Asterisk Dial & Announce Tool
TTTelecom has developed a CTI-integration tool for the Open Source Asterisk PBX. ADAT ("Asterisk Dial & Announce Tool) enables users to monitor and control their extension. CRM-integration is also part of the software.
- Call notification
- Incoming and outgoing calls are announced using a tray icon balloon.
- On incoming calls the calling party name and number is announced (as log as Asterisk provides this information)
- CRM Customer details can be shown when a call is announced by clicking on the balloon
- The CRM-integration url's can be defined seperately for inbound and outbound calls
- Manual PC Dialer
- Using the ADAT dialer form, any number can be manually entered (or pasted) and dialed.
- ADAT will setup the calll for you by dialing your extension first. When you answer the outbound call will be setup.
- Copy & Paste Dialer
- Faster dialing with a simple copy & paste allows you to use any source of information (Windows program that allows select and copy)
- Just paste the number into ADAT and dial.
- Click-To-Dial using callto: links
- It is possible to let ADAT handle CallTo URL's, resulting in fast click-to-dial functionality.
- ADAT will automatically setup a call between your extension and the CallTo URL number you click on.
- The use if CallTo URL's on the public web is growing, but can easily be implemented in your own personal Intranet and/or CRM enviroment
- Internet Explorer and Firefox context-menu dialer
- ADAT includes an Internet Explorer plugin that can be used for context-menu dialing on any web page.
- Just select the number, right-click and choose "Dial using ADAT" to initiate a dial out.
- The Firefox plugin is available as a separate .XPI download.
- Call history
- Using the call history you can lookup placed and received calls.
- You can also easily (re)dial any number from your history.
- ADAT can be configured (available through .ini setting at the moment) to send a special SIP header which instructs SIP phones to auto-answer the call request.
- Your phone answers the ADAT call-request immediately (putting it on speakerphone) and then dials the requested desitination without touching the phone!
- The auto-answer feature has been tested with Grandstream phones. Other SIP phones may work also.
We appreciate you feedback and look forward to any feature request. Please visit the ADAT Forum. For ADAT R2 you can go to the ADAT Wiki for general (configuration) guidance.
Asterisk (ADAT R1) Server Configuration
In order to use ADAT, you will need to (let your System Administrator) add an account to the Asterisk Manager API. User accounts are configured in /etc/asterisk/manager.conf. A user account consists of a set of permitted IP hosts, an authentication secret (password), and a list of granted permissions. The account will need reade/write permisson for the "call" priviledge. For example:
secret = adat
read = call
write = call
Windows Client Configuration
Connection - settings
In the ADAT "Connection" settings enter Asterisk Manager credentials with the appropriate permissions (see Asterisk Server Configuration). You also need to specify the extension you want to manage & monitor.
Server = 192.168.1.10:5038
Password = adat
Extension = SIP/200
Context = from-internal
Events - settings
Set Events to "Dial" for now and enter the same Extension. For example:
Events = Dial
Extension = SIP/200
Set AutoPickupOnDial=True to enable the auto-answer SIP header to be sent. Unfortunately, due to the Asterisk Manager API design, the auto-answer is also forwarded to the SIP destination.