Arkansas Website Design, Fort Smith - Asttapi for Asterisk 1.2 - Comments

Hi Thanks for the info. I

Anonymous — Mon, 08 Sep 2008 06:35:59 -0500

Thanks for the info.
I have installed the patch and asttapi OK.

But I cannot work out what to put in the extensions.conf to notify ASTTAPI. It doesnt make much sense.

My standard context for incoming calls is:

[incoming]
exten=>039012xxxx,1,Dial(SIP/clientphone,30,r)

How do I adapt the required context into the above string?

Got it working

Cygnus — Wed, 23 Jul 2008 12:08:13 -0500

Hi!

This works now for me. Thanks for providing information on Asterisk 1.2 and the corresponding patched file.

One should note that if you use a different line name and user channel the above given extensions do not work.

The dial line for incoming calls should then read:

exten => s,n,Dial(${ARG1},20,rtM(tapi^${UNIQUEID}|${ARG2}))

when the macro is called in the following way:Macro(stdexten,SIP/sipuserchannel,lineid)

Greetings Torge

Feedback

Cygnus — Wed, 23 Jul 2008 11:17:34 -0500

Hi!

I have followed your configuration using a Outlook 2007 client with Identapro on Windows XP. I can initiate outgoing calls, but incoming calls are not shown in the identapro debug log.

I have verified that asterisk is sending out the incoming phone event through the manager interface on port 5038. But on the astapi side i do not see any incoming call. Any ideas? I can send you a tcpdump on port 5038 of the incoming call.

Greetings Torge

Asttapi for Asterisk 1.2

Tue, 16 May 2006 14:07:28 -0500

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 => s,n,Dial(${ARG2},20,rtM(tapi^${UNIQUEID}|${ARG2}))

exten => s,n,Goto(s-${DIALSTATUS},1)                ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

;exten => s,104,Goto(s-BUSY,1)
exten => s-NOANSWER,1,Voicemail(u${ARG1})       ; If unavailable, send to voicemail w/ unavail announce

exten => s-NOANSWER,2,Goto(default,s,1)         ; If they press #, return to start
exten => s-BUSY,1,Voicemail(b${ARG1})           ; If busy, send to voicemail w/ busy announce

exten => s-BUSY,2,Goto(default,s,1)             ; If they press #, return to start
exten => _s-.,1,Goto(s-NOANSWER,1)              ; Treat anything else as no answer
exten => a,1,VoicemailMain(${ARG1})             ; If they press *, send the user into VoicemailMain

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: