corrected MD formatting errors

Author: systemcrash

docs/Asterisk-Community/Asterisk-Issue-Guidelines.md

@@ -18,7 +18,7 @@ Feature requests without patches are generally not accepted through the issue tr
 See the forums, mailing lists, IRC channels, or this wiki. For even more information, see <http://www.asterisk.org/community>
 * **Support requests**: (My phone doesn't register! My database connectivity doesn't work! How do I get it to work?)  
 Search and ask on the forums, mailing lists, and IRC. Again, see <http://www.asterisk.org/community> for more information.
-* **Random wishes and feature requests with no patch:** (I want Asterisk to support <insert obscure protocol or gadget>, but I don't know how to code!)  
+* **Random wishes and feature requests with no patch:** (I want Asterisk to support `<insert obscure protocol or gadget>`, but I don't know how to code!)  
 See the [How to request a feature section](#how-to-request-a-feature-or-improvement) for more information on requesting a feature.
 * **Business development requests** (I will pay you to make Asterisk support fancy unicorn protocol!)  
 Please head to the Commercial category at <https://community.asterisk.org/>. If what you want is a specific feature or bug fixed, you may want to consider [requesting a bug bounty](/Development/Asterisk-Bug-Bounties).

docs/Configuration/Applications/Conferencing-Applications/ConfBridge/ConfBridge-CLI-Commands.md

@@ -8,7 +8,7 @@ ConfBridge CLI Commands
 
 ConfBridge offers several commands that may be invoked from the Asterisk CLI.
 
-confbridge kick <conference> <channel>
+`confbridge kick <conference> <channel>`
 --------------------------------------
 
 Removes the specified channel from the conference, e.g.:
@@ -19,7 +19,7 @@ Kicking SIP/mypeer-00000000 from confbridge 1111
 
 ```
 
-On This Pageconfbridge list
+`On This Pageconfbridge list`
 ---------------
 
 Shows a summary listing of all bridges, e.g.:
@@ -32,7 +32,7 @@ Conference Bridge Name Users Marked Locked?
 
 ```
 
-confbridge list <conference>
+`confbridge list <conference>`
 ----------------------------
 
 Shows a detailed listing of participants in a specified conference, e.g.:
@@ -45,7 +45,7 @@ SIP/mypeer-00000001 default_user 1111 sample_user_menu
 
 ```
 
-confbridge lock <conference>
+`confbridge lock <conference>`
 ----------------------------
 
 Locks a specified conference so that only Admin users can join, e.g.:
@@ -56,7 +56,7 @@ Conference 1111 is locked.
 
 ```
 
-confbridge unlock <conference>
+`confbridge unlock <conference>`
 ------------------------------
 
 Unlocks a specified conference so that only Admin users can join, e.g.:
@@ -67,7 +67,7 @@ Conference 1111 is unlocked.
 
 ```
 
-confbridge mute <conference> <channel>
+`confbridge mute <conference> <channel>`
 --------------------------------------
 
 Mutes a specified user in a specified conference, e.g.:
@@ -78,7 +78,7 @@ Muting SIP/mypeer-00000001 from confbridge 1111
 
 ```
 
-confbridge unmute <conference> <channel>
+`confbridge unmute <conference> <channel>`
 ----------------------------------------
 
 Unmutes a specified user in a specified conference, e.g.:
@@ -89,7 +89,7 @@ Unmuting SIP/mypeer-00000001 from confbridge 1111
 
 ```
 
-confbridge record start <conference> <file>
+`confbridge record start <conference> <file>`
 -------------------------------------------
 
 Begins recording a conference. If "file" is specified, it will be used, otherwise, the Bridge Profile record_file will be used. If the Bridge Profile does not specify a record_file, one will be automatically generated in Asterisk's monitor directory. Usage:
@@ -101,7 +101,7 @@ Recording started
 
 ```
 
-confbridge record stop <confererence>
+`confbridge record stop <confererence>`
 -------------------------------------
 
 Stops recording the specified conference, e.g.:
@@ -114,7 +114,7 @@ Recording stopped.
 
 ```
 
-confbridge show menus
+`confbridge show menus`
 ---------------------
 
 Shows a listing of Conference Menus as defined in confbridge.conf, e.g.:
@@ -127,7 +127,7 @@ sample_user_menu
 
 ```
 
-confbridge show menu <menu name>
+`confbridge show menu <menu name>`
 --------------------------------
 
 Shows a detailed listing of a named Conference Menu, e.g.:
@@ -147,7 +147,7 @@ Name: sample_admin_menu
 
 ```
 
-confbridge show profile bridges
+`confbridge show profile bridges`
 -------------------------------
 
 Shows a listing of Bridge Profiles as defined in confbridge.conf, e.g.:
@@ -160,7 +160,7 @@ default_bridge
 
 ```
 
-confbridge show profile bridge <bridge>
+`confbridge show profile bridge <bridge>`
 ---------------------------------------
 
 Shows a detailed listing of a named Bridge Profile, e.g.:
@@ -193,7 +193,7 @@ sound_error_menu: conf-errormenu
 
 ```
 
-confbridge show profile users
+`confbridge show profile users`
 -----------------------------
 
 Shows a listing of User Profiles as defined in confbridge.conf, e.g.:
@@ -206,7 +206,7 @@ default_user
 
 ```
 
-confbirdge show profile user <user>
+`confbirdge show profile user <user>`
 -----------------------------------
 
 Shows a detailed listing of a named Bridge Profile, e.g.:

docs/Configuration/Applications/Conferencing-Applications/ConfBridge/ConfBridge-Configuration.md

@@ -53,7 +53,7 @@ A Bridge Profile provides the following configuration options:
 | --- | --- | --- | --- |
 | type | bridge | Set this to bridge to configure a bridge profile |   |
 | max_members | integer; e.g. 50 | Limits the number of participants for a single conference to a specific number. By default, conferences have no participant limit. After the limit is reached, the conference will be locked until someone leaves. Admin-level users are exempt from this limit and will still be able to join otherwise-locked, because of limit, conferences. |   |
-| record_conference | yes/no | Records the conference call starting when the first user enters the room, and ending when the last user exits the room. The default recorded filename is 'confbridge-<name of conference bridge>-<start time>.wav and the default format is 8kHz signed linear. By default, this option is disabled. This file will be located in the configured monitoring directory as set in asterisk.conf |   |
+| record_conference | yes/no | Records the conference call starting when the first user enters the room, and ending when the last user exits the room. The default recorded filename is `confbridge-<name of conference bridge>-<start time>.wav` and the default format is 8kHz signed linear. By default, this option is disabled. This file will be located in the configured monitoring directory as set in asterisk.conf |   |
 | record_file | path, e.g. /tmp/myfiles | When record_conference is set to yes, the specific name of the recorded file can be set using this option. Note that since multiple conferences may use the same Bridge profile, this can cause issues, depending on the configuration. It is recommended to only use this option dynamically with the CONFBRIDGE() dialplan function. This allows the recorded name to be specified and a unique name to be chosen. By default, the recorded file is stored in Asterisk's spool/monitory directory, with a unique filename starting with the 'confbridge' prefix. |   |
 | internal_sample_rate | auto, 8000, 12000, 16000, 24000, 32000, 44100, 48000, 96000, 192000 | Sets the internal native sample rate at which to mix the conference. The "auto" option allows Asterisk to adjust the sample rate to the best quality / performance based on the participant makeup. Numbered values lock the rate to the specified numerical rate. If a defined number does not match an internal sampling rate supported by Asterisk, the nearest sampling rate will be used instead. |   |
 | mixing_interval | 10, 20, 40, 80 | Sets, in milliseconds, the internal mixing interval. By default, the mixing interval of a bridge is 20ms. This setting reflects how "tight" or "loose" the mixing will be for the conference. Lower intervals provide a "tighter" sound with less delay in the bridge and consume more system resources. Higher intervals provide a "looser" sound with more delay in the bridge and consume less resources |   |
@@ -68,7 +68,7 @@ A Bridge Profile provides the following configuration options:
 | sound_only_person | filename | The sound played when a user is the only person in the conference. |   |
 | sound_only_one | filename | The sound played to a user when there is only one other person in the conference. |   |
 | sound_there_are | filename | The sound played when announcing how many users there are in a conference. |   |
-| sound_other_in_party | filename | Used in conjunction with the sound_there_are option, used like "sound_there_are" <number of participants> "sound_other_in_party" |   |
+| sound_other_in_party | filename | Used in conjunction with the sound_there_are option, used like `"sound_there_are" <number of participants> "sound_other_in_party"` |   |
 | sound_place_into_conference | filename | The sound played when someone is placed into a conference, after waiting for a marked user. |   |
 | sound_wait_for_leader | filename | The sound played when a user is placed into a conference that cannot start until a marked user enters. |   |
 | sound_leader_has_left | filename | The sound played when the last marked user leaves the conference. |   |
@@ -144,8 +144,8 @@ A Conference Menu provides the following configuration options:
 | Option | Values | Description | Notes |
 | --- | --- | --- | --- |
 | type | menu | Set this to menu to configure a conference menu |   |
-| playback | (<name of audio file1>&<name of audio file2>&...) | Plays back an audio file, or a string of audio files chained together using the & character, to the user and then immediately returns them to the conference. |   |
-| playback_and_continue | (<name of audio file 1>&<name of audio file 2>&...) | Plays back an audio file, or a series of audio files chained together using the & character, to the user while continuing the collect the DTMF sequence. This is useful when using a menu prompt that describes all of the menu options. Note that any DTMF during this action will terminate the prompt's playback. |   |
+| playback | (`<name of audio file1>&<name of audio file2>&...`) | Plays back an audio file, or a string of audio files chained together using the & character, to the user and then immediately returns them to the conference. |   |
+| playback_and_continue | (`<name of audio file 1>&<name of audio file 2>&...`) | Plays back an audio file, or a series of audio files chained together using the & character, to the user while continuing the collect the DTMF sequence. This is useful when using a menu prompt that describes all of the menu options. Note that any DTMF during this action will terminate the prompt's playback. |   |
 | toggle_mute |   | Toggles mute on and off. When a user is muted, they will not be able to speak to other conference users, but they can still listen to other users. While muted, DTMF keys from the caller will continue to be collected. |   |
 | no_op |   | This action does nothing. Its only real purpose exists for being able to reserve a sequence in the configuration as a menu exit sequence. |   |
 | decrease_listening_volume |   | Decreases the caller's listening volume. Everything they hear will sound quieter. |   |

docs/Configuration/Applications/SMS.md

@@ -32,13 +32,13 @@ All text messages are stored in /var/spool/asterisk/sms
 
 A log is recorded in /var/log/asterisk/sms
 
-There are two subdirectories called sc-me.<queuename> holding all messages from service centre to phone, and me-sc.<queuename> holding all messages from phone to service centre.
+There are two subdirectories called `sc-me.<queuename>` holding all messages from service centre to phone, and `me-sc.<queuename>` holding all messages from phone to service centre.
 
 In each directory are messages in files, one per file, using any filename not starting with a dot.
 
-When connected as a service centre, SMS(s) will send all messages waiting in the sc-me-<queuename> directory, deleting the files as it goes. Any received in this mode are placed in the me-sc-<queuename> directory.
+When connected as a service centre, SMS(s) will send all messages waiting in the `sc-me-<queuename>` directory, deleting the files as it goes. Any received in this mode are placed in the `me-sc-<queuename>` directory.
 
-When connected as a client, SMS() will send all messages waiting in the me-sc-<queuename> directory, deleting the files as it goes. Any received in this mode are placed in the sc-me-<queuename> directory.
+When connected as a client, SMS() will send all messages waiting in the `me-sc-<queuename>` directory, deleting the files as it goes. Any received in this mode are placed in the `sc-me-<queuename>` directory.
 
 Message files created by SMS() use a time stamp/reference based filename.
 

docs/Configuration/Applications/Voicemail/Message-Waiting-Indication.md

@@ -59,7 +59,7 @@ Asterisk can subscribe to receive MWI from another SIP server and store it local
 
 ```
 
-MWI received will be stored in the 1234 mailbox of the SIP_Remote context. It can be used by other phones by setting their SIP peers "mailbox" option to the <mailbox_number>@SIP_Remote. e.g. mailbox=1234@SIP_Remote
+MWI received will be stored in the 1234 mailbox of the SIP_Remote context. It can be used by other phones by setting their SIP peers "mailbox" option to the `<mailbox_number>@SIP_Remote`. e.g. `mailbox=1234@SIP_Remote`
 
 Reception of unsolicited MWI NOTIFY with chan_sip
 --------------------------------------------------
@@ -72,7 +72,7 @@ A chan_sip peer can be configured to receive unsolicited MWI NOTIFY messages and
 
 ```
 
-If the remote SIP server sends an unsolicited MWI NOTIFY message the new/old message count will be stored in the configured virtual mailbox. It can be used by any device supporting MWI by specifying mailbox=<configured value>@SIP_Remote as the mailbox for the desired SIP peer.
+If the remote SIP server sends an unsolicited MWI NOTIFY message the new/old message count will be stored in the configured virtual mailbox. It can be used by any device supporting MWI by specifying `mailbox=<configured value>@SIP_Remote` as the mailbox for the desired SIP peer.
 
 res_external_mwi
 ------------------
@@ -91,4 +91,4 @@ External sources can use the API provided by res_external_mwi to communicate MWI
 chan_pjsip
 -----------
 
-The endpoint parameter `incoming_mwi_mailbox` (introduced in 13.18.0 and 14.7.0) takes a <`mailbox>@<context>` value.  When an unsolicited NOTIFY message is received ***from*** this endpoint with an event type of `message-summary` and the `incoming_mwi_mailbox` parameter is set, Asterisk will automatically publish the new/old message counts for the specified mailbox on the internal stasis bus for any other module to use.  For instance, if you have an analog phone and you specify `mailbox=userx@default` in chan_dahdi.conf, when a NOTIFY comes in on a pjsip endpoint with `incoming_mwi_mailbox=userx@default`, chan_dahdi will automatically pick that up and turn the MWI light on on the analog phone.
+The endpoint parameter `incoming_mwi_mailbox` (introduced in 13.18.0 and 14.7.0) takes a `<mailbox>@<context>` value.  When an unsolicited NOTIFY message is received ***from*** this endpoint with an event type of `message-summary` and the `incoming_mwi_mailbox` parameter is set, Asterisk will automatically publish the new/old message counts for the specified mailbox on the internal stasis bus for any other module to use.  For instance, if you have an analog phone and you specify `mailbox=userx@default` in chan_dahdi.conf, when a NOTIFY comes in on a pjsip endpoint with `incoming_mwi_mailbox=userx@default`, chan_dahdi will automatically pick that up and turn the MWI light on on the analog phone.

docs/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/PJSIP-Configuration-Sections-and-Relationships.md

@@ -28,7 +28,7 @@ Reference documentation for all configuration parameters:
 * [Configuration options for outbound registration, provided by res_pjsip_outbound_registration](/Latest_API/API_Documentation/Module_Configuration/res_pjsip_outbound_registration)
 * [Configuration options for endpoint identification by IP address, provided by res_pjsip_endpoint_identifier_ip](/Latest_API/API_Documentation/Module_Configuration/res_pjsip_endpoint_identifier_ip)
 
-The same documentation is available at the Asterisk CLI as well. You can use "config show help <res_pjsip module name> <configobject> <configoption>" to get help on a particular option. That help will typically describe the default value for an option as well.
+The same documentation is available at the Asterisk CLI as well. You can use `config show help <res_pjsip module name> <configobject> <configoption>` to get help on a particular option. That help will typically describe the default value for an option as well.
 
 /// tip|Defaults
 For many config options, it's very helpful to understand their default behavior. For example, for the endpoint section "transport=" option, if no value is assigned then Asterisk will \*DEFAULT\* to the first configured transport in pjsip.conf which is valid for the URI we are trying to contact.

docs/Configuration/Interfaces/Back-end-Database-and-Realtime-Connectivity/ODBC/Configuring-res_odbc.md

@@ -22,7 +22,7 @@ Other pages on the wiki describe that process:
 
 [Using Menuselect to Select Asterisk Options](/Getting-Started/Installing-Asterisk/Installing-Asterisk-From-Source/Using-Menuselect-to-Select-Asterisk-Options)
 
-When using menuselect, verify that the **func_odbc** (you'll probably be using that one), **res_odbc** (required) and **res_odbc_transaction** (required) modules will be built. Then, build Asterisk and make sure those modules were built and exist in  */usr/lib/asterisk/modules** (or whatever directory you use).
+When using menuselect, verify that the **func_odbc** (you'll probably be using that one), **res_odbc** (required) and **res_odbc_transaction** (required) modules will be built. Then, build Asterisk and make sure those modules were built and exist in  **/usr/lib/asterisk/modules** (or whatever directory you use).
 
 Configure res_odbc.conf to connect to your ODBC installation
 =============================================================

docs/Contributing-to-the-Documentation.md

@@ -637,4 +637,4 @@ Now run `make`:
 $ make BRANCHES=22
 ```
 
-The main Makefile will automatically include the Makefile.<branch>.inc file for any branch listed in BRANCHES.  It doesn't make sense to build more than one branch in this situation but you could if you wanted to.  You'll have to make sure there's a `Makefile.<branch>.inc` file for each branch you want to build and that the paths in each file are adjusted to that branch's Asterisk files.
+The main Makefile will automatically include the `Makefile.<branch>.inc` file for any branch listed in BRANCHES.  It doesn't make sense to build more than one branch in this situation but you could if you wanted to.  You'll have to make sure there's a `Makefile.<branch>.inc` file for each branch you want to build and that the paths in each file are adjusted to that branch's Asterisk files.