asterisk users - Mar 2005 - Dial command problem(VOIP+*+TDM400P+Legacy PBX)

Hello,

I just setup the Asterisk to integrate with Panasonic legacy PBX. Config as
followings,

PSTN <-- PanasonicPBX--TDM400P(FXO)--AsteriskPC --> Internet

* is for AA / Voicemail and VOIP in/out

Currently the AA / Voicemail function for incoming PSTN calls are working well. 
My problem is for the incoming VOIP call. It can ring my internal extensions and
talk without problem.
But it's all :-( I can not monitor the calling progress and handle it with *
because of something wrong with
the "Dial" command. For example, ring forever if nobody answer the
call, the call just be disconnected if
called extension busy, etc.

The code is as followings. Incoming VOIP call get into [via-net], then go to
[test] to dial the extension. 
>From the console, I can see the "Dial" command is excuted and
always stay there if nobody answer the call.
1.Why no dial timeout? I doubt if it's because the TDM400P FXO is connected
to the extension port of
Panasonic PBX, so that it's recognized as "answered" just while
DTMF tone is sent. Is it true?

2.MusicOn Hold can be heard but stop right away while extension is ringing.
Seems "answered" even still
nobody pickup the phone.

3.My Panasonic can sent the DTMF tone to indicate the status(called extension is
ringback, busy, etc)
I have used this good function to finish AA for incoming PSTN call. But here I
can do nothing since it stay
on the "Dial" application and not continue (if in
"Background", I can detect the tone).
I even use the option M(macro) try to catch the tone send from Panasonic, but
failed also. Anybody can
give me your comment? Thanks!

BR, Dominic

----------------------------------------
[via-net] 
exten => _1XX,1,Answer 
exten => _1XX,2,SetVar(called_ext=${EXTEN}) 
exten => _1XX,3,Goto(test,s,1)

[test] 
exten => s,1,Background(transfer) 
exten => s,2,Dial(Zap/g1/${called_ext}|10|m) 
exten => s,3,NoOp 
------------------------------------------
Message on console while FOREVER ringing, 
-- Executing Dial("IAX2/dominic@dominic/1",
"Zap/g1/102|10|m") in new stack
-- Called g1/102 
-- Started music on hold, class 'default', on IAX2/dominic@dominic/1 
-- Zap/1-1 answered IAX2/dominic@dominic/1 
-- Stopped music on hold on IAX2/dominic@dominic/1

 
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20050325/a1dafcbd/attachment.htm