The instruction set for "Installing Asterisk From Source" list is ordered alphabetically, instead of chronologically. It is very difficult to follow along with the installation process directly from the documentation, and seems it is likely related to the same issue affecting #24.

https://docs.asterisk.org/Asterisk_21_Documentation/API_Documentation/Asterisk_REST_Interface/Asterisk_REST_Data_Models/#message

Issue: The subtypes of the "Message" model should only be "MissingParams", "Event". The paragraph subtype section here contains all the Event subtypes.

In 20.7.0's stir_shaken.conf.sample, you can read:

-- endpoint_behhavior

Should be:

-- endpoint_behavior

Possibly a wider issue, but here's an example of a page that is completely malformatted (scroll down to "Syntax"): https://docs.asterisk.org/Asterisk_20_Documentation/API_Documentation/Dialplan_Applications/Dial/

Some pages are missing a main heading, such as in the screenshot below (Fundamentals > Types of Asterisk Modules). So a user has to check the hamburger menu, address bar, or other means to remember what page they're currently on.

TODO: Update issue with links to all pages with missing main headings

There seems to be broken toggle functionality for content that is meant to be hidden by default. For example, in Fundamentals > Types of Asterisk Modules.

Possibly related to issue #79 .

Expected behaviour: Clicking on a high-level topic or a label used to organize related subtopics should open an overview page with some content (e.g, Fundamentals > Asterisk Architecture) or load the first sub-item (e.g, Getting Started > Beginning Asterisk).

Actual behaviour: There are several empty overview pages.

Based on pages like the one below, the docs are being developed. I'd still like to suggest including links to the sub-topics or sub-items on the overview page (in addition to the under construction warning) pending when the overview content is created.

TODO: Update issue with links to all empty overview pages.

Admonitions are the call-outs that contain valuable information, for example, Note or Warning.
A lot of these are broken in the documentation.

I suspect it might be because of a missing newline, but I'm not sure.

Here is an example from the official docs:

If you go to docs.asterisk.org and the Asterisk REST Interface (at least in both 18 and 20 versions). Go to the Channels.
https://docs.asterisk.org/Asterisk_20_Documentation/API_Documentation/Asterisk_REST_Interface/Channels_REST_API/

There is a list of Method and path links.
Most work, but a few do not. Not sure if the problem is the link should use all lower case or if the section should be a blend up upper/lower case.

Originate with ID attempts to jump to originateWithId (should be originatewithid)
Continue Dialplan attempts to jump to continueInDialplan (the link that works is continueindialplan)
Play with playback id attempts to jump to playWithId (should be playwithid)
The snoop attempts to jump to the snoopChannel (should be snoopchannel)
Snoop with channel id attempts to jump to snoopChannelWithId (should be snoopchannelwithid)
ExternalMedia attempts to go to externalMedia (link that works is externalmedia)

In Outgoing Call Flow section, point 7 from [1], you can read, that “If there’s no “tn” object matching the caller-id, skip attestation and continue the call.”
Can this matching be further clarified or detailed using an example, such as bellow ?

"Example:
For instance, if CALLERID(num) is valued to 123456789012 and “stir_shaken show tns” displays a line “TN: 123456789012” , then there is match.

Alternatively, if CALLERID(num) is valued to +123456789012 (+E164 notation) and “stir_shaken show tns” still displays a line “TN: 123456789012” , then there is NO match."

[1] STIR-SHAKEN - Asterisk Documentation

E.g. in the table [https://docs.asterisk.org/Asterisk_21_Documentation/API_Documentation/Module_Configuration/res_pjsip/?h=res_pjsip#endpoint]

The “Type” value is shown in the “Default Value” column, the “Default Value” value is in the “Description” column. Looks like when processing the data it seems to skip every second column.

Hello,

It's complicated to understand the body of a POST for the REST API. Here an example with externalMedia.

After clicking through documentation using the left-side menu, the page reloads with correct information on the right-side, but it seems that the left-side menu stays stuck at the top and does not scroll down automatically to the selected menu item in some small percent of cases, such as when clicking on the 91st item in the sub-menu.

Steps to reproduce (on Firefox 102.13.0 ESR from July 2023):

1. Click on Originate AMI Action in Asterisk Version 20 documentation: https://docs.asterisk.org/Asterisk_20_Documentation/API_Documentation/AMI_Actions/Originate/
2. Click on PJSIPNotify in the left-side menu. No problem, it works, page loads and menu scrolls. Note this is the 90th menu item.
3. Click on PJSIPQualify in the left-side menu. Look, it loads the page, but the menu does not scroll. This is the 91st menu item in the AMI Action section.
4. Repeat click on PJSIPRegister, the 92nd item, and it is the same problem as PJSIPQualify.
5. The problem continues for many of the items after that, but it does work correctly for a few eg. Park, QueueAdd, etc. Seemingly at random which items work and which don't, but it is consistent on the items. Which makes you go grrr.

https://docs.asterisk.org/Configuration/Codec-Opus/

The list of options breaks just after the "max_playback_rate" section due to attempting to embed a MD table in another MD table, which makes the remainder of the document very hard to use. Instead, this could be accomplished with an inline html table, making the remainder of the page readable.

For example:

Like mentioned in #18 , There are too many dead links out in the web.

Most of the links to the old documentation in community.asterisk.org and stackoverflow.

In case there is no easy fix - I would like to suggest to try handling 404 a bit better than current.

Instead of redirecting to homepage, you should:

1. Display a not found message, maybe with suggestion to perform a search
2. Log and count 404 requests per URL at server side, so someone could manually add redirects. At least for the most common links.

Severity

Minor

Versions

20.7.0

Components/Modules

app_queue

Operating Environment

Linux, OpenSuse, Leap 15.5

Frequency of Occurrence

Constant

Issue Description

When passing option "b" on calling Queue() it is necessary to always specify a "full qualified" handler by context, extension, priority even if the handler is located within the same context. Otherwise the call fails. This is in contrast to option "b" on calling Dial() where context can be omitted if within the same context.

Relevant log output

No response

Asterisk Issue Guidelines

• Yes, I have read the Asterisk Issue Guidelines

If you find a link to the old wiki.asterisk.org website that just takes you to the new documentation site homepage instead of real content, add it here and we'll try to set up a redirect that takes you to the relevant content instead.

Please include the stale link and, if you found the real content on the new website, that link as well.

Hithere

There is too many dead links:

1. All links that reference asterisk 12 \ 13 docs, for example: https://docs.asterisk.org/Asterisk-12-Application_MessageSend
2. All links that still reference to the old docs site

You can simply scan docs.asterisk.org with a dead link scanner, you will find a lot

It feels a bit awkward to constantly click the hamburger icon to navigate the doc site on a PC. I'd like to suggest adding a sidebar table of contents (TOC) for large screens and only introducing the hamburger menu on smaller screens. Having a prominent TOC on large screens should also help (re-)orient a user regarding their current place in the docs without needing to click the icon.

Overview of the Dial Application says one should look at Asterisk 13 Application_Dial as the example of how to find the detailed documentation, but Asterisk 13 has been expunged from the documentation site, so one gets a 404 page.

There is no links to full set of Actions and Events

https://docs.asterisk.org/Configuration/Interfaces/Asterisk-Manager-Interface-AMI/AMI-v2-Specification/#actions_1

Different issues on that page:

• In the text that follows after "unless you purposely want to fall through to a less specific match." the text that should be monospace is not and vice versa.
• In a note there is an incomplete sentence: "The only characters with special meaning within a set are the '-' character, to define a range between two characters, the '' character to escape a special character available within a set, and"

The following pages have broken links following the removal of CHANGES files in Asterisk branches:

docs/Asterisk_18_Documentation/WhatsNew.md
docs/Asterisk_20_Documentation/WhatsNew.md
docs/Asterisk_21_Documentation/WhatsNew.md
docs/About-the-Project/A-Brief-History-of-the-Asterisk-Project.md
docs/Certified-Asterisk_18.9_Documentation/WhatsNew.md

(oddly enough, the Asterisk 19 link is fine)
Original issue that lead to the removal: asterisk/asterisk#360

https://docs.asterisk.org/Asterisk_21_Documentation/API_Documentation/Dialplan_Functions/CHANNEL/

The note starting at "If not specified, 'audio' is used by default." is never closed.

Looking at the underlying XML documentation in channels/pjsip/dialplan_functions_doc.xml, it appears to be correct (the note is properly closed):

<enumlist>
	<enum name="audio">
		<para>Retrieve information from the audio media stream.</para>
		<note><para>If not specified, <literal>audio</literal> is used
		by default.</para></note>
	</enum>
	<enum name="video">
		<para>Retrieve information from the video media stream.</para>
	</enum>
</enumlist>

Hello Everyone,

Is there asterisk RealTime mySQL/PostgreSQL Complete Schema file available? kindly share a URL to download it.

thanks

This pjproject 2.13.1 is the link is down or not getting downloaded and hence the instllation of asterisk is not working

https://raw.githubusercontent.com/asterisk/third-party/master/pjproject/2.13.1/pjproject-2.13.1.tar.bz2, below is the log where the installation is getting stuck

configure: checking OPENSSL with pkg-config
configure: checking whether system openssl > 1.1.0
checking for OPENSSL... yes
checking for embedded pjproject (may have to download)... configuring
[pjproject] Downloading https://raw.githubusercontent.com/asterisk/third-party/master/pjproject/2.13.1/pjproject-2.13.1.tar.bz2 to /tmp/pjproject-2.13.1.tar.bz2

tried other methods by download offline and giving path, we tried
$ mkdir /tmp/downloads
$ wget -O /tmp/downloads/pjproject-2.6.tar.bz2 http://www.pjsip.org/release/2.6/pjproject-2.6.tar.bz2
$ wget -O /tmp/downloads/pjproject-2.6.md5 http://www.pjsip.org/release/2.6/MD5SUM.txt

we also tried below
Run ./configure with the --with-externals-cache=/tmp/downloads option. ./configure will check there first and only download if the files aren't already there or the tarball checksum doesn't match what's in the md5 file. This is similar to the --with-sounds-cache option. BTW, the --with-externals-cache mechanism works for the precompiled codecs and the Digium Phone Module for Asterisk as well. As of Asterisk 13.18, 14.7 and 15.0, the --with-download-cache option can be used to specify both the externals and sounds cache directory.
Set the PJPROJECT_URL environment variable to any valid URL (including file:// URLs) where ./configure can find the tarball and checksum files. The variable can be set in your environment and exported or specified directly on the ./configure command line. As of Asterisk 13.18, 14.7 and 15.0, the AST_DOWNLOAD_CACHE environment variable can be used to specify both the externals and sounds cache directory.

but nothing seems to be working and stop and download only

See
https://docs.asterisk.org/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/PJSIP-Configuration-Sections-and-Relationships/

esp.
https://docs.asterisk.org/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/PJSIP-Configuration-Sections-and-Relationships/#relationships-of-configuration-objects-in-pjsipconf

The section talks about "a diagram", it has some explanations and even a short overview of Entity Relationship diagram elements, but a concrete diagram is missing.

This one does not show up on the documentation page:
https://github.com/asterisk/documentation/blob/main/docs/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/res_sip_configrelationships.png

Various references to https://docs.asterisk.org/Asterisk-12-CEL-Specification
Old copy at https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CEL+Specification still exists

I'm brand new to Asterisk and looking for a good place to get started. A web search landed me on the page titled The Most Basic PBX .. which sounds perfect, a basic PBX to "help [me] learn the fundamentals of configuring Asterisk" is exactly what I'm after. But that page only covers "Requirements and Assumptions" and it's not clear where to go next from there.

Going up a level in the docs takes me to Basic PBX Functionality which is another introductory page. The pages under that page seem like the right instructions, but they seem out of order (or in alphabetical order by title?). For example, "Adding Voice Mail" is the first subpage listed, but that's a short page, and sounds like something I'd do after setting up other more basic things. "Creating Dialplan Extensions" is next, but it starts with "The last things we need to do ..." so I assume that's not the right place to start either.

Aha! I just found the Hello World page .. maybe this is a better place to start? Would be great to link to that page from more places, and make it clear that's the recommended place to start for complete newbies (assuming that's true).

Anyway, I expect I'll be able to eventually figure things out, but just wanted to share my experience as a new user trying to figure out how to get started with Asterisk.

The qualify timeout includes any DNS lookup(s) required. This should be noted in the documentation.