstn
This sounds common till now. What I have to make is:

1.the call is routed through PSTN to asterisk1 (#1) which has ISDN PRI
interface(s) - leg1
2.#1 doesn't pick up the call, neither rejects, it just place into state
CALL PROGRESS (?) [maybe gives back alerting tone? probably not]
3.#1 notifies that there is a call to number B from number A to system
asterisk2 (#2) - through non-voip protocol [xml-rpc, anything else,
doesn't matter here]
4.#2 dials number B through PSTN - leg2
5.#2 dials #1 (on number #1 sent in notification) through SIP - leg3
6.#2 interconnects leg2 and leg3 without ISDN CONNECT
7.#1 interconnects leg1 and leg2 without ISDN CONNECT
8.when B picks up the phone, channels are getting CONNECT message

I know that my explanation can be a bit mess, but I wasn't able to write
it down better.
I was thinking about putting the incoming call on #1 into meetme, but
meetme looks to establish the call, so it is charged from the 1st second
- that's not good in this case. In #2 I thought, .call feature can
initiate the call out to number B (leg2) which will dial to #1 using SIP.

I have many black holes right now.

Any help appreciated!

Kind regards,
	Tamas

Your explanation is really messy but from what I understand it seem
like you just want to be able to take incoming call from one asterisk server
and forward it to another asterisk sever over IP that would terminate the call
but dont connect calls untill B party answers the phone on second
asterisk server.
If thats the case its very simple to do with just standard iax2 and
proper extensions

regards
m.