asterisk · gtjoseph · Jun 29, 2026 · Jun 2, 2026
diff --git a/docs/About-the-Project/Supported-Platforms.md b/docs/About-the-Project/Supported-Platforms.md
@@ -5,7 +5,9 @@ pageid: 42566067
 
 The Asterisk software can be installed on a wide range of platforms including various Linux distributions. As a project, however, we are only able to test and support a subset of them. The Asterisk project supports 32-bit and 64-bit x86 platforms using non-end of life CentOS, RHEL, Fedora, Ubuntu, and Debian Linux distributions. Support for other platforms and Linux distributions is best effort and is provided by the community. Any changes to allow such platforms must not hinder or break the project supported Linux distributions.
 
-!!! warning 
-    Note that due to changes and improvements in compilers it is possible for Linux distribution upgrades to result in old versions of Asterisk no longer building. If this occurs it is recommended to upgrade to the latest supported version of Asterisk.
-
-[//]: # (end-warning)
+/// warning
+Note that due to changes and improvements in compilers it is possible
+for Linux distribution upgrades to result in old versions of Asterisk
+no longer building. If this occurs it is recommended to upgrade to the
+latest supported version of Asterisk.
+///
diff --git a/docs/Asterisk-Community/Asterisk-Issue-Guidelines.md b/docs/Asterisk-Community/Asterisk-Issue-Guidelines.md
@@ -24,10 +24,14 @@ See the [How to request a feature section](#how-to-request-a-feature-or-improvem
 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).
 * and...
 
-!!! warning 
-    Security vulnerability issues must NEVER be reported as regular bugs in the issue tracker. Instead they must be reported at [Security Vulnerabilities](https://github.com/asterisk/asterisk/security/advisories/new). You can reach this page by navigating to <https://github.com/asterisk/asterisk> and clicking the "Security" tab at the top of the page.
-
-[//]: # (end-warning)
+/// danger
+Security vulnerability issues must **never** be reported as regular bugs
+in the issue tracker. Instead they must be reported as
+[Security Vulnerabilities](https://github.com/asterisk/asterisk/security/advisories/new).
+You can reach this page by navigating to
+[the main Asterisk repository on GitHub](https://github.com/asterisk/asterisk)
+and clicking the "Security and quality" tab at the top of the page.
+///
 
 #### Why should you read this?
 
@@ -36,11 +40,7 @@ The steps here will help you provide all the information the Asterisk team needs
 Bug Reporting Check List
 ------------------------
 
-!!! warning
-    Before filing a bug report...
-    Your issue may not be a bug or could have been fixed already. Run through the check list below to verify you have done your due diligence.
-
-[//]: # (end-warning)
+Your issue may not be a bug or could have been fixed already. Run through the check list below to verify you have done your due diligence.
 
 * **Are** **you reporting a suspected security vulnerability?**
 * **Are you are on a supported version of Asterisk?**
@@ -49,7 +49,7 @@ Bug Reporting Check List
 * **Have you asked for help in the community? (mailing lists, IRC, forums)**  
 You can locate all these services here: <http://www.asterisk.org/community>
 * **Have you searched the Asterisk documentation in case this behavior is expected?**  
-Search the [Asterisk wiki](//) for the problem or messages you are experiencing.
+Search [the documentation](/) for the problem or messages you are experiencing.
 * **Have you searched the Asterisk bug tracker to see if an issue is already filed for this potential bug?**  
 Search the [Asterisk Issue Tracker on GitHub](https://github.com/asterisk/asterisk/issues) for the issue you are seeing. You can search for issues by selecting **Issues -> Search for Issues** in the top menu bar.
 * **Can you reproduce the problem?**  

diff --git a/docs/Configuration/Applications/Asterisk-Queues/Building-Queues.md b/docs/Configuration/Applications/Asterisk-Queues/Building-Queues.md
@@ -9,10 +9,10 @@ Building Queues
 Written by: Leif Madsen  
  Initial version: 2010-01-14
 
-!!! note 
-    Note that this documentation is based on Asterisk 1.6.2, and this is just one approach to creating queues and the dialplan logic. You may create a better way, and in that case, I would encourage you to submit it to the Asterisk issue tracker at <https://github.com/asterisk/asterisk/issues> for inclusion in Asterisk.
-
-[//]: # (end-note)
+/// note
+Note that this documentation is based on Asterisk 1.6.2, and this is
+just *one* approach to creating queues and the dialplan logic.
+///
 
 In this article, we'll look at setting up a pair of queues in Asterisk called 'sales' and 'support'. These queues can be logged into by queue members, and those members will also have the ability to pause and unpause themselves.
 

diff --git a/docs/Configuration/Applications/Voicemail/Message-Waiting-Indication.md b/docs/Configuration/Applications/Voicemail/Message-Waiting-Indication.md
@@ -37,10 +37,12 @@ Depending on your Asterisk version and configuration, there are a few different
 2. **res_external_mwi**: A module providing an API for other systems to communicate MWI state to Asterisk
 3. **chan_pjsip**: Setting `incoming_mwi_mailbox` on an endpoint
 
-!!! note **  **res_pjsip
-    : The functionality for outbound SIP subscription is not available in res_pjsip yet. Internal infrastructure is built that would allow it, so if this is something you want to work on, please contact the [Asterisk development community](http://www.asterisk.org/community/discuss).
-
-[//]: # (end-note)
+/// note | Outbound SIP subscriptions in res_pjsip
+The functionality for outbound SIP subscription is not available in
+res_pjsip yet. Internal infrastructure is built that would allow it,
+so if this is something you want to work on, please contact the
+[Asterisk development community](http://www.asterisk.org/community/discuss).
+///
 
 Outbound MWI subscription with chan_sip
 ----------------------------------------
@@ -81,10 +83,10 @@ External sources can use the API provided by res_external_mwi to communicate MWI
 
 [Asterisk 12 Configuration_res_mwi_external](/Latest_API/API_Documentation/Module_Configuration/res_mwi_external)
 
-!!! warning 
-    res_external_mwi.so is mutually exclusive with app_voicemail.so. You'll have to load only the one you want to use.
-
-[//]: # (end-warning)
+/// warning
+`res_external_mwi.so` is mutually exclusive with
+`app_voicemail.so`. You'll have to load only the one you want to use.
+///
 
 chan_pjsip
 -----------

diff --git a/docs/Configuration/Applications/Voicemail/index.md b/docs/Configuration/Applications/Voicemail/index.md
@@ -17,7 +17,8 @@ The **[VoiceMailMain()](/Latest_API/API_Documentation/Dialplan_Applications/Voic
 1. **Mailbox** - This parameter specifies the mailbox to log into. It should be a mailbox number and a voice mail context, concatenated with an at-sign (@), like 6001@default. If the voice mail context is omitted, it will default to the default voice mail context. If the mailbox number is omitted, the system will prompt the caller for the mailbox number.
 2. **Options** - One or more options for controlling the voicemail system. The most popular option is the s option, which skips asking for the PIN number
 
-!!! warning Direct Access to Voicemail
-    Please exercise extreme caution when using the s option! With this option set, anyone which has access to this extension can retrieve voicemail messages without entering the mailbox passcode.
-
-[//]: # (end-warning)
+/// warning | Direct Access to Voicemail
+Please exercise extreme caution when using the `s` option! With this
+option set, anyone which has access to this extension can retrieve
+voicemail messages without entering the mailbox passcode.
+///
diff --git a/docs/Configuration/Channel-Drivers/.pages b/docs/Configuration/Channel-Drivers/.pages
@@ -2,7 +2,7 @@ nav:
   - index.md
   - SIP
   - DAHDI.md
-  - Inter-Asterisk-eXchange-protocol-version-2-IAX2
+  - IAX2: Inter-Asterisk-eXchange-protocol-version-2-IAX2
   - Local-Channel
   - AudioSocket.md
   - WebSocket.md

diff --git a/docs/Configuration/Channel-Drivers/DAHDI.md b/docs/Configuration/Channel-Drivers/DAHDI.md
@@ -3,22 +3,15 @@ title: DAHDI
 pageid: 28314858
 ---
 
-!!! warning 
-    Under Construction
-
-[//]: # (end-warning)
-
-!!! note **  Top-level page for DAHDI **channel driver
-    information
-
-    What are the software and hardware requirements for chan_dahdi?
-
-    How do I configure it?
-
-    What are some examples of Analog and PRI configurations?
-
-    Links to use:
-
-[DAHDI](/Configuration/Channel-Drivers/DAHDI)
-
-[//]: # (end-note)
+/// warning | Under Construction
+This is intended as the top-level page for DAHDI channel driver
+information but it has not been written yet. Some ideas for the topics
+that may be useful here are:
+
+- What are the software and hardware requirements for chan_dahdi?
+- How do I configure it?
+- What are some examples of Analog and PRI configurations?
+
+If you would like to help write this section, please consider
+[contributing to the documentation](/Contributing-to-the-Documentation/).
+///
diff --git a/docs/Configuration/Channel-Drivers/IP-Quality-of-Service.md b/docs/Configuration/Channel-Drivers/IP-Quality-of-Service.md
@@ -16,17 +16,17 @@ vconfig set_egress_map [vlan-device] [skb-priority] [vlan-qos]
 
 The table below shows all VoIP channel drivers and other Asterisk modules that support QoS settings for network traffic. It also shows the type(s) of traffic for which each module can support setting QoS settings:
 
-|   | Signaling | Audio | Video | Text |
-| --- | --- | --- | --- | --- |
-| chan_sip | + | + | + | + |
-| chan_skinny | + | + | + |   |
-| chan_mgcp | + | + |   |   |
-| chan_unistm | + | + |   |   |
-| chan_h323 |   | + |   |   |
-| chan_iax2 | + |   |   |   |
-| chan_pjsip | + | + | + |  |
-| DUNDI | + (tos setting) |  |  |  |
-| IAXProv | + (tos setting) |  |  |  |
+|             | Signaling       | Audio | Video | Text |
+|-------------|:---------------:|:-----:|:-----:|:----:|
+| chan_sip    | ✔️               | ✔️     | ✔️     | ✔️    |
+| chan_skinny | ✔️               | ✔️     | ✔️     |      |
+| chan_mgcp   | ✔️               | ✔️     |       |      |
+| chan_unistm | ✔️               | ✔️     |       |      |
+| chan_h323   |                 | ✔️     |       |      |
+| chan_iax2   | ✔️               |       |       |      |
+| chan_pjsip  | ✔️               | ✔️     | ✔️     |      |
+| DUNDI       | ✔️ (tos setting) |       |       |      |
+| IAXProv     | ✔️ (tos setting) |       |       |      |
 
 ### IP TOS values
 
@@ -96,10 +96,10 @@ In chan_pjsip, there are three parameters that control the TOS settings: a **tos
 
 Similarly, there are there parameters that control the 802.1p CoS settings: a **cos** option for a **type=transport** that controls the 802.1p value for SIP signaling packets, a**cos_audio** option for a **type=endpoint** that controls the 802.1p value of RTP audio packets, and a **cos_video** option for a **type=endpoint** that controls the 802.1p value for video packets.
 
-!!! tip **  Changes to a chan_pjsip **type=transport
-    require an Asterisk restart to be affected. They are not affected by simply reloading Asterisk.
-
-[//]: # (end-tip)
+/// tip
+Changes to a chan_pjsip `type=transport` require an Asterisk restart
+to take effect. They are not affected by simply reloading Asterisk.
+///
 
 ### Other RTP channels
 

diff --git a/...ge-protocol-version-2-IAX2/IAX2-Configuration/Configuring-chan_iax2-for-IPv6.md b/...ge-protocol-version-2-IAX2/IAX2-Configuration/Configuring-chan_iax2-for-IPv6.md
@@ -26,7 +26,8 @@ bindaddr=[2001:db8::1]:4569
 
 You can specify 'bindaddr' more than once to bind to multiple addresses, but the first will be the default. IPv6 addresses are accepted.
 
-!!! tip 
-    For details IAX configuration examples see the iax.conf.sample file that comes with the source.
-
-[//]: # (end-tip)
+/// tip
+For detailed IAX configuration examples see
+[the iax.conf.sample file](https://github.com/asterisk/asterisk/blob/master/configs/samples/iax.conf.sample)
+that comes with the source.
+///
diff --git a/docs/Configuration/Channel-Drivers/Motif/Calling-using-Google.md b/docs/Configuration/Channel-Drivers/Motif/Calling-using-Google.md
@@ -149,46 +149,59 @@ exten => s,1,NoOp()
  same => n,Dial(SIP/malcolm,20)
 ```
 
-!!! note 
-    Did you know that the Google Chat client does this same thing; it waits, and then sends a DTMF 1. Really.
-
-[//]: # (end-note)
+/// note | Did you know?
+Did you know that the Google Chat client does this same thing; it
+waits, and then sends a DTMF 1. Really.
+///
 
 This example uses the "s" unmatched extension, because we're only configuring one client connection in this example.
 
-In this example, we're Waiting 1 second, answering the call, sending the DTMF "1" back to Google, and **then** dialing the call.  
+In this example, we're waiting 1 second, answering the call, sending the DTMF "1" back to Google, and **then** dialing the call.  
 
-!!! tip Using Google's voicemail** Another method for accomplishing the sending of the DTMF event is to use Dial option "D." The D option tells Asterisk to send a specified DTMF string after the called party has answered. DTMF events specified before a colon are sent to the **called** party. DTMF events specified after a colon are sent to the **calling
-    party.
+/// tip | Using Google's voicemail
+Another method for accomplishing the sending of the DTMF event is to
+use Dial option "D." The D option tells Asterisk to send a specified
+DTMF string after the called party has answered. DTMF events specified
+before a colon are sent to the **called** party. DTMF events specified
+after a colon are sent to the **calling** party.
 
-    In this example then, one does not need to actually answer the call first, though one should still wait at least a second for things, like STUN setup, to finish. This means that if the called party doesn't answer, Google will resort to sending the call to one's Google Voice voicemail box, instead of leaving it at Asterisk.
-[//]: # (end-tip)
+In this example then, one does not need to actually answer the call
+first, though one should still wait at least a second for things, like
+STUN setup, to finish. This means that if the called party doesn't
+answer, Google will resort to sending the call to one's Google Voice
+voicemail box, instead of leaving it at Asterisk.
 
 ```
 exten => s,1,Dial(SIP/malcolm,20,D(:1))
-
----
 ```
+///
 
-!!! tip Filtering Caller ID
-    The inbound CallerID from Google is going to look a bit nasty, e.g.:
-[//]: # (end-tip)
+/// tip | Filtering Caller ID
+The inbound CallerID from Google is going to look a bit nasty, e.g.:
 
 ```
 +15555551212@voice.google.com/srvres-MTAuMjE4LjIuMTk3Ojk4MzM=
+```
 
----
-
-Your VoIP client (SIPDroid) might not like this, so let's simplify that Caller ID a bit, and make it more presentable for your phone's display. Here's the example that we'll step through:
+Your VoIP client (SIPDroid) might not like this, so let's simplify
+that Caller ID a bit, and make it more presentable for your phone's
+display. Here's the example that we'll step through:
 
+```
 exten => s,1,NoOp()
  same => n,Set(crazygooglecid=${CALLERID(name)})
  same => n,Set(stripcrazysuffix=${CUT(crazygooglecid,@,1)})
  same => n,Set(CALLERID(all)=${stripcrazysuffix})
  same => n,Dial(SIP/malcolm,20,D(:1))
 ```
 
-First, we set a variable called **crazygooglecid** to be equal to the name field of the CALLERID function. Next, we use the CUT function to grab everything that's before the @ symbol, and save it in a new variable called **stripcrazysuffix.** We'll set this new variable to the CALLERID that we're going to use for our Dial. Finally, we'll actually Dial our internal destination.
+First, we set a variable called `crazygooglecid` to be equal to the
+name field of the CALLERID function. Next, we use the CUT function to
+grab everything that's before the @ symbol, and save it in a new
+variable called `stripcrazysuffix`. We'll set this new variable to
+the CALLERID that we're going to use for our Dial. Finally, we'll
+actually Dial our internal destination.
+///
 
 ### Outgoing calls
 

diff --git a/docs/Configuration/Channel-Drivers/Motif/index.md b/docs/Configuration/Channel-Drivers/Motif/index.md
@@ -3,12 +3,12 @@ title: Overview
 pageid: 27200344
 ---
 
-!!! warning 
-    Under Construction
-
-[//]: # (end-warning)
-
-!!! note 
-    Page for information on the Motif channel driver, describing configuration, pointing to any resources and a top-level page for any examples or tutorials such as calling with Google Voice.
-
-[//]: # (end-note)
+/// warning | Under Construction
+This page is intended to include information on the Motif channel
+driver, such as describing configuration, pointing to any resources
+and a top-level page for any examples or tutorials such as calling
+with Google Voice.
+
+If you would like to help write this section, please consider
+[contributing to the documentation](/Contributing-to-the-Documentation/).
+///
diff --git a/docs/Configuration/Channel-Drivers/SIP/.pages b/docs/Configuration/Channel-Drivers/SIP/.pages
@@ -0,0 +1,4 @@
+nav:
+  - Concepts
+  - Configuring res_pjsip: Configuring-res_pjsip
+  - Configuring chan_sip: Configuring-chan_sip
diff --git a/...Channel-Drivers/SIP/Concepts/SIP-Direct-Media-Reinvite-Glare-Avoidance/index.md b/...Channel-Drivers/SIP/Concepts/SIP-Direct-Media-Reinvite-Glare-Avoidance/index.md
@@ -1,9 +1,9 @@
 # SIP Direct Media Reinvite Glare Avoidance
 
-!!! note
-    While this page has not yet been updated for chan_pjsip, the
-    concepts are the same.
-[//]: # (end-note)
+/// note | What about chan_pjsip?
+While this page has not yet been updated for chan_pjsip, the concepts
+are the same.
+///
 
 ## Overview
 

diff --git a/...ation/Channel-Drivers/SIP/Configuring-chan_sip/Configuring-chan_sip-for-IPv6.md b/...ation/Channel-Drivers/SIP/Configuring-chan_sip/Configuring-chan_sip-for-IPv6.md
@@ -32,12 +32,14 @@ You can specify a port number by wrapping the address in square brackets and usi
 bindaddr=[::]:5062
 ```
 
-!!! tip 
-    You can choose independently for UDP, TCP, and TLS, by specifying different values for  "udpbindaddr", "tcpbindaddr", and "tlsbindaddr".
-
-    Note that using bindaddr=:: will show only a single IPv6 socket in netstat. IPv4 is supported at the same time using IPv4-mapped IPv6 addresses.)
-
-[//]: # (end-tip)
+/// tip
+You can choose independently for UDP, TCP, and TLS, by specifying
+different values for `udpbindaddr`, `tcpbindaddr`, and `tlsbindaddr`.
+
+Note that using `bindaddr=::` will show only a single IPv6 socket in
+netstat. IPv4 is supported at the same time using IPv4-mapped IPv6
+addresses.
+///
 
 Other Options
 =============

diff --git a/docs/Configuration/Channel-Drivers/SIP/Configuring-chan_sip/index.md b/docs/Configuration/Channel-Drivers/SIP/Configuring-chan_sip/index.md
@@ -6,27 +6,25 @@ pageid: 28934283
 Note about chan_pjsip
 ======================
 
-!!! warning PJSIP is the Standard SIP Driver
-    It is not recommended for new installations to use chan_sip.
+/// warning | PJSIP is the Standard SIP Driver
+It is not recommended for new installations to use chan_sip.
 
-    * chan_sip has been officially removed in Asterisk 21 – Releasing 2023
-    * chan_sip was deprecated in Asterisk 17 – Released: October 2019
-    * Beginning with Asterisk 13.8.0, a stable version of pjproject is included in Asterisk's ./third-party directory and is enabled with the `--with-pjproject-bundled` option to `./configure`.
-    * Beginning with Asterisk 15.0.0, it is enabled by default but can be disabled with the `--without-pjproject-bundled` option to `./configure`.
+* chan_sip has been officially removed in Asterisk 21 – Releasing 2023
+* chan_sip was deprecated in Asterisk 17 – Released: October 2019
+* Beginning with Asterisk 13.8.0, a stable version of pjproject is included in Asterisk's ./third-party directory and is enabled with the `--with-pjproject-bundled` option to `./configure`.
+* Beginning with Asterisk 15.0.0, it is enabled by default but can be disabled with the `--without-pjproject-bundled` option to `./configure`.
 
-    See: [PJSIP-pjproject](/Getting-Started/Installing-Asterisk/Installing-Asterisk-From-Source/Prerequisites/PJSIP-pjproject)
+See: [PJSIP-pjproject](/Getting-Started/Installing-Asterisk/Installing-Asterisk-From-Source/Prerequisites/PJSIP-pjproject)
 
-    See:  [Configuring res_pjsip](/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip)
+See: [Configuring res_pjsip](/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip)
 
-    See: [Migrating from chan_sip to res_pjsip](/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/Migrating-from-chan_sip-to-res_pjsip)
-
-[//]: # (end-warning)
+See: [Migrating from chan_sip to res_pjsip](/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/Migrating-from-chan_sip-to-res_pjsip)
+///
 
 Configuring chan_sip
 =====================
 
 There is documentation that resides in the **sip.conf.sample** file included with the source.
 
-\* Please be advised that limited support will be available on the mailing list, IRC, and bug tracker for issues with chan_sip
-
+* Please be advised that limited support will be available on the mailing list, IRC, and bug tracker for issues with chan_sip
 * Further development and bug fixes for chan_sip are not likely