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Asterisk TAPI

Created by: JustRumours,Last modification on Thu 20 of Sep, 2007 [16:51 UTC] by hehol

The Asterisk TAPI project

Open source project - Asterisk TAPI driver for Win32. Adds functionality to any TAPI compliant application for click to dial and screen popping on inbound calls.
Outlook can perform click to dial natively but has no inbuilt support for screen popping - this can be added with a utility such as IdentaFone's IdentaPop Pro or the Resource Software International Ltd. (RSI) Visual Rapport desktop productivity software.

The open source project is hosted on SourceForge,please see the project page to make a donation to help this project.

The project was brought to you from Omniis

Asterisk 1.2 Support

AstTapi build 0.10 was broken for Asterisk 1.2 by changes to the Asterisk Management Interface. I've made several code changes to get it working with Asterisk 1.2 and released a debug build for testing. The AstTapi for Asterisk 1.2 can be found here. Feedback is appreciated.


Looking for problems using AstTapi

We have tested AstTapi on a number of systems and appear to have little or no bugs - I m sure there are some there though!
If you are having problems using AstTapi then the following maybe useful.

Inbound Calls

  • Ensure your TAPI driver is logging onto your Asterisk server. Because it is a TSP it is generally very quiet about problems like this (maybe there is a way round it). Ssh into your Asterisk box and "asterisk -gcvvvr" now run your TSP driver by loading whatever TAPI software you are using you should see your manager logon - or an error message. - If you do not see it log on then you may not be running your asterisk manager - if this is the case edit your manager.conf and reload.
  • If it logs on successfully then it could be you do not have your channel configured correctly. Ensure the channel in the AstTapi matches your channel defined in Asterisk � for example I use Sip/nick in both the sip.conf and in my TAPI driver. It is not case sensitive and can even be a little short!
  • The "Inbound channel" is usually the same as the "User channel", if you use the same phone for in- and outgoing calls (example: SIP/555 , where 555 is the extension of the phone). This is specifically of interest for people using Identafone's IdentaPop Pro
  • If you installed AstTapi on a PC behind a router, while your Asterisk runs off a server at a provider's site for example, make sure to set up your router to forward port 5038 (or whichever port you put in manager.conf, default is 5038) to the internal IP address of the computer you installed AstTapi on. Else the signalling of calls to your PC will never arrive.. This too is specifically of interest for people using Identafone's IdentaPop Pro
  • If this does not fix it then there is a bug � please send me the details.

Outbound calls

  • The biggest problems I have suffered with on outbound calls are because of the dialling properties in Windows. This is the process by which a TAPI enabled app decides whether to put a 9 in front of the number or not (or whatever the case maybe). Make sure this is setup correctly.The second part to this is, windows will not following a dialling rule if you do not have a defined area code for a contact � typically Outlook users have all of the number stored in the number field � this is incorrect, the area code needs separating out for correct dialling outbound.
  • Ensure your dialling properties are setup correctly. Either setup the dialling channel correctly � or preferable the dialling context, this needs to be the same context as your phones which can place outbound calls. Again this type of this can be debugged by logging into the asterisk interface.

Programm specific settings

cobra Adress Plus

This CRM system supports various Telephony APIs. Using the Microsoft TAPI with the AstTAPI TSP is working. Make sure the following settings are set
  • OWNER is unchecked
  • Monitor is checked
  • Phone does not send "hangup" is checked
  • Always show is selected

Everything else is kept as default.

See also


Broken links



Go back to Asterisk GUI


Comments

Comments Filter
222

333AstTapi: Auto-Answer of the originating call leg?

by srl100, Thursday 11 of January, 2007 [11:22:48 UTC]
Does anyone know of a way to make AstTapi include the relevant Alert-Info SIP headers to enable the originating phone to auto-answer?
<p>
I've tried setting up a custom context (see below), but the dial plan is only entered AFTER the originating call is answered, so the SIP header is added to the terminating call leg, not the originating call leg.
<p>
[click-to-call-custom]
exten => _X.,1,NoOp("Click to Call")
exten => _X.,2,SetVar(_ALERT_INFO=info=alert-autoanswer)
exten => _X.,3,Goto(from-internal,${EXTEN},1)

222

333TAPI for Cisco

by tombar11, Tuesday 04 of April, 2006 [09:18:26 UTC]
For Cisco CM 4.X - Any freeware that integrate the TAPI CID popup screen based on searching microsoft contact list BUT with the option to answer or reject the call from the popup screen?
222

333TAPI for Cisco

by tombar11, Tuesday 04 of April, 2006 [07:28:34 UTC]
For Cisco CM 4.X - Any freeware that integrate the TAPI CID popup screen based on searching microsoft contact list BUT with the option to answer or reject the call from the popup screen?
222

333Re: Trouble using Identapop thru Nortel BCM 400

by jcoomans, Tuesday 20 of December, 2005 [23:45:36 UTC]
How do you have your Asterisk connected to the BCM? The BCM will not pass Caller ID info through on analog ports (i.e. ASM ports) unless the ASM module is relatively new (ASM8+ model), and the software on the BCM is relatively recent (i.e. 3.6, I believe, or higher).
222

333Trouble using Identapop thru Nortel BCM 400

by Trip, Tuesday 08 of November, 2005 [19:50:20 UTC]
I am using Asterisk on the other side of a Nortel BCM 400 and the BCM does not seem to be passing the caller info thru to the Tapi file for identapop to display???

Any help would be appreciated..

TIA
222

333TSP crashes after first dial

by j.koopmann, Wednesday 12 of October, 2005 [09:51:57 UTC]
Hi,

just installed the latest version. The first call can be established but the moment the other side picks up, the calling application (Outlook or whatever) crashed and afterwards the TSP seems dead. Only thing that helps is rebooting the computer. Any ideas? Who can be contacted to debug this?
222

333Re: Asterisk stops when dialing out with TAPI

by AnDi, Friday 26 of August, 2005 [14:40:01 UTC]
(:redface:) Sorry guys - after i started to read what is written in the dialog box i used "Dial by Context" instead of "Dial out by using the 'Dial' application" - and guess, everything works!

After all, it is indead an interesting issue that asterisk is exiting. Maybe someone smarter then me should take a look on that - Maybe some problem with my Austrian system, as the last message of asterisk server task wes "Speicherzugriffsfehler" in german language?
 == Parsing '/etc/asterisk/manager.conf': Found
 == Manager 'akammerer' logged on from 10.0.1.60
      > Channel SIP/10.0.1.53-a1f1 was answered.
      > Lauching Dial(CAPI/contr1/4282121) on SIP/10.0.1.53-a1f1
   — creating pipe for PLCI=0
Speicherzugriffsfehler

222

333Asterisk stops when dialing out with TAPI

by AnDi, Friday 26 of August, 2005 [14:27:13 UTC]
When i try to dial out from Outlook2003 / Windows 2003 Server my Phone rings, i lift the earphone and my asterisk says goodby:

 == Parsing '/etc/asterisk/manager.conf': Found
 == Manager 'akammerer' logged on from 10.0.1.60
      > Channel SIP/10.0.1.53-a1f1 was answered.
      > Lauching Dial(CAPI/contr1/4282121) on SIP/10.0.1.53-a1f1
   — creating pipe for PLCI=0
asterisk*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
Asterisk cleanly ending (0).

(:frown:) Anyone any idea?
222

333can only call myself ? ?

by joashherbrink, Friday 10 of June, 2005 [17:35:49 UTC]
(:confused:) i have the tapi driver installed
i see the manager login in (asterisk console)

but, whatever i do, when useing the tapi from outlook, it always returns the call to my own extension.

anybody?

thing is, i am only trying to make a call to another SIP phone on the same asterisk server.

so, what would be needed in outgoing chan value?

joash
222

333tapi rocks

by dean.collins, Tuesday 17 of May, 2005 [14:18:38 UTC]
this software rocks, you guys should set up a paypal donation account.