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Update README.adoc

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A Holt 2021-08-18 10:02:08 -04:00 committed by GitHub
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roles/pbx/README.adoc
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@ -138,7 +138,7 @@ image::files/linphone_connected.jpg[width='33%']
. Try the script link:templates/iiab-asterisk-test[*iiab-asterisk-test*] to:
* Create two extensions *301* and *302* and configures a dialplan for routing calls (as specified in link:templates/pjsip_custom.conf[pjsip_custom.conf] and link:templates/extensions_custom.conf[extensions_custom.conf], located within `/opt/iiab/iiab/roles/pbx/templates`).
* Create two extensions *301* and *302*, and configure a dialplan for routing calls (as specified in link:templates/pjsip_custom.conf[pjsip_custom.conf] and link:templates/extensions_custom.conf[extensions_custom.conf], located within `/opt/iiab/iiab/roles/pbx/templates`).
* Make a test call to extension 1000 (that has no physical device associated with it) that plays some sound files.
* After the script completes, 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.
@ -164,7 +164,7 @@ Restarts asterisk, no other actions are performed
* `sudo ./iiab-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.
Makes a test call from Asterisk's 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_