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Fixes #163

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gtjoseph merged 3 commits into from
Dec 4, 2025
Merged

Fixes #163

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843d0e9
grammar fixes
systemcrash Dec 3, 2025
f5fd5cd
corrected MD formatting errors
systemcrash Dec 3, 2025
513ebe0
corrected MD formatting errors - replace <0xa0> with space
systemcrash Dec 3, 2025
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2 changes: 1 addition & 1 deletion docs/Asterisk-Community/Asterisk-Issue-Guidelines.md
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Expand Up @@ -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).
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2 changes: 1 addition & 1 deletion docs/Asterisk_20_Documentation/Upgrading.md
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Expand Up @@ -13,7 +13,7 @@
in the amd.conf configuration file.

### app_bridgewait
* Adds the n option to not answer the channel when
* Adds the n option so as not to answer the channel when
the BridgeWait application is called.

### features
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30 changes: 15 additions & 15 deletions ...on/Applications/Conferencing-Applications/ConfBridge/ConfBridge-CLI-Commands.md
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Expand Up @@ -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.:
Expand All @@ -19,7 +19,7 @@ Kicking SIP/mypeer-00000000 from confbridge 1111

```

On This Pageconfbridge list
`confbridge list`
---------------

Shows a summary listing of all bridges, e.g.:
Expand All @@ -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.:
Expand All @@ -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.:
Expand All @@ -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.:
Expand All @@ -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.:
Expand All @@ -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.:
Expand All @@ -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:
Expand All @@ -101,7 +101,7 @@ Recording started

```

confbridge record stop <confererence>
`confbridge record stop <confererence>`
-------------------------------------

Stops recording the specified conference, e.g.:
Expand All @@ -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.:
Expand All @@ -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.:
Expand All @@ -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.:
Expand All @@ -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.:
Expand Down Expand Up @@ -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.:
Expand All @@ -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.:
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8 changes: 4 additions & 4 deletions ...n/Applications/Conferencing-Applications/ConfBridge/ConfBridge-Configuration.md
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Expand Up @@ -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 | |
Expand All @@ -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. | |
Expand Down Expand Up @@ -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. | |
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2 changes: 1 addition & 1 deletion docs/Configuration/Applications/External-IVR-Interface.md
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Expand Up @@ -92,7 +92,7 @@ The child process can send one of the following commands:
* `P,TIMESTAMP`
* `T,TIMESTAMP`

The `S` command checks to see if there is a playable audio file with the specified name, and if so, clears the generator's playlist and places the file onto the list. Note that the playability check does not take into account transcoding requirements, so it is possible for the file to not be played even though it was found. If the file does not exist it sends a `Z` response with the data element set to the file requested. If the generator is not currently playing silence, then `T` and `D` events will be sent to signal the playlist interruption and notify it of the files that will not be played.
The `S` command checks to see if there is a playable audio file with the specified name, and if so, clears the generator's playlist and places the file onto the list. Note that the playability check does not take into account transcoding requirements, so it is possible for the file not to be played even though it was found. If the file does not exist it sends a `Z` response with the data element set to the file requested. If the generator is not currently playing silence, then `T` and `D` events will be sent to signal the playlist interruption and notify it of the files that will not be played.

The `A` command checks to see if there is a playable audio file with the specified name, and if so, appends it to the generator's playlist. The same playability and exception rules apply as for the `S` command.

Expand Down
6 changes: 3 additions & 3 deletions docs/Configuration/Applications/SMS.md
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Expand Up @@ -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.

Expand Down
6 changes: 3 additions & 3 deletions docs/Configuration/Applications/Voicemail/Message-Waiting-Indication.md
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Expand Up @@ -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
--------------------------------------------------
Expand All @@ -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
------------------
Expand All @@ -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.
2 changes: 1 addition & 1 deletion docs/Configuration/Channel-Drivers/Local-Channel/Local-Channel-Modifiers.md
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Expand Up @@ -25,7 +25,7 @@ Local/101@mycontext/nj

## List of Modifiers

* 'n' - Instructs the Local channel to not do a native transfer (the "n" stands for *No release*) upon the remote end answering the line. This is an esoteric, but important feature if you expect the Local channel to handle calls exactly like a normal channel. If you do not have the "no release" feature set, then as soon as the destination (inside of the Local channel) answers the line and one audio frame passes, the variables and dial plan will revert back to that of the original call, and the Local channel will become a zombie and be removed from the active channels list. This is desirable in some circumstances, but can result in unexpected dialplan behavior if you are doing fancy things with variables in your call handling. Read about [Local Channel Optimization](../Local-Channel-Optimization) to better understand when this option is necessary.
* 'n' - Instructs the Local channel not to do a native transfer (the "n" stands for *No release*) upon the remote end answering the line. This is an esoteric, but important feature if you expect the Local channel to handle calls exactly like a normal channel. If you do not have the "no release" feature set, then as soon as the destination (inside of the Local channel) answers the line and one audio frame passes, the variables and dial plan will revert back to that of the original call, and the Local channel will become a zombie and be removed from the active channels list. This is desirable in some circumstances, but can result in unexpected dialplan behavior if you are doing fancy things with variables in your call handling. Read about [Local Channel Optimization](../Local-Channel-Optimization) to better understand when this option is necessary.
* 'j' - Allows you to use the generic jitterbuffer on incoming calls going to Asterisk applications. For example, this would allow you to use a jitterbuffer for an incoming SIP call to Voicemail by putting a Local channel in the middle. The 'j' option must be used in conjunction with the 'n' option to make sure that the Local channel does not get optimized out of the call.
* 'm' - Will cause the Local channel to forward music on hold (MoH) start and stop requests. Normally the Local channel acts on them and it is started or stopped on the Local channel itself. This options allows those requests to be forwarded through the Local channel.
* 'b' - This option causes the Local channel to return the actual channel that is behind it when queried. This is useful for transfer scenarios as the actual channel will be transferred, not the Local channel.
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