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Markdown Updates (#274)

* Update lets_encrypt.md

* Update provision.md

* Update provision.md

* Update languages.md

* Update extensions.md

* Update outbound_routes.md

* Update outbound_routes.md

* Update outbound_routes.md

* Update advanced_dialplans.md

* Update advanced_dialplans.md

* Update advanced_dialplans.md

* Update advanced_dialplans.md

* Update advanced_dialplans.md

* Update advanced_dialplans.md

* Update advanced_dialplans.md

* Update call_block.md

* Update call_block.md

* Update call_detail_record.md

* Update call_routing.md

* Update conference_profiles.md

* Update contacts.md

* Update contacts.md

* Update contacts.md

* Update contacts.md

* Update fax_server.md

* Update fax_server.md

* Update fax_server.md

* Update fax_server.md

* Update fax_server.md

* Update music_on_hold.md

* Update recordings.md

* Update streams.md

* Update voicemail_transcription.md

* Create voicemail.md

* Update voicemail.md

* Update follow_me.md

* Update voicemail.md

* Update voicemail.md
AdSecIT 4 months ago
parent
commit
22c4eae176

+ 3 - 3
source/accounts/extensions.md

@@ -485,14 +485,14 @@ Always back up your settings before making adjustments.
 > **Extension Caller ID input type select**
 >
 > If you want extension caller ID name and number to be input type text,
-> make sure permission outbound[caller_id_select]{#caller_id_select} is
+> make sure permission outbound_caller_id_select is
 > assigned to groups in Groups Manager. By default,
-> outbound[caller_id_select]{#caller_id_select} is not assigned to any
+> outbound_caller_id_select is not assigned to any
 > user groups.
 >
 > **Extension Caller ID input type select**
 >
 > If you want a select option for caller ID, you would assign
-> outbound[caller_id_select]{#caller_id_select} permission to groups of
+> outbound_caller_id_select permission to groups of
 > your choice using Group Manager and define Caller ID information in
 > Dialplan Destinations.

+ 1 - 1
source/applications/call_block.md

@@ -45,7 +45,7 @@ below the form.*
 Outbound call blocking requires an additional dialplan entry.
 
 If one doesn\'t exist already, add a new **call-direction** Outbound
-Route to set [call_direction=outbound]{.title-ref}\...
+Route to set `call_direction=outbound`.
 
 ![image](../_static/images/fusionpbx_call_block1.jpg)
 

+ 5 - 17
source/applications/call_detail_record.md

@@ -93,11 +93,7 @@ carefully.
 300\") \| crontab
 ```
 
-- Once you've made these changes you can save the file. You could restart
-your server, or you could reloadxml and then restart the xml[cdr]{#cdr}
-module. Either is ok, it is up to you. Then your changes will have taken
-effect and you should no longer lose your menu bar when looking at CDR
-information.
+- Once you've made these changes you can save the file. You could restart your server, or you could reloadxml and then restart the xml_cdr module.  Either is ok, it is up to you. Then your changes will have taken effect and you should no longer lose your menu bar when looking at CDR information.
 ```
 fs[cli]{#cli} -x \'reloadxml\'
 
@@ -106,21 +102,13 @@ fs[cli]{#cli} -x \'reload mod[xml_cdr]{#xml_cdr}\'
 
 ## XML CDR configuration
 
-For more detailed configuration go to the XML editor (Menu -\> Advanced
--\> XML Editor) and in autoload configs look at
-xml[cdr.conf.xml]{#cdr.conf.xml}
+For more detailed configuration go to the XML editor (Menu -> Advanced -> XML Editor) and
+in autoload configs look at xml_cdr.conf.xml
 
->**Note:** By default only the a-leg of the call is logged therefore if you make      
->a recording of the b-leg you won\'t be able to retrieve it using the Call    
->Detail Records. If you want the b-leg as well you need to change   
->log-b-leg=true in this config and in the default settings.
+>**Note:** By default only the a-leg of the call is logged therefore if you make a recording of the b-leg you won't be able to retrieve it using the Call Detail Records.  If you want the b-leg as well you need to change log-b-leg=true in this config and in the default settings.
 
 ## Harddrive space usage
 
->**Note:** Recordings also take up space and may be manually deleted   
->if you want the space back these are kept in (source install)   
->/usr/local/freeswitch/recordings/{Domian[Name]{#name}}/archive or   
->(package install) /var/lib/freeswitch/recordings/{Domain Name}/archive   
->and inside that by year, month and day.
+>**Note:** Recordings also take up space and may be manually deleted if you want the space back these are kept in (source install) /usr/local/freeswitch/recordings/{Domian_Name}/archive or (package install) /var/lib/freeswitch/recordings/{Domain Name}/archive and inside that by year, month and day.
 
 ## [CDR Default Settings](/en/latest/advanced/default_settings.html#id4)

+ 14 - 20
source/applications/call_routing.md

@@ -34,27 +34,21 @@ Add to Default Settings:
 ### Enable Feature Sync on the Device
 
 -   Yealink
-    -   Web Interface -\> Features -\> General Information -\> Feature
-        Key Synchronization set to Enabled
-    -   Config Files -\>
-        features.feature[key_sync.enable]{#key_sync.enable}
-    -   Might be addition settings needed for the latest firmware. I
-        tested with 81.0.110
-    -   FusionPBX Default Settings -\> Provision -\>
-        yealink[feature_key_sync]{#feature_key_sync}
+    -   Web Interface -> Features -> General Information -> Feature Key Synchronization set to Enabled
+    -   Config Files -> features.feature_key_sync.enable
+    -   Might be addition settings needed for the latest firmware. I tested with 81.0.110
+    -   FusionPBX Default Settings -> Provision -> yealink_feature_key_sync
+    
 -   Polycom
-    -   reg.{\$row.line[number]{#number}}.serverFeatureControl.cf=\"1\"
-    -   reg.{\$row.line[number]{#number}}.serverFeatureControl.dnd=\"1\"
-    -   FusionPBX Default Settings -\> Provision -\>
-        polycom[feature_key_sync]{#feature_key_sync}
+    -   reg.{$row.line_number}.serverFeatureControl.cf="1"
+    -   reg.{$row.line_number}.serverFeatureControl.dnd="1"
+    -   FusionPBX Default Settings -> Provision -> polycom_feature_key_sync
+
 -   Cisco SPA
-    -   \<[Feature_Key_Sync_1]()
-        group=\"Ext1/Call[Feature_Settings]{#feature_settings}\"\>Yes\</[Feature_Key_Sync_1]()\>
-    -   FusionPBX Default Settings -\> Provision -\>
-        spa[feature_key_sync]{#feature_key_sync}
+    -   <Feature_Key_Sync_1_ group="Ext_1/Call_Feature_Settings">Yes</Feature_Key_Sync_1_>
+    -   FusionPBX Default Settings -> Provision -> spa_feature_key_sync
+
 -   Grandstream GXP and GRP
-    -   Web Interface -\> Accounts -\> Account X -\> SIP Settings -\>
-        Advanced Features -\> Feature Key Synchronization
+    -   Web Interface -> Accounts -> Account X -> SIP Settings -> Advanced Features -> Feature Key Synchronization
     -   Config file P2325
-    -   FusionPBX Default Settings -\> Provision -\>
-        grandstream[feature_key_sync]{#feature_key_sync}
+    -   FusionPBX Default Settings -> Provision -> grandstream_feature_key_sync

+ 3 - 5
source/applications/conference_profiles.md

@@ -22,11 +22,9 @@ A group of conference parameters saved together as a profile.
 -   muted-sound: conference/conf-muted.wav is the default.
 -   unmuted-sound: conference/conf-unmuted.wav is the default.
 -   alone-sound: conference/conf-alone.wav is the default.
--   moh-sound: local[stream]{#stream}://default is the default.
--   enter-sound: tone[stream]{#stream}://%(200,0,500,600,700) is the
-    default.
--   exit-sound: tone[stream]{#stream}://%(500,0,300,200,100,50,25) is
-    the default.
+-   moh-sound: local_stream://default is the default.
+-   enter-sound: tone_stream://%(200,0,500,600,700) is the default.
+-   exit-sound: tone_stream://%(500,0,300,200,100,50,25) is the default.
 -   kicked-sound: conference/conf-kicked.wav is the default.
 -   locked-sound: conference/conf-locked.wav is the default.
 -   is-locked-sound: conference/conf-is-locked.wav is the default.

+ 3 - 1
source/applications/contacts.md

@@ -19,4 +19,6 @@ Contacts is a list of individuals and organizations.
 
 -   To generate a QR code click the **QR CODE** button at the top right
 
-![image](../_static/images/fusionpbx_qr_code.jpg){.align-center}
+<div style="text-align: center;">
+  <img src="../_static/images/fusionpbx_qr_code.jpg" alt="image">
+</div>

+ 13 - 36
source/applications/fax_server.md

@@ -93,15 +93,9 @@ Menu -\> Advanced -\> Default Settings then category Fax
 
 -   *fax_enable_t38_request=false* (Can be true or false)
 -   *ignore_early_media=true* (Can be true or false)
--   Some carriers it\'s better for
-    fax[enable_t38_request]{#enable_t38_request}=true and for some its
-    better for it to be false.
--   It\'s best not to make an assumption and to do testing with
-    different settings to get the best results for your particular
-    carrier.
--   The variable *fax_enable_t38_request=false* will send a T38 reinvite
-    when a fax tone is detected. In some cases the re-invite always
-    fails for some carriers which is why it is default to false.
+-   Some carriers it's better for fax_enable_t38_request=true and for some its better for it to be false.
+-   It's best not to make an assumption and to do testing with different settings to get the best results for your particular carrier.
+-   The variable *fax_enable_t38_request=false* will send a T38 reinvite when a fax tone is detected. In some cases the re-invite always fails for some carriers which is why it is default to false.
 
 ### Troubleshooting Tips
 
@@ -109,39 +103,22 @@ Faxing will fail at times. Fax Server should automatically try different
 methods for sending. There are different combinations like;
 
 -   With T-38 on/off
-
 -   ECC on/off
-
 -   Sending a wav file
-
--   Send a fax to HP faxback. This will test sending and receiving
-    1-888-473-2963
-
--   Test sending with Faxtoy.net This will display what is faxed on
-    their website. 1-855-330-1239 or 1-213-294-2943
-
-
-    Turn on verbose log in FreeSWITCH fax.conf.xml
-
-  -   From your FusionPBX installation go to ADVANCED \> XML
-      Editor and a new window will open.
-  -   Choose autoload[configs]{#configs} folder from the list,
-      then choose fax.conf.xml.
-  -   In fax.conf.xml there is an option that by default sets a
-      variable called verbose = false. If you change this to true
-      you get more logging details as the fax is actually
-      received, such as the quality of the connection etc.
-  -   You can see these details when you run the freeswitch
-      command line ie. **fs_cli**
-
+-   Send a fax to HP faxback.  This will test sending and receiving 1-888-473-2963
+-   Test sending with Faxtoy.net This will display what is faxed on their website. 1-855-330-1239 or 1-213-294-2943
+-   Turn on verbose log in FreeSWITCH fax.conf.xml
+    *   From your FusionPBX installation go to ADVANCED > XML Editor and a new window will open.
+    *   Choose autoload_configs folder from the list, then choose fax.conf.xml.
+    *   In fax.conf.xml there is an option that by default sets a variable called verbose = false. If you change this to true you get more logging details as the fax is actually received, such as the quality of the connection etc.
+    *   You can see these details when you run the freeswitch command line ie. **fs_cli** 
 
 ### Command Line Fax Statistics
 
-Grep from ssh or console access your freeswitch.log files for
-FAX[RETRY_STATS]{#retry_stats} to start keeping track of
-success/failure. Examples
+Grep from ssh or console access your freeswitch.log files for FAX_RETRY_STATS to start keeping track of success/failure.
+Examples
 
-Here\'s how you can get some totals.
+Here's how you can get some totals.
 
 **Total:**
 ```

+ 13 - 21
source/applications/follow_me.md

@@ -2,29 +2,21 @@
 
 Define alternate inbound call handling for the following extensions.
 
-![image](../_static/images/fusionpbx_follow_me.jpg)
+![Follow Me Settings](../_static/images/fusionpbx_follow_me.jpg)
 
--   **Call Forward-** (Disabled or Enabled) Input the destination number
--   **On Busy-** (Disabled or Enabled) If enabled, it overrides the
-    value of voicemail enabling in extension
--   **No Answer-** (Disabled or Enabled) If enabled, it overrides the
-    value of voicemail enabling in extension
--   **Not Registered-** (Disabled or Enabled) If endpoint is not
-    reachable, forward to this destination before going to voicemail
--   **Follow Me-** (Disabled or Enabled)
--   **Destinations-** Can set Delay, Timeout and Prompt to accept the
-    call.
--   **Ignore Busy-** (Disabled or Enabled)
--   **Do Not Disturbe-** (Disabled or Enabled)
+- **Call Forward**: (Disabled or Enabled) Input the destination number
+- **On Busy**: (Disabled or Enabled) If enabled, it overrides the value of voicemail enabling in extension
+- **No Answer**: (Disabled or Enabled) If enabled, it overrides the value of voicemail enabling in extension
+- **Not Registered**: (Disabled or Enabled) If endpoint is not reachable, forward to this destination before going to voicemail
+- **Follow Me**: (Disabled or Enabled)
+- **Destinations**: Can set Delay, Timeout, and Prompt to accept the call
+- **Ignore Busy**: (Disabled or Enabled)
+- **Do Not Disturb**: (Disabled or Enabled)
 
-This example has both the extension 1301 itself and and external number
-to call. If you don\'t put the extension itself the extension wont ring
-when in Follow Me. This is due to the flexible nature of FusionPBX where
-if you didn\'t want that extension to ring like if you were out of the
-office on a business trip.
+This example has both the extension 1301 itself and an external number to call. If you don’t include the extension itself, the extension won’t ring when in Follow Me. This is due to the flexible nature of FusionPBX—e.g., if you’re out of the office on a business trip and don’t want that extension to ring.
 
-![image](../_static/images/fusionpbx_follow_me1.jpg)
+![Follow Me Example](../_static/images/fusionpbx_follow_me1.jpg)
 
-## [Follow Me Default Settings](/en/latest/advanced/default_settings.html#id13)
+## Follow Me Default Settings
 
-Click the link above for Follow Me default settings.
+Click the link above for Follow Me default settings: [Follow Me Default Settings](/en/latest/advanced/default_settings.html#id13)

+ 6 - 20
source/applications/music_on_hold.md

@@ -1,9 +1,6 @@
 # Music on Hold
 
-Music on hold can be in WAV or MP3 format. To play an MP3 file you must
-have mod[shout]{#shout} enabled on the \'Modules\' tab. You can adjust
-the volume of the MP3 audio from the \'Settings\' tab. For best
-performance upload 16 bit, 8/16/32/48 kHz mono WAV files.
+Music on hold can be in WAV or MP3 format. To play an MP3 file you must have mod_shout enabled on the 'Modules' tab. You can adjust the volume of the MP3 audio from the 'Settings' tab. For best performance upload 16 bit, 8/16/32/48 kHz mono WAV files.
 
 ![image](../_static/images/fusionpbx_moh.jpg)
 
@@ -30,29 +27,19 @@ performance upload 16 bit, 8/16/32/48 kHz mono WAV files.
 
 ## Music on Hold Tips
 
--   When a new music on hold category mod[local_stream]{#local_stream}
-    will be restarted. If it is busy then it will not restart
-    automatically. A manual restart of the module is required when it is
-    not in use. The module can be restarted from the Menu -\> Advanced
-    -\> Modules or from the console and fs[cli]{#cli} with following
-    command.
+-   When a new music on hold category mod_local_stream will be restarted. If it is busy then it will not restart automatically. A manual restart of the module is required when it is not in use. The module can be restarted from the Menu -> Advanced -> Modules or from the console and fs_cli with following command.
 
 ```
     reload mod_local_stream
 ```
 
--   Each music on hold category is given a name. If the domain is set to
-    global the name will be the name in the example below the protocol
-    that is used is local[stream]{#stream} and the music on hold
-    category is default and domain is set to global.
+-   Each music on hold category is given a name. If the domain is set to global the name will be the name in the example below the protocol that is used is local_stream and the music on hold category is default and domain is set to global.
 
 ```
     local_stream://default
 ```
 
--   It is possible that a domain or tenant can have its own category of
-    music. In this example the name is \'custom\' and the domain was
-    assigned automatically to the current domain.
+-   It is possible that a domain or tenant can have its own category of music. In this example the name is 'custom' and the domain was assigned automatically to the current domain.
 
 ```
     local_stream://domain_name/custom
@@ -83,10 +70,9 @@ category. Otherwise, click the +button to create a new MOH category.
 
 - SSH into your server and run the following commands:
 
-- fs[cli]{#cli} This command opens the FreeSwitch CLI
+- fs_cli  This command opens the FreeSwitch CLI
 
-- reload mod[local_stream]{#local_stream} This command reloads the new
-category
+- reload mod_local_stream This command reloads the new category
 
 - Press Ctrl+D to exit the CLI.
 

+ 1 - 2
source/applications/provision.md

@@ -84,8 +84,7 @@ In order to use the phone book a few steps are needed.
 
 ![image](../_static/images/provision/fusionpbx_remote_phonebook1.png)
 
--   Set **Enabled True** for contact[extensions]{#extensions},
-    contact[users]{#users} and contact[groups]{#groups} in [Default
+-   Set **Enabled True** for contact_extensions, contact_users and contact_groups in [Default
     Settings](/en/latest/advanced/default_settings.html).
 
 ![image](../_static/images/provision/fusionpbx_phone_book2.png)

+ 6 - 11
source/applications/recordings.md

@@ -8,28 +8,23 @@ video.](https://youtu.be/CkqlsVvvv2U)
 <iframe width="100%" height="350" src="https://www.youtube.com/embed/CkqlsVvvv2U?rel=0" frameborder="0" ; encrypted-media" allowfullscreen></iframe>
 </div>
 
-To view and set the pin number goto Dialplan \> Dialplan Manager \>
-Click on Recordings \> pin[number]{#number}=8675309 at the bottom.
+To view and set the pin number goto Dialplan > Dialplan Manager > Click on Recordings > pin_number=8675309 at the bottom.
 
 :::: note
 ::: title
 Note
 :::
 
-Pin number is recomended but can be left empty if no pin number is
-desired then pin[number]{#number}=
+Pin number is recomended but can be left empty if no pin number is desired then pin_number=
 ::::
 
 ## Create a Recording
 
 1.  Dial \*732 and wait for the voice prompt
-2.  Enter the password (pin[number]{#number}) followed by the pound
-    sign# Enter at least a 3 digit number. This will label the recording
-    file. (recording100.wav)
-3.  start talking to make the recording after the voice prompt and press
-    the pound key \#
-4.  Press 1 to accept the recording then hang up or press 2 to start
-    over.
+2.  Enter the password (pin_number) followed by the pound sign# 
+   Enter at least a 3 digit number.  This will label the recording file. (recording100.wav)
+3.  start talking to make the recording after the voice prompt and press the pound key #
+4.  Press 1 to accept the recording then hang up or press 2 to start over.
 
 ![image](../_static/images/applications/recording/fusionpbx_call_recordings1.png)
 

+ 1 - 1
source/applications/streams.md

@@ -4,7 +4,7 @@ Define details for streaming audio.
 
 ![image](../_static/images/applications/streams/fusionpbx_streams1.png)
 
--   Make sure mod[shout]{#shout} is installed and is started.
+-   Make sure mod_shout is installed and is started.
 
 -   Have a shoutcast url ready to use. (shout://domain.tld/path/to/)
 

+ 93 - 0
source/applications/voicemail.md

@@ -0,0 +1,93 @@
+# Voicemail
+
+To edit voicemail settings, click the pencil edit icon on the right of the extension number.
+
+![Voicemail Settings](../_static/images/voicemail/fusionpbx_voicemail.jpg)
+
+Here you can edit voicemail settings:
+
+- **Play Tutorial**: Play the voicemail tutorial after the next voicemail login
+- **Greeting**: When you dial ***97**, record a greeting and set a number you can choose which greeting to use
+- **Alternate Greet ID**: An alternative greet ID used in the default greeting
+- **Options**: Define caller options for the voicemail greeting
+- **Mail to**: Have voicemails emailed to this address
+- **Voicemail File**: Select a listening option to include with the email notification
+- **Keep Local**: Choose whether to keep the voicemail in the system after sending the email notification
+- **Forward Destinations**: Forward voicemail messages to additional destinations
+- **Enabled**: Enable or disable the voicemail box
+
+![Voicemail Settings Continued](../_static/images/voicemail/fusionpbx_voicemail2.jpg)
+
+> **Note**: Starting version 4.2, remote access to voicemail by interrupting the greeting message by pressing "*" and entering the password is disabled by default.
+
+To enable remote access to voicemail:
+
+1. Go to your FusionPBX installation menu.
+2. Navigate to **Advanced**.
+3. Select **Default Settings**.
+4. Go to the **Voicemail** category.
+5. Enable and set `remote_access` to `true`.
+
+## Voicemail Options
+
+To access an extension’s voicemail **away** from the extension:
+
+- Dial the extension and interrupt the greeting with the ***star** key.
+
+| Key       | Description                                      |
+|-----------|--------------------------------------------------|
+| ***97**   | To access **that** extension’s voicemail **from the extension** or the voicemail button |
+| ***98**   | To access **any** extension’s voicemail          |
+| ***99[ext]** | To access **a specific** extension voicemail  |
+
+### Main Menu
+
+| Key       | Action                  |
+|-----------|-------------------------|
+| **press 5** | For advanced options  |
+
+### Advanced Options
+
+| Key       | Action                  |
+|-----------|-------------------------|
+| **press 1** | Record a greeting     |
+| **press 2** | Choose a greeting     |
+| **press 3** | Record name           |
+| **press 6** | Change password       |
+| **press 0** | For main menu         |
+
+### Email Setup/Default Settings
+Click the link for setting up email server settings: [Email Setup/Default Settings](http://docs.fusionpbx.com/en/latest/advanced/default_settings.html#email). These are the settings needed to enable your FusionPBX installation to send email notifications.
+
+### Voicemail Variables
+Using switch variables provides the ability to adjust FusionPBX Voicemail features. These variables can be set in either **Dialplan > Global Variables** or per domain with **Domain Variables Dialplan**.
+
+#### Leave a Voicemail
+
+| Name                      | Value          |
+|---------------------------|----------------|
+| skip_greeting             | true or false  |
+| skip_instructions         | true or false  |
+| voicemail_greeting_number | 0-9            |
+| vm_disk_quota             | 0-3600 seconds |
+| vm_message_ext            | wav or mp3     |
+
+#### Check Voicemails
+
+| Name                      | Value          |
+|---------------------------|----------------|
+| voicemail_authorized      | true or false  |
+| vm_say_caller_id_number   | true or false  |
+| vm_say_date_time          | true or false  |
+
+> **Note**: 'wav' format is the default voicemail message file type. A value of 'mp3' requires *mod_shout* be installed and running.
+
+### Not Found Message
+When an extension is unavailable and no voicemail is configured, there is an option to play a message to the caller alerting them to this.
+
+To enable/disable this, change the option for the **not_found_message** setting in **Advanced > Default Settings > Voicemail** category to suit your preference.
+
+Please note that enabling this option means that the call must be answered in order to play the message to the caller, and so the call will complete with a 200 OK rather than a 480 Unavailable or 486 Busy. In some jurisdictions, this could potentially be illegal as it turns an otherwise toll-free call into a chargeable one.
+
+## Voicemail Transcription
+FusionPBX supports Voicemail Transcription, where emails will include a transcribed version of the voicemail the email was sent in regards to. To configure this feature, see [Voicemail Transcription](http://docs.fusionpbx.com/en/latest/applications/voicemail_transcription.html).

+ 143 - 281
source/applications/voicemail_transcription.md

@@ -1,22 +1,19 @@
 # Voicemail Transcription in FusionPBX 5.3
 
 ## Overview
+This document provides a step-by-step guide for setting up the email transcription service in FusionPBX version 5.3. It supports various service providers such as OpenAI, Google, Azure, or a custom solution.
 
-This document provides a step-by-step guide for setting up the email
-transcription service in FusionPBX version 5.3. It supports various
-service providers such as OpenAI, Google, Azure, or a custom solution.
+## Step-by-Step
+1. **Log in to FusionPBX**
+   - Access the FusionPBX administrative interface.
 
-### Step-by-step
+2. **Check for the "Transcribe" Option**
+   - Navigate to **Advanced** > **Default Settings**.
+   - Use the drop-down filter to select "**Transcribe**".
+   - If the "**Transcribe**" option is already available, skip to step **5**. Otherwise, proceed to step **3**.
 
-1.  **Log in to FusionPBX**
-    -   Access the FusionPBX administrative interface.
-2.  **Check for the \"Transcribe\" Option**
-    -   Navigate to **Advanced** \> **Default Settings**.
-    -   Use the drop-down filter to select \"\*\*Transcribe\*\*\".
-    -   If the \"\*\*Transcribe\*\*\" option is already available, skip
-        to step **5**. Otherwise, proceed to step **3**.
-3.  **Install the Transcribe and Speech Apps**
-    -   SSH into your server and run the following commands:
+3. **Install the Transcribe and Speech Apps**
+   - SSH into your server and run the following commands:
 
         ``` console
         cd /var/www/fusionpbx/app
@@ -25,292 +22,157 @@ service providers such as OpenAI, Google, Azure, or a custom solution.
         chown -R www-data:www-data /var/www/fusionpbx
         php /var/www/fusionpbx/core/upgrade/upgrade.php
         ```
-4.  **Reload the FusionPBX Interface**
-    -   Navigate back to **Advanced** \> **Default Settings**.
-    -   The \"\*\*Transcribe\*\*\" section should now be available.
-5.  **Configure Transcription Settings**
-    -   In the \"\*\*Transcribe\*\*\" category, enable the following
-        settings:
-
-          Subcategory      Type      Value               Enabled   Description
-          ---------------- --------- ------------------- --------- ----------------------------------------------------------------
-          api[key]{#key}   text      secret[key]{#key}   true      Speech to Text API key
-          api[url]{#url}   text      <https://api>\...   false     **\*Leave this alone unless you are using a custom service**\*
-          enabled          boolean   true                true      Speech to Text Enabled
-          engine           text      openai              true      Options: openai, google, azure, custom
-
-    -   Click **Reload** to apply the changes.
-6.  **Enable Transcription for a Single Extension**
-    -   Navigate to **Accounts** \> **Extensions**.
-    -   Select the desired extension.
-    -   Set **Transcription Enabled** to **True**.
-7.  **Enable Transcription by Default for All Extensions**
-    -   Navigate to **Advanced** \> **Default Settings**.
-    -   Use the drop-down filter to select \"\*\*Voicemail\*\*\".
-    -   Enable the **transcription_enabled_default** setting.
-8.  **Test the Service**
-    -   Leave a voicemail for the enabled extension to verify that the
-        transcription works correctly.
-
-:::: note
-::: title
-Note
-:::
-
-The primary function handling voicemail transcriptions is defined in
-[transcribe.php](https://github.com/fusionpbx/fusionpbx/blob/master/app/email_queue/resources/functions/transcribe.php).
-::::
-
-\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\* Voicemail Transcription in FusionPBX
-5.2 and older \*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*
-
-| 
-
-Uses API services to transcribe voicemails into text to be used in the
-app-sms and the voicemail to email options. Bing\'s Speech API or other
-generic APIs can be used.
+        
+4. **Reload the FusionPBX Interface**
+- Navigate back to **Advanced** > **Default Settings**.
+- The "**Transcribe**" section should now be available.
+
+5. **Configure Transcription Settings**
+- In the "**Transcribe**" category, enable the following settings:
+
+| Subcategory | Type   | Value            | Enabled | Description                                           |
+|-------------|--------|------------------|---------|-------------------------------------------------------|
+| api_key     | text   | secret_key       | true    | Speech to Text API key                                |
+| api_url     | text   | https://api...   | false   | ***Leave this alone unless using a custom service*** |
+| enabled     | boolean| true            | true    | Speech to Text Enabled                                |
+| engine      | text   | openai           | true    | Options: openai, google, azure, custom                |
+
+- Click **Reload** to apply the changes.
+
+6. **Enable Transcription for a Single Extension**
+- Navigate to **Accounts** > **Extensions**.
+- Select the desired extension.
+- Set **Transcription Enabled** to **True**.
+
+7. **Enable Transcription by Default for All Extensions**
+- Navigate to **Advanced** > **Default Settings**.
+- Use the drop-down filter to select "**Voicemail**".
+- Enable the **transcription_enabled_default** setting.
+
+8. **Test the Service**
+- Leave a voicemail for the enabled extension to verify that the transcription works correctly.
+
+> **Note**: The primary function handling voicemail transcriptions is defined in [transcribe.php](https://github.com/fusionpbx/fusionpbx/blob/master/app/email_queue/resources/functions/transcribe.php).
+
+---
+
+# Voicemail Transcription in FusionPBX 5.2 and Older
+
+Uses API services to transcribe voicemails into text for use in the app-sms and voicemail-to-email options. Bing's Speech API or other generic APIs can be used.
 
 ## IBM Watson API
+Sign up and language information is located on [IBM Watson's Site](https://cloud.ibm.com/catalog/services/speech-to-text).
 
-Sign up and language information is located on [IBM Watson\'s
-Site](https://cloud.ibm.com/catalog/services/speech-to-text)
-
-:::: warning
-::: title
-Warning
-:::
-
-We cannot use mod[shout]{#shout} to record Voicemails because the
-transcription service needs an uncompressed version of the audio.
-Therefore we will record in WAV and then use LAME to re-encode in MP3.
-This could cause added resource utilization to your system.
-::::
-
-**Goto Advanced \> Default Settings.** Add the following entries
-
-> +---------+-----------------+--------+--------------------+--------+
-> | > C     | > Subcategory   | > Type | > Value            | > E    |
-> | ategory |                 |        |                    | nabled |
-> +=========+=================+========+====================+========+
-> | > vo    | > t             | > text | > watson           | > True |
-> | icemail | ranscribe[provi |        |                    |        |
-> |         | der]{#provider} |        |                    |        |
-> +---------+-----------------+--------+--------------------+--------+
-> | > vo    | > wa            | > text | > { your watson    | > True |
-> | icemail | tson[key]{#key} |        | > key }            |        |
-> +---------+-----------------+--------+--------------------+--------+
-> | > vo    | > wa            | > text | > { watson url }   |        |
-> | icemail | tson[url]{#url} |        | > \| True \|       |        |
-> +---------+-----------------+--------+--------------------+--------+
-> | > vo    | > t             | > text | > en-US            | > True |
-> | icemail | ranscribe[langu |        |                    |        |
-> |         | age]{#language} |        |                    |        |
-> +---------+-----------------+--------+--------------------+--------+
-> | > vo    | >               | > b    | > true             | > True |
-> | icemail |  transcribe[ena | oolean |                    |        |
-> |         | bled]{#enabled} |        |                    |        |
-> +---------+-----------------+--------+--------------------+--------+
-> | > vo    | > json[ena      | > b    | > true             | > True |
-> | icemail | bled]{#enabled} | oolean |                    |        |
-> +---------+-----------------+--------+--------------------+--------+
-
-*NOTE: Watson URL used for testing was the following:
-https://example.url.api.us-south.speech-to-text.watson.cloud.ibm.com/instances/{GUID}/v1/recognize?model=en-US_Telephony&smart_formatting=true*
+> **Warning**: We cannot use mod_shout to record voicemails because the transcription service needs an uncompressed version of the audio. Therefore, we will record in WAV and then use LAME to re-encode in MP3. This could cause added resource utilization to your system.
+
+**Go to Advanced > Default Settings.**  
+Add the following entries:
 
-**List of available IBM Watson speech to text models**
-<https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-models>
+| Category  | Subcategory          | Type   | Value               | Enabled |
+|-----------|----------------------|--------|---------------------|---------|
+| voicemail | transcribe_provider  | text   | watson              | True    |
+| voicemail | watson_key           | text   | {your watson key}   | True    |
+| voicemail | watson_url           | text   | {watson url}        | True    |
+| voicemail | transcribe_language  | text   | en-US               | True    |
+| voicemail | transcribe_enabled   | boolean| true               | True    |
+| voicemail | json_enabled         | boolean| true               | True    |
 
-> Click \"Reload\" at the top of the page.
+*NOTE: Watson URL used for testing was:  
+https://example.url.api.us-south.speech-to-text.watson.cloud.ibm.com/instances/{GUID}/v1/recognize?model=en-US_Telephony&smart_formatting=true*
+
+**List of available IBM Watson speech-to-text models**:  
+[https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-models](https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-models)
 
-**Goto Status \> Sip Status.**
+Click "Reload" at the top of the page.
 
-Click \"Flush Cache\", \"Reload XML\" and \"Rescan\".
+**Go to Status > SIP Status.**  
+Click "Flush Cache", "Reload XML", and "Rescan".
 
-If you entered your key\'s correctly, you should now start getting
-transcriptions delivered in your voicemail to email and you will also
-see them on the Messages page.
+If you entered your keys correctly, you should now start getting transcriptions delivered in your voicemail-to-email and see them on the Messages page.
 
 ## Azure API
+**Go to Advanced > Default Settings.**  
+Add the following entries:
+
+| Category  | Subcategory          | Type   | Value                 | Enabled |
+|-----------|----------------------|--------|-----------------------|---------|
+| voicemail | transcribe_provider  | text   | azure                 | True    |
+| voicemail | azure_key            | text   | {your azure key}      | True    |
+| voicemail | azure_server_region  | text   | {your server region}  | True    |
+| voicemail | transcribe_language  | text   | en-US                 | True    |
+| voicemail | transcribe_enabled   | boolean| true                 | True    |
+| voicemail | json_enabled         | boolean| true                 | True    |
+
+Click "Reload" at the top of the page.
 
-**Goto Advanced \> Default Settings.** Add the following entries
-
-> +---------+-----------------+--------+--------------------+--------+
-> | > C     | > Subcategory   | > Type | > Value            | > E    |
-> | ategory |                 |        |                    | nabled |
-> +=========+=================+========+====================+========+
-> | > vo    | > t             | > text | > azure            | > True |
-> | icemail | ranscribe[provi |        |                    |        |
-> |         | der]{#provider} |        |                    |        |
-> +---------+-----------------+--------+--------------------+--------+
-> | > vo    | > a             | > text | > { your azure key | > True |
-> | icemail | zure[key]{#key} |        | > }                |        |
-> +---------+-----------------+--------+--------------------+--------+
-> | > vo    | > azure[        | > text | > { your server    | > True |
-> | icemail | server_region]{ |        | > region }         |        |
-> |         | #server_region} |        |                    |        |
-> +---------+-----------------+--------+--------------------+--------+
-> | > vo    | > t             | > text | > en-US            | > True |
-> | icemail | ranscribe[langu |        |                    |        |
-> |         | age]{#language} |        |                    |        |
-> +---------+-----------------+--------+--------------------+--------+
-> | > vo    | >               | > b    | > true             | > True |
-> | icemail |  transcribe[ena | oolean |                    |        |
-> |         | bled]{#enabled} |        |                    |        |
-> +---------+-----------------+--------+--------------------+--------+
-> | > vo    | > json[ena      | > b    | > true             | > True |
-> | icemail | bled]{#enabled} | oolean |                    |        |
-> +---------+-----------------+--------+--------------------+--------+
->
-> Click \"Reload\" at the top of the page.
-
-**Goto Status \> Sip Status.**
-
-Click \"Flush Cache\", \"Reload XML\" and \"Rescan\".
-
-If you entered your key\'s correctly, you should now start getting
-transcriptions delivered in your voicemail to email and you will also
-see them on the Messages page.
+**Go to Status > SIP Status.**  
+Click "Flush Cache", "Reload XML", and "Rescan".
+
+If you entered your keys correctly, you should now start getting transcriptions delivered in your voicemail-to-email and see them on the Messages page.
 
 ## Google API
+**Go to Advanced > Default Settings.**  
+Add the following entries:
 
-**Goto Advanced \> Default Settings.** Add the following entries
-
-> +---------+-----------------+--------+--------------------+--------+
-> | > C     | > Subcategory   | > Type | > Value            | > E    |
-> | ategory |                 |        |                    | nabled |
-> +=========+=================+========+====================+========+
-> | > vo    | > t             | > text | > google           | > True |
-> | icemail | ranscribe[provi |        |                    |        |
-> |         | der]{#provider} |        |                    |        |
-> +---------+-----------------+--------+--------------------+--------+
-> | > vo    | > go            | > text | > { your google    | > True |
-> | icemail | ogle[key]{#key} |        | > key }            |        |
-> +---------+-----------------+--------+--------------------+--------+
-> | > vo    | > go            | > text | > { your google    | > True |
-> | icemail | ogle[url]{#url} |        | > url }            |        |
-> +---------+-----------------+--------+--------------------+--------+
-> | > vo    | > t             | > text | > en-US            | > True |
-> | icemail | ranscribe[langu |        |                    |        |
-> |         | age]{#language} |        |                    |        |
-> +---------+-----------------+--------+--------------------+--------+
-> | > vo    | >               | > b    | > true             | > True |
-> | icemail |  transcribe[ena | oolean |                    |        |
-> |         | bled]{#enabled} |        |                    |        |
-> +---------+-----------------+--------+--------------------+--------+
-> | > vo    | > json[ena      | > b    | > true             | > True |
-> | icemail | bled]{#enabled} | oolean |                    |        |
-> +---------+-----------------+--------+--------------------+--------+
->
-> Click \"Reload\" at the top of the page.
-
-**Goto Status \> Sip Status.**
-
-Click \"Flush Cache\", \"Reload XML\" and \"Rescan\".
-
-If you entered your key\'s correctly, you should now start getting
-transcriptions delivered in your voicemail to email and you will also
-see them on the Messages page.
+| Category  | Subcategory          | Type   | Value               | Enabled |
+|-----------|----------------------|--------|---------------------|---------|
+| voicemail | transcribe_provider  | text   | google              | True    |
+| voicemail | google_key           | text   | {your google key}   | True    |
+| voicemail | google_url           | text   | {your google url}   | True    |
+| voicemail | transcribe_language  | text   | en-US               | True    |
+| voicemail | transcribe_enabled   | boolean| true               | True    |
+| voicemail | json_enabled         | boolean| true               | True    |
 
-## Bing API
+Click "Reload" at the top of the page.
+
+**Go to Status > SIP Status.**  
+Click "Flush Cache", "Reload XML", and "Rescan".
+
+If you entered your keys correctly, you should now start getting transcriptions delivered in your voicemail-to-email and see them on the Messages page.
 
+## Bing API
 Recommend using Azure as an alternative to Bing.
 
-Sign up and language information is located on [Microsoft
-Site](https://www.microsoft.com/cognitive-services/en-us/Speech-api/documentation/API-Reference-REST/BingVoiceRecognition)
-Note: The Bing Speech API is deprecated as of October 2018, this works
-for now but needs to be ported to [the new
-API](https://github.com/MicrosoftDocs/azure-docs/blob/master/articles/cognitive-services/Speech-Service/how-to-migrate-from-bing-speech.md)
-
-:::: warning
-::: title
-Warning
-:::
-
-We cannot use mod[shout]{#shout} to record Voicemails because the
-transcription service needs an uncompressed version of the audio.
-Therefore we will record in WAV and then use LAME to re-encode in MP3.
-This could cause added resource utilization to your system.
-::::
-
-**Goto Advanced \> Default Settings.** Add the following entries
-
-> +---------+-----------------+--------+--------------------+--------+
-> | > C     | > Subcategory   | > Type | > Value            | > E    |
-> | ategory |                 |        |                    | nabled |
-> +=========+=================+========+====================+========+
-> | > vo    | > t             | > text | > microsoft        | > True |
-> | icemail | ranscribe[provi |        |                    |        |
-> |         | der]{#provider} |        |                    |        |
-> +---------+-----------------+--------+--------------------+--------+
-> | > vo    | > microso       | > text | > {your microsoft  | > True |
-> | icemail | ft[key1]{#key1} |        | > key #1}          |        |
-> +---------+-----------------+--------+--------------------+--------+
-> | > vo    | > microso       | > text | > {your microsoft  | > True |
-> | icemail | ft[key2]{#key2} |        | > key #2}          |        |
-> +---------+-----------------+--------+--------------------+--------+
-> | > vo    | > t             | > text | > en-US            | > True |
-> | icemail | ranscribe[langu |        |                    |        |
-> |         | age]{#language} |        |                    |        |
-> +---------+-----------------+--------+--------------------+--------+
-> | > vo    | >               | > b    | > true             | > True |
-> | icemail |  transcribe[ena | oolean |                    |        |
-> |         | bled]{#enabled} |        |                    |        |
-> +---------+-----------------+--------+--------------------+--------+
->
-> Click \"Reload\" at the top of the page.
-
-**Goto Status \> Sip Status.**
-
-Click \"Flush Cache\", \"Reload XML\" and \"Rescan\".
-
-If you entered your key\'s correctly, you should now start getting
-transcriptions delivered in your voicemail to email and you will also
-see them on the Messages page.
+Sign up and language information is located on [Microsoft Site](https://www.microsoft.com/cognitive-services/en-us/Speech-api/documentation/API-Reference-REST/BingVoiceRecognition).  
+Note: The Bing Speech API is deprecated as of October 2018; this works for now but needs to be ported to [the new API](https://github.com/MicrosoftDocs/azure-docs/blob/master/articles/cognitive-services/Speech-Service/how-to-migrate-from-bing-speech.md).
+
+> **Warning**: We cannot use mod_shout to record voicemails because the transcription service needs an uncompressed version of the audio. Therefore, we will record in WAV and then use LAME to re-encode in MP3. This could cause added resource utilization to your system.
+
+**Go to Advanced > Default Settings.**  
+Add the following entries:
+
+| Category  | Subcategory          | Type   | Value                 | Enabled |
+|-----------|----------------------|--------|-----------------------|---------|
+| voicemail | transcribe_provider  | text   | microsoft             | True    |
+| voicemail | microsoft_key1       | text   | {your microsoft key #1} | True  |
+| voicemail | microsoft_key2       | text   | {your microsoft key #2} | True  |
+| voicemail | transcribe_language  | text   | en-US                 | True    |
+| voicemail | transcribe_enabled   | boolean| true                 | True    |
+
+Click "Reload" at the top of the page.
+
+**Go to Status > SIP Status.**  
+Click "Flush Cache", "Reload XML", and "Rescan".
+
+If you entered your keys correctly, you should now start getting transcriptions delivered in your voicemail-to-email and see them on the Messages page.
 
 ## Custom API
+Currently does not work with the FusionPBX email_queue.
+
+API info from the Speech-to-Text provider of your choice is needed, or you can self-host a transcription engine like [Mozilla DeepSpeech](https://git.callpipe.com/fusionpbx/deepspeech_frontend) or [Kaldi ASR](https://github.com/dialogflow/asr-server).
+
+**Go to Advanced > Default Settings.**  
+Add the following entries:
+
+| Category  | Subcategory          | Type   | Value            | Enabled | Required? |
+|-----------|----------------------|--------|------------------|---------|-----------|
+| voicemail | transcribe_provider  | text   | custom           | True    |           |
+| voicemail | transcription_server | text   | https://yourserver | True  |           |
+| voicemail | json_enabled         | boolean| true            | True    | Optional  |
+| voicemail | api_key              | text   | your_api_key     | True    | Optional  |
+| voicemail | transcribe_language  | text   | en-US            | True    |           |
+| voicemail | transcribe_enabled   | boolean| true            | True    |           |
+
+Click "Reload" at the top of the page.
 
-Currently does not work with the FusionPBX email[queue]{#queue}.
-
-API info from the Speech to Text provider of your choice is needed, or
-you can self host a transcription engine like [Mozilla
-DeepSpeech](https://git.callpipe.com/fusionpbx/deepspeech_frontend) or
-[Kaldi ASR](https://github.com/dialogflow/asr-server)
-
-**Goto Advanced \> Default Settings.** Add the following entries
-
-> +--------+---------------+-------+-----------------+-------+--------+
-> | > Ca   | > Subcategory | >     | > Value         | > En  | > Req  |
-> | tegory |               |  Type |                 | abled | uired? |
-> +========+===============+=======+=================+=======+========+
-> | > voi  | > trans       | >     | > custom        | >     |        |
-> | cemail | cribe[provide |  text |                 |  True |        |
-> |        | r]{#provider} |       |                 |       |        |
-> +--------+---------------+-------+-----------------+-------+--------+
-> | > voi  | > tran        | >     | > <http         | >     |        |
-> | cemail | scription[ser |  text | s://yourserver> |  True |        |
-> |        | ver]{#server} |       |                 |       |        |
-> +--------+---------------+-------+-----------------+-------+--------+
-> | > voi  | > json[enabl  | > bo  | > true          | >     | > Op   |
-> | cemail | ed]{#enabled} | olean |                 |  True | tional |
-> +--------+---------------+-------+-----------------+-------+--------+
-> | > voi  | > a           | text  | > your[api      | >     | > Op   |
-> | cemail | pi[key]{#key} |       | _key]{#api_key} |  True | tional |
-> +--------+---------------+-------+-----------------+-------+--------+
-> | > voi  | > trans       | >     | > en-US         | >     |        |
-> | cemail | cribe[languag |  text |                 |  True |        |
-> |        | e]{#language} |       |                 |       |        |
-> +--------+---------------+-------+-----------------+-------+--------+
-> | > voi  | > tra         | > bo  | > true          | >     |        |
-> | cemail | nscribe[enabl | olean |                 |  True |        |
-> |        | ed]{#enabled} |       |                 |       |        |
-> +--------+---------------+-------+-----------------+-------+--------+
->
-> Click \"Reload\" at the top of the page.
-
-**Goto Status \> Sip Status.**
-
-Click \"Flush Cache\", \"Reload XML\" and \"Rescan\".
-
-If you entered your key\'s correctly, you should now start getting
-transcriptions delivered in your voicemail to email and you will also
-see them on the Messages page.
+**Go to Status > SIP Status.**

+ 6 - 11
source/dialplan/advanced_dialplans.md

@@ -45,17 +45,12 @@ prefix can be any number that you choose to use and the 4 digit
 extension must match the destination tenant. So if the destination
 extensions are 3 digit then you would use 3 instead of 4.
 
-  ---------------------------------------------------------------------------------------------------------------------------------
-  Tag         Type                               Data                                              Break   Inline   Group   Order
-  ----------- ---------------------------------- ------------------------------------------------- ------- -------- ------- -------
-  condition   \${destination[number]{#number}}   \^5(d{4})\$                                                                5
-
-  action      set                                domain[name]{#name}=customer.domain.tld                   True             10
-
-  action      set                                domain[uuid]{#uuid}=correct-uuid-for-the-domain           True             15
-
-  action      transfer                           \$1 XML \${domain[name]{#name}}                                            20
-  ---------------------------------------------------------------------------------------------------------------------------------
+| Tag       | Type                  | Data                                | Dialplan Detail Break | Break | Inline | Group | Order |
+|-----------|-----------------------|--------------------------------------|-----------------------|-------|--------|-------|-------|
+| condition | ${destination_number} | ^5(\d{4})$                         |                       |       |        |       | 5     |
+| action    | set                   | domain_name=customer.domain.tld     |                       |       | True   |       | 10    |
+| action    | set                   | domain_uuid=correct-uuid-for-the-domain |                   |       | True   |       | 15    |
+| action    | transfer              | 1XML {domain_name}                  |                       |       |        |       | 20    |
 
 -   Be sure to set the **Continue dropdown box True**
 -   Finally we have the desired dialplan to call from tenant A to tenant

+ 4 - 6
source/dialplan/outbound_routes.md

@@ -2,7 +2,7 @@
 
 Route outbound calls to gateways, tdm, enum and more. When a call
 matches the conditions the call to outbound routes. [Check out the
-youtube video](https://youtu.be/rhyfCKLBI-Y) .
+youtube video](https://youtu.be/rhyfCKLBI-Y).
 
 <div style="text-align: center; margin-bottom: 2em;">
 <iframe width="100%" height="350" src="https://www.youtube.com/embed/rhyfCKLBI-Y?rel=0" frameborder="0" ; encrypted-media" allowfullscreen></iframe>
@@ -20,17 +20,15 @@ youtube video](https://youtu.be/rhyfCKLBI-Y) .
 
 ![image](../_static/images/dialplan/fusionpbx_outbound_routes2.png)
 
-## 
-
+```
     Gateway: VoiceTel
     Dialplan Expression: ^(?:\+?1)?(\d{10})$ (You can also choose more than one from the drop down list also as needed)
     Order: 000
     Enabled: true
     Description: VoiceTel-out
+```
 
-## 
-
-**By using** [VoiceTel](http://tiny.cc/voicetel) **you help support
+**By using [VoiceTel](http://tiny.cc/voicetel) you help support
 FusionPBX. Thank you for your support!**
 
 ### Pin Numbers

+ 3 - 5
source/getting_started/languages.md

@@ -67,7 +67,7 @@ installation method.
 -   **Free:**
     <https://freeswitch.org/stash/projects/FS/repos/freeswitch-sounds/browse>
 
-## app[languages.php]{#languages.php}
+## app_languages.php
 
 **Guidelines** The words used in the text variable name
 
@@ -109,8 +109,7 @@ following.
 
 **Example File**
 
-An excerpt from the app[languages.php]{#languages.php} for Conference
-Center.
+An excerpt from the app_languages.php for Conference Center.
 
     <?php
 
@@ -155,8 +154,7 @@ Center.
 
     ?>
 
-To use inside the code on each page that displays text. Place the
-following code at the top just after the permision[exists]{#exists}
+To use inside the code on each page that displays text. Place the following code at the top just after the permision_exists
 
     //add multi-lingual support
        require_once "app_languages.php";

+ 2 - 4
source/getting_started/lets_encrypt.md

@@ -163,14 +163,12 @@ example.
 Tip
 :::
 
-Use the dig command to check that the txt record is correct. dig -t txt
-[acme-challenge.domain.tld]{#acme-challenge.domain.tld}
+Use the dig command to check that the txt record is correct. dig -t txt _acme-challenge.domain.tld
 
 Output should show:
 
 ;; ANSWER SECTION:
-[acme-challenge.domain.tld]{#acme-challenge.domain.tld}. 1799 IN TXT
-\"PY7ttk6no[5eG7WtAbO6qs5-NzA-Kigko375omKc0nw]{#eg7wtabo6qs5-nza-kigko375omkc0nw}\"
+_acme-challenge.domain.tld. 1799 IN TXT  "PY7ttk6no_5eG7WtAbO6qs5-NzA-Kigko375omKc0nw"
 ::::
 
 ##### Setup for multiple domains on Let\'s Encrypt