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Add details about asterisk_test to readme

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@ -106,7 +106,7 @@ image::files/add_extn_01_top.jpg[]
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image::files/add_extn_02.jpg[]
. Register the extension, on your smartphone or laptop
. [[Register_Extension]]Register the extension, on your smartphone or laptop
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You can now register the extension using a softphone (SIP) app on your smartphone or laptop. In this example we will use the https://en.wikipedia.org/wiki/Linphone[Linphone] app, on an Android phone. After you open the app, follow these steps:
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** Please create a https://en.wikipedia.org/wiki/Privacy_policy[Privacy Policy] against abusive surveillance, and explain it to the people in your community. Strongly consider giving them access to their own statistics with the <<#UCP,User Control Panel>> summarized further below.
=== Try using our script to create extensions and make a quick console call
. The script *asterisk_test*
* creates two extensions *301* and *302* and configures a dialplan for routing calls (as specified in pjsip_custom.conf and extensions_custom.conf located at: `roles/pbx/templates`)
* makes a test call to extension 1000 (that has no physical device associated with it) that plays some sound files.
* after the script is complete, it deletes the extensions and reverts file changes to restore asterisk to its original state. If you'd like your changes to persist, read the options below.
. Usage
* `sudo ./asterisk_test`
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Default option - Executes the complete script and reverts file changes done during the test.
* `sudo ./asterisk_test retain`
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Extensions created during the test are retained. Files are not reverted after the test.
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You may try making calls to extension 302 or 1000 after registering your softphone as <<#Register_Extension,mentioned above>>. The password for the extension is present in pjsip_custom.conf, feel free to change it before executing the script.
* `sudo ./asterisk_test revert`
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Use this option if you executed the script using the *retain* option, but would like to have the changes reverted now. This will only revert the changes to the files and restart asterisk, no other actions are performed.
* `sudo ./asterisk_test restart`
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Restarts asterisk, no other actions are performed
* `sudo ./asterisk_test testcall`
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Makes a test call from asterisk console to extension 1000 and confirms from the logs if the call was successful.
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_Note: This will only work if you've created extension 1000 manually, or using FreePBX or using the *retain* option of this script_
You can read more about creating _extensions_ and _dialplans_ https://wiki.asterisk.org/wiki/display/AST/Creating+SIP+Accounts[here]
=== Troubleshooting
. Check if Asterisk is up and running: