I've patched Asttapi 0.10 to work with Asterisk 1.2. This requires things to be properly setup in your dial plan per the Asttapi 0.10 documentation, which admittedly is pretty poor. I'll try to simplify that a bit here.

Asterisk signals Asttapi through the UserEvent command in the dialplan. Asttapi needs to be informed of call progress as originated and incoming calls are progressing. Partially a problem exists that as the calls are connected, two unique IDs are generated which means that some of the commands need to pass the old unique ID back to asttapi as well. To setup Asttapi properly in the dial plan, you need the following:

Prior to the Dial() command, you need to send some UserEvents to Asttapi to let it know there's a call coming into your phone. I've modified my stdexten macro to look like this:


exten => s,1,UserEvent(TAPI|TAPIEVENT: LINE_NEWCALL ${ARG2})
exten => s,n,UserEvent(TAPI|TAPIEVENT: LINE_CALLSTATE LINECALLSTATE_OFFERING)
exten => s,n,UserEvent(TAPI|TAPIEVENT: SET CALLERID ${CALLERID})
exten => s,n,UserEvent(TAPI|TAPIEVENT: LINE_CALLINFO LINECALLINFOSTATE_CALLERID)

exten => h,1,UserEvent(TAPI|TAPIEVENT: LINE_CALLSTATE LINECALLSTATE_IDLE)


This sends both a dial event and a hangup event to asttapi to let it know an incoming call is progressing. Notice in the Dial command for stdexten, we also call a macro, tapi, that sends additional call progression information after the call is connected. The macro looks like this:


[macro-tapi];
exten => s,1,UserEvent(TAPI|TAPIEVENT: [~${ARG1}&${ARG2}] LINE_CALLSTATE LINECALLSTATE_CONNECTED)
exten => s,2,UserEvent(TAPI|TAPIEVENT: [~${ARG1}&!${ARG2}] LINE_CALLSTATE LINECALLSTATE_HANGUP)


For outbound calls, we need a macro that also properly signals asttapi. I've modified my trunklocal and trunkld macros to call another macro, which was originally designed to fall back to VoIP trunks if all the analog trunks were busy, but I'll show you the excerpt that signals asttapi that should be relevant. Replacing the Dial() command in trunklocal and trunkld with this macro should be sufficient:


[macro-fallback]
; ${ARG1} : Trunk to call out on
; ${ARG2} : Number to call
; ${ARG3} : Time to let ring
; ${ARG4} : Dial Options

exten => s,1,UserEvent(TAPI|TAPIEVENT: LINE_NEWCALL ${CHANNEL:0:7})
exten => s,2,UserEvent(TAPI|TAPIEVENT: LINE_CALLSTATE LINECALLSTATE_DIALTONE)
exten => s,3,UserEvent(TAPI|TAPIEVENT: LINE_CALLSTATE LINECALLSTATE_DIALING)
exten => s,4,UserEvent(TAPI|TAPIEVENT: LINE_CALLSTATE LINECALLSTATE_PROCEEDING)
exten => s,5,Dial(${ARG1}/${ARG2},${ARG3},${ARG4}M(tapi^${UNIQUEID}|${CHANNEL:0:7}))


Note in the Dial() command, the macro called sends ${CHANNEL:0:7} as the second parameter. This is to signal to asttapi that we're on the right line. My asttapi line is set as SIP/101, but asterisk ${CHANNEL} holds something like SIP/101-asdf, so I limit the output to 7 characters. Make sure to adjust your settings inside the macro and inside asttapi appropriately. The line inside asttapi and inside the macro is arbitrary, and confusingly enough inside the original asttapi documentation, he uses hard coded values here, which would only work in testing and isn't scalable.

Also, on originated calls, whatever context they originate in or the default context, you need a hangup extension like I setup in the stdexten macro above.

To use the latest asttapi build, you'll need to install asttapi 0.10 from here. Then download the asttapi.tsp file from here and place it in your system32 directory and reboot (optionally, you can stop outlook and run a net stop tapisrv from cmd and probably be fine, but don't send me an email until you've rebooted :) ).

Let me know how it works for you. Sorry the documentation is poor, I'm looking for help there, I don't have time to write it well.

Update: Source is available here