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<html><head><title>ChangeLog for asterisk-certified-20.7-cert6</title></head><body>
<h2>Change Log for Release asterisk-certified-20.7-cert6</h2>
<h3>Links:</h3>
<ul>
<li><a href="https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-20.7-cert6.html">Full ChangeLog</a> </li>
<li><a href="https://github.com/asterisk/asterisk/compare/certified-20.7-cert5...certified-20.7-cert6">GitHub Diff</a> </li>
<li><a href="https://downloads.asterisk.org/pub/telephony/certified-asterisk/asterisk-certified-20.7-cert6.tar.gz">Tarball</a> </li>
<li><a href="https://downloads.asterisk.org/pub/telephony/certified-asterisk">Downloads</a> </li>
</ul>
<h3>Summary:</h3>
<ul>
<li>Commits: 31</li>
<li>Commit Authors: 5</li>
<li>Issues Resolved: 16</li>
<li>Security Advisories Resolved: 0</li>
</ul>
<h3>User Notes:</h3>
<ul>
<li>
<h4>res_stir_shaken: Allow sending Identity headers for unknown TNs</h4>
<p>You can now set the "unknown_tn_attest_level" option
in the attestation and/or profile objects in stir_shaken.conf to
enable sending Identity headers for callerid TNs not explicitly
configured.</p>
</li>
<li>
<h4>res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"</h4>
<p>The new "suppress_moh_on_sendonly" endpoint option
can be used to prevent playing MOH back to a caller if the remote
end sends "sendonly" or "inactive" (hold) to Asterisk in an SDP.</p>
</li>
<li>
<h4>app_mixmonitor: Add 'D' option for dual-channel audio.</h4>
<p>The MixMonitor application now has a new 'D' option which
interleaves the recorded audio in the output frames. This allows for
stereo recording output with one channel being the transmitted audio and
the other being the received audio. The 't' and 't' options are
compatible with this.</p>
</li>
<li>
<h4>db.c: Remove limit on family/key length</h4>
<p>The <code>ast_db_*()</code> APIs have had the 253 byte limit on
"/family/key" removed and will now accept families and keys with a
total length of up to SQLITE_MAX_LENGTH (currently 1e9!). This
affects the <code>DB*</code> dialplan applications, dialplan functions,
manager actions and <code>databse</code> CLI commands. Since the
media_cache also uses the <code>ast_db_*()</code> APIs, you can now store
resources with URIs longer than 253 bytes.</p>
</li>
</ul>
<h3>Upgrade Notes:</h3>
<h3>Commit Authors:</h3>
<ul>
<li>Ben Ford: (3)</li>
<li>Chrsmj: (1)</li>
<li>George Joseph: (22)</li>
<li>Joshua C. Colp: (1)</li>
<li>Sean Bright: (4)</li>
</ul>
<h2>Issue and Commit Detail:</h2>
<h3>Closed Issues:</h3>
<ul>
<li>879: [bug]: res_stir_shaken/verification.c: Getting verification errors when global_disable=yes</li>
<li>881: [bug]: Long URLs are being rejected by the media cache because of an astdb key length limit</li>
<li>884: [bug]: A ':' at the top of in stir_shaken.conf make Asterisk producing a core file when starting</li>
<li>889: [bug]: res_stir_shaken/verification.c has a stale include for jansson.h that can cause compilation to fail</li>
<li>904: [bug]: stir_shaken: attest_level isn't being propagated correctly from attestation to profile to tn</li>
<li>921: [bug]: Stir-Shaken doesn’t allow B or C attestation for unknown callerid which is allowed by ATIS-1000074.v003, §5.2.4</li>
<li>937: [bug]: Wrong format for sample config file 'geolocation.conf.sample'</li>
<li>938: [bug]: memory leak - CBAnn leaks a small amount format_cap related memory for every confbridge</li>
<li>945: [improvement]: Add stereo recording support for app_mixmonitor</li>
<li>974: [improvement]: change and/or remove some wiki mentions to docs mentions in the sample configs</li>
<li>979: [improvement]: Add ability to suppress MOH when a remote endpoint sends "sendonly" or "inactive"</li>
<li>982: [bug]: The addition of tenantid to the ast_sip_endpoint structure broke ABI compatibility</li>
<li>995: [bug]: suppress_moh_on_sendonly should use AST_BOOL_VALUES instead of YESNO_VALUES in alembic script</li>
<li>1112: [bug]: STIR/SHAKEN verification doesn't allow anonymous callerid to be passed to the dialplan.</li>
<li>1122: Need status on CVE-2024-57520 claim.</li>
<li>1131: [bug]: CHANGES link broken in README.md</li>
</ul>
<h3>Commits By Author:</h3>
<ul>
<li>
<h4>Ben Ford (3):</h4>
</li>
<li>app_mixmonitor: Add 'D' option for dual-channel audio.</li>
<li>Add res_pjsip_config_sangoma external module.</li>
<li>
<p>documentation: Update Gosub, Goto, and add new documentationtype.</p>
</li>
<li>
<h4>George Joseph (22):</h4>
</li>
<li>res_stir_shaken: Check for disabled before param validation</li>
<li>res_stir_shaken.c: Fix crash when stir_shaken.conf is invalid</li>
<li>res_stir_shaken: Remove stale include for jansson.h in verification.c</li>
<li>db.c: Remove limit on family/key length</li>
<li>Fix application references to Background</li>
<li>stir_shaken: Fix propagation of attest_level and a few other values</li>
<li>manager.c: Add unit test for Originate app and appdata permissions</li>
<li>geolocation.sample.conf: Fix comment marker at end of file</li>
<li>core_unreal.c: Fix memory leak in ast_unreal_new_channels()</li>
<li>res_pjsip: Move tenantid to end of ast_sip_endpoint</li>
<li>res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"</li>
<li>res_pjsip: Change suppress_moh_on_sendonly to OPT_BOOL_T</li>
<li>res_stir_shaken: Allow sending Identity headers for unknown TNs</li>
<li>Allow C++ source files (as extension .cc) in the main directory</li>
<li>gcc14: Fix issues caught by gcc 14</li>
<li>manager.c: Split XML docs into separate file</li>
<li>docs: Enable since/version handling for XML, CLI and ARI documentation</li>
<li>README.md, asterisk.c: Update Copyright Dates</li>
<li>res_stir_shaken: Allow missing or anonymous CID to continue to the dialplan.</li>
<li>swagger_model.py: Fix invalid escape sequence in get_list_parameter_type().</li>
<li>manager.c: Check for restricted file in action_createconfig.</li>
<li>
<p>README.md: Updates and Fixes</p>
</li>
<li>
<h4>Joshua C. Colp (1):</h4>
</li>
<li>
<p>LICENSE: Update company name, email, and address.</p>
</li>
<li>
<h4>Sean Bright (4):</h4>
</li>
<li>res_pjsip_pubsub: Persist subscription 'generator_data' in sorcery</li>
<li>alembic: Drop redundant voicemail_messages index.</li>
<li>manager.c: Rename restrictedFile to is_restricted_file.</li>
<li>
<p>manager: Add <code>&lt;since&gt;</code> tags for all AMI actions.</p>
</li>
<li>
<h4>chrsmj (1):</h4>
</li>
<li>samples: remove and/or change some wiki mentions</li>
</ul>
<h3>Commit List:</h3>
<ul>
<li>documentation: Update Gosub, Goto, and add new documentationtype.</li>
<li>README.md: Updates and Fixes</li>
<li>manager.c: Check for restricted file in action_createconfig.</li>
<li>swagger_model.py: Fix invalid escape sequence in get_list_parameter_type().</li>
<li>res_stir_shaken: Allow missing or anonymous CID to continue to the dialplan.</li>
<li>LICENSE: Update company name, email, and address.</li>
<li>README.md, asterisk.c: Update Copyright Dates</li>
<li>docs: Enable since/version handling for XML, CLI and ARI documentation</li>
<li>manager: Add <code>&lt;since&gt;</code> tags for all AMI actions.</li>
<li>manager.c: Split XML docs into separate file</li>
<li>manager.c: Rename restrictedFile to is_restricted_file.</li>
<li>gcc14: Fix issues caught by gcc 14</li>
<li>Allow C++ source files (as extension .cc) in the main directory</li>
<li>res_stir_shaken: Allow sending Identity headers for unknown TNs</li>
<li>res_pjsip: Change suppress_moh_on_sendonly to OPT_BOOL_T</li>
<li>res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"</li>
<li>samples: remove and/or change some wiki mentions</li>
<li>res_pjsip: Move tenantid to end of ast_sip_endpoint</li>
<li>Add res_pjsip_config_sangoma external module.</li>
<li>app_mixmonitor: Add 'D' option for dual-channel audio.</li>
<li>core_unreal.c: Fix memory leak in ast_unreal_new_channels()</li>
<li>geolocation.sample.conf: Fix comment marker at end of file</li>
<li>manager.c: Add unit test for Originate app and appdata permissions</li>
<li>alembic: Drop redundant voicemail_messages index.</li>
<li>res_pjsip_pubsub: Persist subscription 'generator_data' in sorcery</li>
<li>stir_shaken: Fix propagation of attest_level and a few other values</li>
<li>Fix application references to Background</li>
<li>db.c: Remove limit on family/key length</li>
<li>res_stir_shaken: Remove stale include for jansson.h in verification.c</li>
<li>res_stir_shaken.c: Fix crash when stir_shaken.conf is invalid</li>
<li>res_stir_shaken: Check for disabled before param validation</li>
</ul>
<h3>Commit Details:</h3>
<h4>documentation: Update Gosub, Goto, and add new documentationtype.</h4>
<p>Author: Ben Ford
Date: 2025-03-14</p>
<p>Gosub and Goto were not displaying their syntax correctly on the docs
site. This change adds a new way to specify an optional context, an
optional extension, and a required priority that the xml stylesheet can
parse without having to know which optional parameters come in which
order. In Asterisk, it looks like this:</p>
<pre><code>parameter name="context" documentationtype="dialplan_context"
parameter name="extension" documentationtype="dialplan_extension"
parameter name="priority" documentationtype="dialplan_priority" required="true"
</code></pre>
<p>The stylesheet will ignore the context and extension parameters, but for
priority, it will automatically inject the following:</p>
<pre><code>[[context,]extension,]priority
</code></pre>
<p>This is the correct oder for applications such as Gosub and Goto.</p>
<h4>README.md: Updates and Fixes</h4>
<p>Author: George Joseph
Date: 2025-03-05</p>
<ul>
<li>Outdated information has been removed.</li>
<li>New links added.</li>
<li>Placeholder added for link to change logs.</li>
</ul>
<p>Going forward, the release process will create HTML versions of the README
and change log and will update the link in the README to the current
change log for the branch...</p>
<ul>
<li>In the development branches, the link will always point to the current
release on GitHub.</li>
<li>In the "releases/*" branches and the tarballs, the link will point to the
ChangeLogs/ChangeLog-<version>.html file in the source directory.</li>
<li>On the downloads website, the link will point to the
ChangeLog-<version>.html file in the same directory.</li>
</ul>
<p>Resolves: #1131</p>
<h4>manager.c: Check for restricted file in action_createconfig.</h4>
<p>Author: George Joseph
Date: 2025-03-03</p>
<p>The <code>CreateConfig</code> manager action now ensures that a config file can
only be created in the AST_CONFIG_DIR unless <code>live_dangerously</code> is set.</p>
<p>Resolves: #1122</p>
<h4>swagger_model.py: Fix invalid escape sequence in get_list_parameter_type().</h4>
<p>Author: George Joseph
Date: 2025-03-04</p>
<p>Recent python versions complain when backslashes in strings create invalid
escape sequences. This causes issues for strings used as regex patterns like
<code>'^List\[(.*)\]$'</code> where you want the regex parser to treat <code>[</code> and <code>]</code>
as literals. Double-backslashing is one way to fix it but simply converting
the string to a raw string <code>re.match(r'^List\[(.*)\]$', text)</code> is easier
and less error prone.</p>
<h4>res_stir_shaken: Allow missing or anonymous CID to continue to the dialplan.</h4>
<p>Author: George Joseph
Date: 2025-02-05</p>
<p>The verification check for missing or anonymous callerid was happening before
the endpoint's profile was retrieved which meant that the failure_action
parameter wasn't available. Therefore, if verification was enabled and there
was no callerid or it was "anonymous", the call was immediately terminated
instead of giving the dialplan the ability to decide what to do with the call.</p>
<ul>
<li>
<p>The callerid check now happens after the verification context is created and
the endpoint's stir_shaken_profile is available.</p>
</li>
<li>
<p>The check now processes the callerid failure just as it does for other
verification failures and respects the failure_action parameter. If set
to "continue" or "continue_return_reason", <code>STIR_SHAKEN(0,verify_result)</code>
in the dialplan will return "invalid_or_no_callerid".</p>
</li>
<li>
<p>If the endpoint's failure_action is "reject_request", the call will be
rejected with <code>433 "Anonymity Disallowed"</code>.</p>
</li>
<li>
<p>If the endpoint's failure_action is "continue_return_reason", the call will
continue but a <code>Reason: STIR; cause=433; text="Anonymity Disallowed"</code>
header will be added to the next provisional or final response.</p>
</li>
</ul>
<p>Resolves: #1112</p>
<h4>LICENSE: Update company name, email, and address.</h4>
<p>Author: Joshua C. Colp
Date: 2025-01-21</p>
<h4>README.md, asterisk.c: Update Copyright Dates</h4>
<p>Author: George Joseph
Date: 2025-01-20</p>
<h4>docs: Enable since/version handling for XML, CLI and ARI documentation</h4>
<p>Author: George Joseph
Date: 2025-01-09</p>
<ul>
<li>
<p>Added the "since" element to the XML configObject and configOption elements
in appdocsxml.dtd.</p>
</li>
<li>
<p>Added the "Since" section to the following CLI output:
<code>config show help &lt;module&gt; &lt;object&gt;
config show help &lt;module&gt; &lt;object&gt; &lt;option&gt;
core show application &lt;app&gt;
core show function &lt;func&gt;
manager show command &lt;command&gt;
manager show event &lt;event&gt;
agi show commands topic &lt;topic&gt;</code></p>
</li>
<li>
<p>Refactored the commands above to output their sections in the same order:
Synopsis, Since, Description, Syntax, Arguments, SeeAlso</p>
</li>
<li>
<p>Refactored the commands above so they all use the same pattern for writing
the output to the CLI.</p>
</li>
<li>
<p>Fixed several memory leaks caused by failure to free temporary output
buffers.</p>
</li>
<li>
<p>Added a "since" array to the mustache template for the top-level resources
(Channel, Endpoint, etc.) and to the paths/methods underneath them. These
will be added to the generated markdown if present.
Example:
```
"resourcePath": "/api-docs/channels.{format}",
"requiresModules": [
"res_stasis_answer",
"res_stasis_playback",
"res_stasis_recording",
"res_stasis_snoop"
],
"since": [
"18.0.0",
"21.0.0"
],
"apis": [
{
"path": "/channels",
"description": "Active channels",
"operations": [
{
"httpMethod": "GET",
"since": [
"18.6.0",
"21.8.0"
],
"summary": "List all active channels in Asterisk.",
"nickname": "list",
"responseClass": "List[Channel]"
},</p>
<p>```</p>
</li>
</ul>
<p>NOTE: No versioning information is actually added in this commit.
Those will be added separately and instructions for adding and maintaining
them will be published on the documentation site at a later date.</p>
<h4>manager: Add <code>&lt;since&gt;</code> tags for all AMI actions.</h4>
<p>Author: Sean Bright
Date: 2025-01-02</p>
<h4>manager.c: Split XML docs into separate file</h4>
<p>Author: George Joseph
Date: 2025-01-13</p>
<p>To keep the source tree somewhat compatible with the base 20 branch
the XML documentation from manager.c has been extracted into manager_doc.xml.
This will give future cherry-picks a better channce of succeeding without
manual intervention.</p>
<h4>manager.c: Rename restrictedFile to is_restricted_file.</h4>
<p>Author: Sean Bright
Date: 2025-01-09</p>
<p>Also correct the spelling of 'privileges.'</p>
<h4>gcc14: Fix issues caught by gcc 14</h4>
<p>Author: George Joseph
Date: 2025-01-03</p>
<ul>
<li>reqresp_parser.c: Fix misuse of "static" with linked list definitions</li>
<li>test_message.c: Fix segfaults caused by passing NULL as an sprintf fmt</li>
</ul>
<h4>Allow C++ source files (as extension .cc) in the main directory</h4>
<p>Author: George Joseph
Date: 2024-12-09</p>
<p>Although C++ files (as extension .cc) have been handled in the module
directories for many years, the main directory was missing one line in its
Makefile that prevented C++ files from being recognised there.</p>
<h4>res_stir_shaken: Allow sending Identity headers for unknown TNs</h4>
<p>Author: George Joseph
Date: 2024-11-08</p>
<p>Added a new option "unknown_tn_attest_level" to allow Identity
headers to be sent when a callerid TN isn't explicitly configured
in stir_shaken.conf. Since there's no TN object, a private_key_file
and public_cert_url must be configured in the attestation or profile
objects.</p>
<p>Since "unknown_tn_attest_level" uses the same enum as attest_level,
some of the sorcery macros had to be refactored to allow sharing
the enum and to/from string conversion functions.</p>
<p>Also fixed a memory leak in crypto_utils:pem_file_cb().</p>
<p>Resolves: #921</p>
<p>UserNote: You can now set the "unknown_tn_attest_level" option
in the attestation and/or profile objects in stir_shaken.conf to
enable sending Identity headers for callerid TNs not explicitly
configured.</p>
<h4>res_pjsip: Change suppress_moh_on_sendonly to OPT_BOOL_T</h4>
<p>Author: George Joseph
Date: 2024-11-15</p>
<p>The suppress_moh_on_sendonly endpoint option should have been
defined as OPT_BOOL_T in pjsip_configuration.c and AST_BOOL_VALUES
in the alembic script instead of OPT_YESNO_T and YESNO_VALUES.</p>
<p>Also updated contrib/ast-db-manage/README.md to indicate that
AST_BOOL_VALUES should always be used and provided an example.</p>
<p>Resolves: #995</p>
<h4>res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"</h4>
<p>Author: George Joseph
Date: 2024-11-05</p>
<p>Normally, when one party in a call sends Asterisk an SDP with
a "sendonly" or "inactive" attribute it means "hold" and causes
Asterisk to start playing MOH back to the other party. This can be
problematic if it happens at certain times, such as in a 183
Progress message, because the MOH will replace any early media you
may be playing to the calling party. If you set this option
to "yes" on an endpoint and the endpoint receives an SDP
with "sendonly" or "inactive", Asterisk will NOT play MOH back to
the other party.</p>
<p>Resolves: #979</p>
<p>UserNote: The new "suppress_moh_on_sendonly" endpoint option
can be used to prevent playing MOH back to a caller if the remote
end sends "sendonly" or "inactive" (hold) to Asterisk in an SDP.</p>
<h4>samples: remove and/or change some wiki mentions</h4>
<p>Author: chrsmj
Date: 2024-11-01</p>
<p>Cleaned some dead links. Replaced word wiki with
either docs or link to https://docs.asterisk.org/</p>
<p>Resolves: #974</p>
<h4>res_pjsip: Move tenantid to end of ast_sip_endpoint</h4>
<p>Author: George Joseph
Date: 2024-11-06</p>
<p>The tenantid field was originally added to the ast_sip_endpoint
structure at the end of the AST_DECLARE_STRING_FIELDS block. This
caused everything after it in the structure to move down in memory
and break ABI compatibility. It's now at the end of the structure
as an AST_STRING_FIELD_EXTENDED. Given the number of string fields
in the structure now, the initial string field allocation was
also increased from 64 to 128 bytes.</p>
<p>Resolves: #982</p>
<h4>Add res_pjsip_config_sangoma external module.</h4>
<p>Author: Ben Ford
Date: 2024-11-01</p>
<p>Adds res_pjsip_config_sangoma as an external module that can be
downloaded via menuselect. It lives under the Resource Modules section.</p>
<h4>app_mixmonitor: Add 'D' option for dual-channel audio.</h4>
<p>Author: Ben Ford
Date: 2024-10-28</p>
<p>Adds the 'D' option to app_mixmonitor that interleaves the input and
output frames of the channel being recorded in the monitor output frame.
This allows for two streams in the recording: the transmitted audio and
the received audio. The 't' and 'r' options are compatible with this.</p>
<p>Fixes: #945</p>
<p>UserNote: The MixMonitor application now has a new 'D' option which
interleaves the recorded audio in the output frames. This allows for
stereo recording output with one channel being the transmitted audio and
the other being the received audio. The 't' and 't' options are
compatible with this.</p>
<h4>core_unreal.c: Fix memory leak in ast_unreal_new_channels()</h4>
<p>Author: George Joseph
Date: 2024-10-15</p>
<p>When the channel tech is multistream capable, the reference to
chan_topology was passed to the new channel. When the channel tech
isn't multistream capable, the reference to chan_topology was never
released. "Local" channels are multistream capable so it didn't
affect them but the confbridge "CBAnn" and the bridge_media
"Recorder" channels are not so they caused a leak every time one
of them was created.</p>
<p>Also added tracing to ast_stream_topology_alloc() and
stream_topology_destroy() to assist with debugging.</p>
<p>Resolves: #938</p>
<h4>geolocation.sample.conf: Fix comment marker at end of file</h4>
<p>Author: George Joseph
Date: 2024-10-08</p>
<p>Resolves: #937</p>
<h4>manager.c: Add unit test for Originate app and appdata permissions</h4>
<p>Author: George Joseph
Date: 2024-10-03</p>
<p>This unit test checks that dialplan apps and app data specified
as parameters for the Originate action are allowed with the
permissions the user has.</p>
<h4>alembic: Drop redundant voicemail_messages index.</h4>
<p>Author: Sean Bright
Date: 2024-09-26</p>
<p>The <code>voicemail_messages_dir</code> index is a left prefix of the table's
primary key and therefore unnecessary.</p>
<h4>res_pjsip_pubsub: Persist subscription 'generator_data' in sorcery</h4>
<p>Author: Sean Bright
Date: 2024-09-23</p>
<p>Fixes #895</p>
<h4>stir_shaken: Fix propagation of attest_level and a few other values</h4>
<p>Author: George Joseph
Date: 2024-09-24</p>
<p>attest_level, send_mky and check_tn_cert_public_url weren't
propagating correctly from the attestation object to the profile
and tn.</p>
<ul>
<li>
<p>In the case of attest_level, the enum needed to be changed
so the "0" value (the default) was "NOT_SET" instead of "A". This
now allows the merging of the attestation object, profile and tn
to detect when a value isn't set and use the higher level value.</p>
</li>
<li>
<p>For send_mky and check_tn_cert_public_url, the tn default was
forced to "NO" which always overrode the profile and attestation
objects. Their defaults are now "NOT_SET" so the propagation
happens correctly.</p>
</li>
<li>
<p>Just to remove some redundant code in tn_config.c, a bunch of calls to
generate_sorcery_enum_from_str() and generate_sorcery_enum_to_str() were
replaced with a single call to generate_acfg_common_sorcery_handlers().</p>
</li>
</ul>
<p>Resolves: #904</p>
<h4>Fix application references to Background</h4>
<p>Author: George Joseph
Date: 2024-09-20</p>
<p>The app is actually named "BackGround" but several references
in XML documentation were spelled "Background" with the lower
case "g". This was causing documentation links to return
"not found" messages.</p>
<h4>db.c: Remove limit on family/key length</h4>
<p>Author: George Joseph
Date: 2024-09-11</p>
<p>Consumers like media_cache have been running into issues with
the previous astdb "/family/key" limit of 253 bytes when needing
to store things like long URIs. An Amazon S3 URI is a good example
of this. Now, instead of using a static 256 byte buffer for
"/family/key", we use ast_asprintf() to dynamically create it.</p>
<p>Both test_db.c and test_media_cache.c were also updated to use
keys/URIs over the old 253 character limit.</p>
<p>Resolves: #881</p>
<p>UserNote: The <code>ast_db_*()</code> APIs have had the 253 byte limit on
"/family/key" removed and will now accept families and keys with a
total length of up to SQLITE_MAX_LENGTH (currently 1e9!). This
affects the <code>DB*</code> dialplan applications, dialplan functions,
manager actions and <code>databse</code> CLI commands. Since the
media_cache also uses the <code>ast_db_*()</code> APIs, you can now store
resources with URIs longer than 253 bytes.</p>
<h4>res_stir_shaken: Remove stale include for jansson.h in verification.c</h4>
<p>Author: George Joseph
Date: 2024-09-17</p>
<p>verification.c had an include for jansson.h left over from previous
versions of the module. Since res_stir_shaken no longer has a
dependency on jansson, the bundled version wasn't added to GCC's
include path so if you didn't also have a jansson development package
installed, the compile would fail. Removing the stale include
was the only thing needed.</p>
<p>Resolves: #889</p>
<h4>res_stir_shaken.c: Fix crash when stir_shaken.conf is invalid</h4>
<p>Author: George Joseph
Date: 2024-09-13</p>
<ul>
<li>
<p>If the call to ast_config_load() returns CONFIG_STATUS_FILEINVALID,
check_for_old_config() now returns LOAD_DECLINE instead of continuing
on with a bad pointer.</p>
</li>
<li>
<p>If CONFIG_STATUS_FILEMISSING is returned, check_for_old_config()
assumes the config is being loaded from realtime and now returns
LOAD_SUCCESS. If it's actually not being loaded from realtime,
sorcery will catch that later on.</p>
</li>
<li>
<p>Also refactored the error handling in load_module() a bit.</p>
</li>
</ul>
<p>Resolves: #884</p>
<h4>res_stir_shaken: Check for disabled before param validation</h4>
<p>Author: George Joseph
Date: 2024-09-11</p>
<p>For both attestation and verification, we now check whether they've
been disabled either globally or by the profile before validating
things like callerid, orig_tn, dest_tn, etc. This prevents useless
error messages.</p>
<p>Resolves: #879</p>
</body></html>

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ChangeLogs/ChangeLog-certified-20.7-cert6.md Normal file
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@@ -0,0 +1,618 @@
## Change Log for Release asterisk-certified-20.7-cert6
### Links:
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-20.7-cert6.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/certified-20.7-cert5...certified-20.7-cert6)
- [Tarball](https://downloads.asterisk.org/pub/telephony/certified-asterisk/asterisk-certified-20.7-cert6.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/certified-asterisk)
### Summary:
- Commits: 31
- Commit Authors: 5
- Issues Resolved: 16
- Security Advisories Resolved: 0
### User Notes:
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
You can now set the "unknown_tn_attest_level" option
in the attestation and/or profile objects in stir_shaken.conf to
enable sending Identity headers for callerid TNs not explicitly
configured.
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
The new "suppress_moh_on_sendonly" endpoint option
can be used to prevent playing MOH back to a caller if the remote
end sends "sendonly" or "inactive" (hold) to Asterisk in an SDP.
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
The MixMonitor application now has a new 'D' option which
interleaves the recorded audio in the output frames. This allows for
stereo recording output with one channel being the transmitted audio and
the other being the received audio. The 't' and 't' options are
compatible with this.
- #### db.c: Remove limit on family/key length
The `ast_db_*()` APIs have had the 253 byte limit on
"/family/key" removed and will now accept families and keys with a
total length of up to SQLITE_MAX_LENGTH (currently 1e9!). This
affects the `DB*` dialplan applications, dialplan functions,
manager actions and `databse` CLI commands. Since the
media_cache also uses the `ast_db_*()` APIs, you can now store
resources with URIs longer than 253 bytes.
### Upgrade Notes:
### Commit Authors:
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (22)
- Joshua C. Colp: (1)
- Sean Bright: (4)
## Issue and Commit Detail:
### Closed Issues:
- 879: [bug]: res_stir_shaken/verification.c: Getting verification errors when global_disable=yes
- 881: [bug]: Long URLs are being rejected by the media cache because of an astdb key length limit
- 884: [bug]: A ':' at the top of in stir_shaken.conf make Asterisk producing a core file when starting
- 889: [bug]: res_stir_shaken/verification.c has a stale include for jansson.h that can cause compilation to fail
- 904: [bug]: stir_shaken: attest_level isn't being propagated correctly from attestation to profile to tn
- 921: [bug]: Stir-Shaken doesn’t allow B or C attestation for unknown callerid which is allowed by ATIS-1000074.v003, §5.2.4
- 937: [bug]: Wrong format for sample config file 'geolocation.conf.sample'
- 938: [bug]: memory leak - CBAnn leaks a small amount format_cap related memory for every confbridge
- 945: [improvement]: Add stereo recording support for app_mixmonitor
- 974: [improvement]: change and/or remove some wiki mentions to docs mentions in the sample configs
- 979: [improvement]: Add ability to suppress MOH when a remote endpoint sends "sendonly" or "inactive"
- 982: [bug]: The addition of tenantid to the ast_sip_endpoint structure broke ABI compatibility
- 995: [bug]: suppress_moh_on_sendonly should use AST_BOOL_VALUES instead of YESNO_VALUES in alembic script
- 1112: [bug]: STIR/SHAKEN verification doesn't allow anonymous callerid to be passed to the dialplan.
- 1122: Need status on CVE-2024-57520 claim.
- 1131: [bug]: CHANGES link broken in README.md
### Commits By Author:
- #### Ben Ford (3):
- app_mixmonitor: Add 'D' option for dual-channel audio.
- Add res_pjsip_config_sangoma external module.
- documentation: Update Gosub, Goto, and add new documentationtype.
- #### George Joseph (22):
- res_stir_shaken: Check for disabled before param validation
- res_stir_shaken.c: Fix crash when stir_shaken.conf is invalid
- res_stir_shaken: Remove stale include for jansson.h in verification.c
- db.c: Remove limit on family/key length
- Fix application references to Background
- stir_shaken: Fix propagation of attest_level and a few other values
- manager.c: Add unit test for Originate app and appdata permissions
- geolocation.sample.conf: Fix comment marker at end of file
- core_unreal.c: Fix memory leak in ast_unreal_new_channels()
- res_pjsip: Move tenantid to end of ast_sip_endpoint
- res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
- res_pjsip: Change suppress_moh_on_sendonly to OPT_BOOL_T
- res_stir_shaken: Allow sending Identity headers for unknown TNs
- Allow C++ source files (as extension .cc) in the main directory
- gcc14: Fix issues caught by gcc 14
- manager.c: Split XML docs into separate file
- docs: Enable since/version handling for XML, CLI and ARI documentation
- README.md, asterisk.c: Update Copyright Dates
- res_stir_shaken: Allow missing or anonymous CID to continue to the dialplan.
- swagger_model.py: Fix invalid escape sequence in get_list_parameter_type().
- manager.c: Check for restricted file in action_createconfig.
- README.md: Updates and Fixes
- #### Joshua C. Colp (1):
- LICENSE: Update company name, email, and address.
- #### Sean Bright (4):
- res_pjsip_pubsub: Persist subscription 'generator_data' in sorcery
- alembic: Drop redundant voicemail_messages index.
- manager.c: Rename restrictedFile to is_restricted_file.
- manager: Add `<since>` tags for all AMI actions.
- #### chrsmj (1):
- samples: remove and/or change some wiki mentions
### Commit List:
- documentation: Update Gosub, Goto, and add new documentationtype.
- README.md: Updates and Fixes
- manager.c: Check for restricted file in action_createconfig.
- swagger_model.py: Fix invalid escape sequence in get_list_parameter_type().
- res_stir_shaken: Allow missing or anonymous CID to continue to the dialplan.
- LICENSE: Update company name, email, and address.
- README.md, asterisk.c: Update Copyright Dates
- docs: Enable since/version handling for XML, CLI and ARI documentation
- manager: Add `<since>` tags for all AMI actions.
- manager.c: Split XML docs into separate file
- manager.c: Rename restrictedFile to is_restricted_file.
- gcc14: Fix issues caught by gcc 14
- Allow C++ source files (as extension .cc) in the main directory
- res_stir_shaken: Allow sending Identity headers for unknown TNs
- res_pjsip: Change suppress_moh_on_sendonly to OPT_BOOL_T
- res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
- samples: remove and/or change some wiki mentions
- res_pjsip: Move tenantid to end of ast_sip_endpoint
- Add res_pjsip_config_sangoma external module.
- app_mixmonitor: Add 'D' option for dual-channel audio.
- core_unreal.c: Fix memory leak in ast_unreal_new_channels()
- geolocation.sample.conf: Fix comment marker at end of file
- manager.c: Add unit test for Originate app and appdata permissions
- alembic: Drop redundant voicemail_messages index.
- res_pjsip_pubsub: Persist subscription 'generator_data' in sorcery
- stir_shaken: Fix propagation of attest_level and a few other values
- Fix application references to Background
- db.c: Remove limit on family/key length
- res_stir_shaken: Remove stale include for jansson.h in verification.c
- res_stir_shaken.c: Fix crash when stir_shaken.conf is invalid
- res_stir_shaken: Check for disabled before param validation
### Commit Details:
#### documentation: Update Gosub, Goto, and add new documentationtype.
Author: Ben Ford
Date: 2025-03-14
Gosub and Goto were not displaying their syntax correctly on the docs
site. This change adds a new way to specify an optional context, an
optional extension, and a required priority that the xml stylesheet can
parse without having to know which optional parameters come in which
order. In Asterisk, it looks like this:
parameter name="context" documentationtype="dialplan_context"
parameter name="extension" documentationtype="dialplan_extension"
parameter name="priority" documentationtype="dialplan_priority" required="true"
The stylesheet will ignore the context and extension parameters, but for
priority, it will automatically inject the following:
[[context,]extension,]priority
This is the correct oder for applications such as Gosub and Goto.
#### README.md: Updates and Fixes
Author: George Joseph
Date: 2025-03-05
* Outdated information has been removed.
* New links added.
* Placeholder added for link to change logs.
Going forward, the release process will create HTML versions of the README
and change log and will update the link in the README to the current
change log for the branch...
* In the development branches, the link will always point to the current
release on GitHub.
* In the "releases/*" branches and the tarballs, the link will point to the
ChangeLogs/ChangeLog-<version>.html file in the source directory.
* On the downloads website, the link will point to the
ChangeLog-<version>.html file in the same directory.
Resolves: #1131
#### manager.c: Check for restricted file in action_createconfig.
Author: George Joseph
Date: 2025-03-03
The `CreateConfig` manager action now ensures that a config file can
only be created in the AST_CONFIG_DIR unless `live_dangerously` is set.
Resolves: #1122
#### swagger_model.py: Fix invalid escape sequence in get_list_parameter_type().
Author: George Joseph
Date: 2025-03-04
Recent python versions complain when backslashes in strings create invalid
escape sequences. This causes issues for strings used as regex patterns like
`'^List\[(.*)\]$'` where you want the regex parser to treat `[` and `]`
as literals. Double-backslashing is one way to fix it but simply converting
the string to a raw string `re.match(r'^List\[(.*)\]$', text)` is easier
and less error prone.
#### res_stir_shaken: Allow missing or anonymous CID to continue to the dialplan.
Author: George Joseph
Date: 2025-02-05
The verification check for missing or anonymous callerid was happening before
the endpoint's profile was retrieved which meant that the failure_action
parameter wasn't available. Therefore, if verification was enabled and there
was no callerid or it was "anonymous", the call was immediately terminated
instead of giving the dialplan the ability to decide what to do with the call.
* The callerid check now happens after the verification context is created and
the endpoint's stir_shaken_profile is available.
* The check now processes the callerid failure just as it does for other
verification failures and respects the failure_action parameter. If set
to "continue" or "continue_return_reason", `STIR_SHAKEN(0,verify_result)`
in the dialplan will return "invalid_or_no_callerid".
* If the endpoint's failure_action is "reject_request", the call will be
rejected with `433 "Anonymity Disallowed"`.
* If the endpoint's failure_action is "continue_return_reason", the call will
continue but a `Reason: STIR; cause=433; text="Anonymity Disallowed"`
header will be added to the next provisional or final response.
Resolves: #1112
#### LICENSE: Update company name, email, and address.
Author: Joshua C. Colp
Date: 2025-01-21
#### README.md, asterisk.c: Update Copyright Dates
Author: George Joseph
Date: 2025-01-20
#### docs: Enable since/version handling for XML, CLI and ARI documentation
Author: George Joseph
Date: 2025-01-09
* Added the "since" element to the XML configObject and configOption elements
in appdocsxml.dtd.
* Added the "Since" section to the following CLI output:
```
config show help <module> <object>
config show help <module> <object> <option>
core show application <app>
core show function <func>
manager show command <command>
manager show event <event>
agi show commands topic <topic>
```
* Refactored the commands above to output their sections in the same order:
Synopsis, Since, Description, Syntax, Arguments, SeeAlso
* Refactored the commands above so they all use the same pattern for writing
the output to the CLI.
* Fixed several memory leaks caused by failure to free temporary output
buffers.
* Added a "since" array to the mustache template for the top-level resources
(Channel, Endpoint, etc.) and to the paths/methods underneath them. These
will be added to the generated markdown if present.
Example:
```
"resourcePath": "/api-docs/channels.{format}",
"requiresModules": [
"res_stasis_answer",
"res_stasis_playback",
"res_stasis_recording",
"res_stasis_snoop"
],
"since": [
"18.0.0",
"21.0.0"
],
"apis": [
{
"path": "/channels",
"description": "Active channels",
"operations": [
{
"httpMethod": "GET",
"since": [
"18.6.0",
"21.8.0"
],
"summary": "List all active channels in Asterisk.",
"nickname": "list",
"responseClass": "List[Channel]"
},
```
NOTE: No versioning information is actually added in this commit.
Those will be added separately and instructions for adding and maintaining
them will be published on the documentation site at a later date.
#### manager: Add `<since>` tags for all AMI actions.
Author: Sean Bright
Date: 2025-01-02
#### manager.c: Split XML docs into separate file
Author: George Joseph
Date: 2025-01-13
To keep the source tree somewhat compatible with the base 20 branch
the XML documentation from manager.c has been extracted into manager_doc.xml.
This will give future cherry-picks a better channce of succeeding without
manual intervention.
#### manager.c: Rename restrictedFile to is_restricted_file.
Author: Sean Bright
Date: 2025-01-09
Also correct the spelling of 'privileges.'
#### gcc14: Fix issues caught by gcc 14
Author: George Joseph
Date: 2025-01-03
* reqresp_parser.c: Fix misuse of "static" with linked list definitions
* test_message.c: Fix segfaults caused by passing NULL as an sprintf fmt
#### Allow C++ source files (as extension .cc) in the main directory
Author: George Joseph
Date: 2024-12-09
Although C++ files (as extension .cc) have been handled in the module
directories for many years, the main directory was missing one line in its
Makefile that prevented C++ files from being recognised there.
#### res_stir_shaken: Allow sending Identity headers for unknown TNs
Author: George Joseph
Date: 2024-11-08
Added a new option "unknown_tn_attest_level" to allow Identity
headers to be sent when a callerid TN isn't explicitly configured
in stir_shaken.conf. Since there's no TN object, a private_key_file
and public_cert_url must be configured in the attestation or profile
objects.
Since "unknown_tn_attest_level" uses the same enum as attest_level,
some of the sorcery macros had to be refactored to allow sharing
the enum and to/from string conversion functions.
Also fixed a memory leak in crypto_utils:pem_file_cb().
Resolves: #921
UserNote: You can now set the "unknown_tn_attest_level" option
in the attestation and/or profile objects in stir_shaken.conf to
enable sending Identity headers for callerid TNs not explicitly
configured.
#### res_pjsip: Change suppress_moh_on_sendonly to OPT_BOOL_T
Author: George Joseph
Date: 2024-11-15
The suppress_moh_on_sendonly endpoint option should have been
defined as OPT_BOOL_T in pjsip_configuration.c and AST_BOOL_VALUES
in the alembic script instead of OPT_YESNO_T and YESNO_VALUES.
Also updated contrib/ast-db-manage/README.md to indicate that
AST_BOOL_VALUES should always be used and provided an example.
Resolves: #995
#### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
Author: George Joseph
Date: 2024-11-05
Normally, when one party in a call sends Asterisk an SDP with
a "sendonly" or "inactive" attribute it means "hold" and causes
Asterisk to start playing MOH back to the other party. This can be
problematic if it happens at certain times, such as in a 183
Progress message, because the MOH will replace any early media you
may be playing to the calling party. If you set this option
to "yes" on an endpoint and the endpoint receives an SDP
with "sendonly" or "inactive", Asterisk will NOT play MOH back to
the other party.
Resolves: #979
UserNote: The new "suppress_moh_on_sendonly" endpoint option
can be used to prevent playing MOH back to a caller if the remote
end sends "sendonly" or "inactive" (hold) to Asterisk in an SDP.
#### samples: remove and/or change some wiki mentions
Author: chrsmj
Date: 2024-11-01
Cleaned some dead links. Replaced word wiki with
either docs or link to https://docs.asterisk.org/
Resolves: #974
#### res_pjsip: Move tenantid to end of ast_sip_endpoint
Author: George Joseph
Date: 2024-11-06
The tenantid field was originally added to the ast_sip_endpoint
structure at the end of the AST_DECLARE_STRING_FIELDS block. This
caused everything after it in the structure to move down in memory
and break ABI compatibility. It's now at the end of the structure
as an AST_STRING_FIELD_EXTENDED. Given the number of string fields
in the structure now, the initial string field allocation was
also increased from 64 to 128 bytes.
Resolves: #982
#### Add res_pjsip_config_sangoma external module.
Author: Ben Ford
Date: 2024-11-01
Adds res_pjsip_config_sangoma as an external module that can be
downloaded via menuselect. It lives under the Resource Modules section.
#### app_mixmonitor: Add 'D' option for dual-channel audio.
Author: Ben Ford
Date: 2024-10-28
Adds the 'D' option to app_mixmonitor that interleaves the input and
output frames of the channel being recorded in the monitor output frame.
This allows for two streams in the recording: the transmitted audio and
the received audio. The 't' and 'r' options are compatible with this.
Fixes: #945
UserNote: The MixMonitor application now has a new 'D' option which
interleaves the recorded audio in the output frames. This allows for
stereo recording output with one channel being the transmitted audio and
the other being the received audio. The 't' and 't' options are
compatible with this.
#### core_unreal.c: Fix memory leak in ast_unreal_new_channels()
Author: George Joseph
Date: 2024-10-15
When the channel tech is multistream capable, the reference to
chan_topology was passed to the new channel. When the channel tech
isn't multistream capable, the reference to chan_topology was never
released. "Local" channels are multistream capable so it didn't
affect them but the confbridge "CBAnn" and the bridge_media
"Recorder" channels are not so they caused a leak every time one
of them was created.
Also added tracing to ast_stream_topology_alloc() and
stream_topology_destroy() to assist with debugging.
Resolves: #938
#### geolocation.sample.conf: Fix comment marker at end of file
Author: George Joseph
Date: 2024-10-08
Resolves: #937
#### manager.c: Add unit test for Originate app and appdata permissions
Author: George Joseph
Date: 2024-10-03
This unit test checks that dialplan apps and app data specified
as parameters for the Originate action are allowed with the
permissions the user has.
#### alembic: Drop redundant voicemail_messages index.
Author: Sean Bright
Date: 2024-09-26
The `voicemail_messages_dir` index is a left prefix of the table's
primary key and therefore unnecessary.
#### res_pjsip_pubsub: Persist subscription 'generator_data' in sorcery
Author: Sean Bright
Date: 2024-09-23
Fixes #895
#### stir_shaken: Fix propagation of attest_level and a few other values
Author: George Joseph
Date: 2024-09-24
attest_level, send_mky and check_tn_cert_public_url weren't
propagating correctly from the attestation object to the profile
and tn.
* In the case of attest_level, the enum needed to be changed
so the "0" value (the default) was "NOT_SET" instead of "A". This
now allows the merging of the attestation object, profile and tn
to detect when a value isn't set and use the higher level value.
* For send_mky and check_tn_cert_public_url, the tn default was
forced to "NO" which always overrode the profile and attestation
objects. Their defaults are now "NOT_SET" so the propagation
happens correctly.
* Just to remove some redundant code in tn_config.c, a bunch of calls to
generate_sorcery_enum_from_str() and generate_sorcery_enum_to_str() were
replaced with a single call to generate_acfg_common_sorcery_handlers().
Resolves: #904
#### Fix application references to Background
Author: George Joseph
Date: 2024-09-20
The app is actually named "BackGround" but several references
in XML documentation were spelled "Background" with the lower
case "g". This was causing documentation links to return
"not found" messages.
#### db.c: Remove limit on family/key length
Author: George Joseph
Date: 2024-09-11
Consumers like media_cache have been running into issues with
the previous astdb "/family/key" limit of 253 bytes when needing
to store things like long URIs. An Amazon S3 URI is a good example
of this. Now, instead of using a static 256 byte buffer for
"/family/key", we use ast_asprintf() to dynamically create it.
Both test_db.c and test_media_cache.c were also updated to use
keys/URIs over the old 253 character limit.
Resolves: #881
UserNote: The `ast_db_*()` APIs have had the 253 byte limit on
"/family/key" removed and will now accept families and keys with a
total length of up to SQLITE_MAX_LENGTH (currently 1e9!). This
affects the `DB*` dialplan applications, dialplan functions,
manager actions and `databse` CLI commands. Since the
media_cache also uses the `ast_db_*()` APIs, you can now store
resources with URIs longer than 253 bytes.
#### res_stir_shaken: Remove stale include for jansson.h in verification.c
Author: George Joseph
Date: 2024-09-17
verification.c had an include for jansson.h left over from previous
versions of the module. Since res_stir_shaken no longer has a
dependency on jansson, the bundled version wasn't added to GCC's
include path so if you didn't also have a jansson development package
installed, the compile would fail. Removing the stale include
was the only thing needed.
Resolves: #889
#### res_stir_shaken.c: Fix crash when stir_shaken.conf is invalid
Author: George Joseph
Date: 2024-09-13
* If the call to ast_config_load() returns CONFIG_STATUS_FILEINVALID,
check_for_old_config() now returns LOAD_DECLINE instead of continuing
on with a bad pointer.
* If CONFIG_STATUS_FILEMISSING is returned, check_for_old_config()
assumes the config is being loaded from realtime and now returns
LOAD_SUCCESS. If it's actually not being loaded from realtime,
sorcery will catch that later on.
* Also refactored the error handling in load_module() a bit.
Resolves: #884
#### res_stir_shaken: Check for disabled before param validation
Author: George Joseph
Date: 2024-09-11
For both attestation and verification, we now check whether they've
been disabled either globally or by the profile before validating
things like callerid, orig_tn, dest_tn, etc. This prevents useless
error messages.
Resolves: #879

209
README.html
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@@ -1,25 +1,22 @@
<html><head><title>Readme for asterisk-certified-20.7-cert5</title></head><body>
<html><head><title>Readme for asterisk-certified-20.7-cert6</title></head><body>
<h1>The Asterisk(R) Open Source PBX</h1>
<pre><code class="language-text"> By Mark Spencer &lt;markster@digium.com&gt; and the Asterisk.org developer community.
Copyright (C) 2001-2021 Sangoma Technologies Corporation and other copyright holders.
<pre><code>By Mark Spencer &lt;markster@digium.com&gt; and the Asterisk.org developer community.
Copyright (C) 2001-2025 Sangoma Technologies Corporation and other copyright holders.
</code></pre>
<h2>SECURITY</h2>
<p>It is imperative that you read and fully understand the contents of
the security information document before you attempt to configure and run
an Asterisk server.</p>
<p>See <a href="https://docs.asterisk.org/Deployment/Important-Security-Considerations/">Important Security Considerations</a> for more information.</p>
<p>See <a href="https://docs.asterisk.org/Deployment/Important-Security-Considerations">Important Security Considerations</a> for more information.</p>
<h2>WHAT IS ASTERISK ?</h2>
<p>Asterisk is an Open Source PBX and telephony toolkit. It is, in a
sense, middleware between Internet and telephony channels on the bottom,
and Internet and telephony applications at the top. However, Asterisk supports
more telephony interfaces than just Internet telephony. Asterisk also has a
vast amount of support for traditional PSTN telephony, as well.</p>
<p>For more information on the project itself, please visit the Asterisk
<a href="https://www.asterisk.org">home page</a> and the official <a href="https://docs.asterisk.org/">documentation</a>. In addition you'll find lots
of information compiled by the Asterisk community at <a href="http://www.voip-info.org/wiki-Asterisk">voip-info.org</a>.</p>
<p>There is a book on Asterisk published by O'Reilly under the Creative Commons
License. It is available in book stores as well as in a downloadable version on
the <a href="http://www.asteriskdocs.org">asteriskdocs.org</a> web site.</p>
<p>For more information on the project itself, please visit the <a href="https://www.asterisk.org">Asterisk
Home Page</a> and the official
<a href="https://docs.asterisk.org">Asterisk Documentation</a>.</p>
<h2>SUPPORTED OPERATING SYSTEMS</h2>
<h3>Linux</h3>
<p>The Asterisk Open Source PBX is developed and tested primarily on the
@@ -27,26 +24,22 @@ GNU/Linux operating system, and is supported on every major GNU/Linux
distribution.</p>
<h3>Others</h3>
<p>Asterisk has also been 'ported' and reportedly runs properly on other
operating systems as well, including Sun Solaris, Apple's Mac OS X, Cygwin,
and the BSD variants.</p>
operating systems as well, Apple's Mac OS X, and the BSD variants.</p>
<h2>GETTING STARTED</h2>
<p>First, be sure you've got supported hardware (but note that you don't need
ANY special hardware, not even a sound card) to install and run Asterisk.</p>
<p>Most users are using VoIP/SIP exclusively these days but if you need to
interface to TDM or analog services or devices, be sure you've got supported
hardware.</p>
<p>Supported telephony hardware includes:
* All Analog and Digital Interface cards from <a href="https://www.sangoma.com/">Sangoma</a>
* QuickNet Internet PhoneJack and LineJack
* any full duplex sound card supported by ALSA, OSS, or PortAudio
* any ISDN card supported by mISDN on Linux
* The Xorcom Astribank channel bank
* VoiceTronix OpenLine products</p>
* All Analog and Digital Interface cards from Sangoma
* Any full duplex sound card supported by PortAudio
* The Xorcom Astribank channel bank</p>
<h3>UPGRADING FROM AN EARLIER VERSION</h3>
<p>If you are updating from a previous version of Asterisk, make sure you
read the <a href="UPGRADE.txt">UPGRADE.txt</a> file in the source directory. There are some files
and configuration options that you will have to change, even though we
made every effort possible to maintain backwards compatibility.</p>
<p>In order to discover new features to use, please check the configuration
examples in the <a href="configs">configs</a> directory of the source code distribution. For a
list of new features in this version of Asterisk, see the <a href="CHANGES">CHANGES</a> file.</p>
read the Change Logs.</p>
<!-- CHANGELOGS (the URL will change based on the location of this README) -->
<p><a href="ChangeLogs/ChangeLog-certified-20.7-cert6.html">Change Logs</a></p>
<!-- END-CHANGELOGS -->
<h3>NEW INSTALLATIONS</h3>
<p>Ensure that your system contains a compatible compiler and development
libraries. Asterisk requires either the GNU Compiler Collection (GCC) version
@@ -58,111 +51,79 @@ libraries are being looked for, see <code>./configure --help</code>, or run
<code>make menuselect</code> to view the dependencies for specific modules.</p>
<p>On many distributions, these dependencies are installed by packages with names
like 'glibc-devel', 'ncurses-devel', 'openssl-devel' and 'zlib-devel'
or similar.</p>
<p>So, let's proceed:
1. Read this file.</p>
<p>There are more documents than this one in the <a href="doc">doc</a> directory. You may also
want to check the configuration files that contain examples and reference
guides in the <a href="configs">configs</a> directory.</p>
or similar. The <code>contrib/scripts/install_prereq</code> script can be used to install
the dependencies for most Debian and Redhat based Linux distributions.
The script also handles SUSE, Arch, Gentoo, FreeBSD, NetBSD and OpenBSD but
those distributions mightnoit have complete support or they might be out of date.</p>
<p>So, let's proceed:</p>
<ol>
<li>Run <code>./configure</code></li>
</ol>
<p>Execute the configure script to guess values for system-dependent
variables used during compilation. If the script indicates that some required
<li>
<p>Read the documentation.<br>
The <a href="https://docs.asterisk.org">Asterisk Documentation</a> website has full
information for building, installing, configuring and running Asterisk.</p>
</li>
<li>
<p>Run <code>./configure</code><br>
Execute the configure script to guess values for system-dependent
variables used during compilation. If the script indicates that some required
components are missing, you can run <code>./contrib/scripts/install_prereq install</code>
to install the necessary components. Note that this will install all dependencies for every functionality of Asterisk. After running the script, you will need
to install the necessary components. Note that this will install all dependencies
for every functionality of Asterisk. After running the script, you will need
to rerun <code>./configure</code>.</p>
<ol>
<li>Run <code>make menuselect</code> <em>[optional]</em></li>
</ol>
<p>This is needed if you want to select the modules that will be compiled and to
</li>
<li>
<p>Run <code>make menuselect</code><br>
This is needed if you want to select the modules that will be compiled and to
check dependencies for various optional modules.</p>
<ol>
<li>Run <code>make</code></li>
</ol>
<p>Assuming the build completes successfully:</p>
<ol>
<li>Run <code>make install</code></li>
</ol>
<p>If this is your first time working with Asterisk, you may wish to install
</li>
<li>
<p>Run <code>make</code><br>
Assuming the build completes successfully:</p>
</li>
<li>
<p>Run <code>make install</code><br>
If this is your first time working with Asterisk, you may wish to install
the sample PBX, with demonstration extensions, etc. If so, run:</p>
<ol>
<li>Run <code>make samples</code></li>
</ol>
<p>Doing so will overwrite any existing configuration files you have installed.</p>
<ol>
<li>Finally, you can launch Asterisk in the foreground mode (not a daemon) with:</li>
</ol>
<pre><code> # asterisk -vvvc
</code></pre>
<p>You'll see a bunch of verbose messages fly by your screen as Asterisk
</li>
<li>
<p>Run <code>make samples</code><br>
Doing so will overwrite any existing configuration files you have installed.</p>
</li>
<li>
<p>Finally, you can launch Asterisk in the foreground mode (not a daemon) with
<code>asterisk -vvvc</code><br>
You'll see a bunch of verbose messages fly by your screen as Asterisk
initializes (that's the "very very verbose" mode). When it's ready, if
you specified the "c" then you'll get a command line console, that looks
like this:</p>
<pre><code> *CLI&gt;
</code></pre>
<p>You can type "core show help" at any time to get help with the system. For help
with a specific command, type "core show help <command>". To start the PBX using
your sound card, you can type "console dial" to dial the PBX. Then you can use
"console answer", "console hangup", and "console dial" to simulate the actions
of a telephone. Remember that if you don't have a full duplex sound card
(and Asterisk will tell you somewhere in its verbose messages if you do/don't)
then it won't work right (not yet).</p>
<p>"man asterisk" at the Unix/Linux command prompt will give you detailed
like this:<br>
<code>*CLI&gt;</code><br>
You can type <code>core show help</code> at any time to get help with the system. For help
with a specific command, type <code>core show help &lt;command&gt;</code>.</p>
</li>
</ol>
<p><code>man asterisk</code> at the Unix/Linux command prompt will give you detailed
information on how to start and stop Asterisk, as well as all the command
line options for starting Asterisk.</p>
<p>Feel free to look over the configuration files in <code>/etc/asterisk</code>, where you
will find a lot of information about what you can do with Asterisk.</p>
<h3>ABOUT CONFIGURATION FILES</h3>
<p>All Asterisk configuration files share a common format. Comments are
delimited by ';' (since '#' of course, being a DTMF digit, may occur in
delimited by <code>;</code> (since <code>#</code> of course, being a DTMF digit, may occur in
many places). A configuration file is divided into sections whose names
appear in []'s. Each section typically contains two types of statements,
those of the form 'variable = value', and those of the form 'object =&gt;
parameters'. Internally the use of '=' and '=&gt;' is exactly the same, so
they're used only to help make the configuration file easier to
understand, and do not affect how it is actually parsed.</p>
<p>Entries of the form 'variable=value' set the value of some parameter in
asterisk. For example, in <a href="configs/samples/chan_dahdi.conf.sample">chan_dahdi.conf</a>, one might specify:</p>
<pre><code> switchtype=national
</code></pre>
<p>In order to indicate to Asterisk that the switch they are connecting to is
of the type "national". In general, the parameter will apply to
instantiations which occur below its specification. For example, if the
configuration file read:</p>
<pre><code> switchtype = national
channel =&gt; 1-4
channel =&gt; 10-12
switchtype = dms100
channel =&gt; 25-47
</code></pre>
<p>The "national" switchtype would be applied to channels one through
four and channels 10 through 12, whereas the "dms100" switchtype would
apply to channels 25 through 47.</p>
<p>The "object =&gt; parameters" instantiates an object with the given
parameters. For example, the line "channel =&gt; 25-47" creates objects for
the channels 25 through 47 of the card, obtaining the settings
from the variables specified above.</p>
appear in <code>[]</code>'s. Each section typically contains statements in the form
<code>variable = value</code> although you may see <code>variable =&gt; value</code> in older samples.</p>
<h3>SPECIAL NOTE ON TIME</h3>
<p>Those using SIP phones should be aware that Asterisk is sensitive to
large jumps in time. Manually changing the system time using date(1)
(or other similar commands) may cause SIP registrations and other
internal processes to fail. If your system cannot keep accurate time
by itself use <a href="http://www.ntp.org/">NTP</a> to keep the system clock
synchronized to "real time". NTP is designed to keep the system clock
synchronized by speeding up or slowing down the system clock until it
is synchronized to "real time" rather than by jumping the time and
causing discontinuities. Most Linux distributions include precompiled
versions of NTP. Beware of some time synchronization methods that get
the correct real time periodically and then manually set the system
clock.</p>
<p>Apparent time changes due to daylight savings time are just that,
apparent. The use of daylight savings time in a Linux system is
purely a user interface issue and does not affect the operation of the
Linux kernel or Asterisk. The system clock on Linux kernels operates
on UTC. UTC does not use daylight savings time.</p>
<p>Also note that this issue is separate from the clocking of TDM
channels, and is known to at least affect SIP registrations.</p>
internal processes to fail. For this reason, you should always use
a time synchronization package to keep your system time accurate.
All OS/distributions make one or more of the following packages
available:</p>
<ul>
<li>ntpd/ntpsec</li>
<li>chronyd</li>
<li>systemd-timesyncd</li>
</ul>
<p>Be sure to install and configure one (and only one) of them.</p>
<h3>FILE DESCRIPTORS</h3>
<p>Depending on the size of your system and your configuration,
Asterisk can consume a large number of file descriptors. In UNIX,
@@ -192,14 +153,22 @@ these changes to take effect.</p>
above you can try adding the command <code>ulimit -n 8192</code> to the script
that starts Asterisk.</p>
<h2>MORE INFORMATION</h2>
<p>See the <a href="doc">doc</a> directory for more documentation on various features.
Again, please read all the configuration samples that include documentation
on the configuration options.</p>
<p>Finally, you may wish to visit the <a href="https://www.asterisk.org/support">support</a> site and join the <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">mailing
list</a> if you're interested in getting more information.</p>
<p>Visit the <a href="https://docs.asterisk.org">Asterisk Documentation</a> website
for more documentation on various features and please read all the
configuration samples that include documentation on the configuration options.</p>
<p>Finally, you may wish to join the
<a href="https://community.asterisk.org">Asterisk Community Forums</a></p>
<p>Welcome to the growing worldwide community of Asterisk users!</p>
<pre><code> Mark Spencer, and the Asterisk.org development community
</code></pre>
<hr>
<p>Asterisk is a trademark of Sangoma Technologies Corporation</p>
<p>[<a href="https://www.sangoma.com/">Sangoma</a>]
[<a href="https://www.asterisk.org">Home Page</a>]
[<a href="https://www.asterisk.org/support">Support</a>]
[<a href="https://docs.asterisk.org">Documentation</a>]
[<a href="https://community.asterisk.org">Community Forums</a>]
[<a href="https://github.com/asterisk/asterisk/releases">Release Notes</a>]
[<a href="https://docs.asterisk.org/Deployment/Important-Security-Considerations/">Security</a>]
[<a href="https://lists.digium.com">Mailing List Archive</a>] </p>
</body></html>

2
README.md
View File

@@ -55,7 +55,7 @@ If you are updating from a previous version of Asterisk, make sure you
read the Change Logs.
<!-- CHANGELOGS (the URL will change based on the location of this README) -->
[Change Logs](https://downloads.asterisk.org/pub/telephony/asterisk)
[Change Logs](ChangeLogs/ChangeLog-certified-20.7-cert6.html)
<!-- END-CHANGELOGS -->
### NEW INSTALLATIONS

12
contrib/realtime/mysql/mysql_config.sql
View File

@@ -1663,3 +1663,15 @@ ALTER TABLE ps_endpoints ADD COLUMN tenantid VARCHAR(80);
UPDATE alembic_version SET version_num='655054a68ad5' WHERE alembic_version.version_num = '6c475a93f48a';
-- Running upgrade 655054a68ad5 -> 801b9fced8b7
ALTER TABLE ps_subscription_persistence ADD COLUMN generator_data TEXT;
UPDATE alembic_version SET version_num='801b9fced8b7' WHERE alembic_version.version_num = '655054a68ad5';
-- Running upgrade 801b9fced8b7 -> 4f91fc18c979
ALTER TABLE ps_endpoints ADD COLUMN suppress_moh_on_sendonly ENUM('0','1','off','on','false','true','no','yes');
UPDATE alembic_version SET version_num='4f91fc18c979' WHERE alembic_version.version_num = '801b9fced8b7';

6
contrib/realtime/mysql/mysql_voicemail.sql
View File

@@ -37,3 +37,9 @@ UPDATE alembic_version SET version_num='39428242f7f5' WHERE alembic_version.vers
UPDATE alembic_version SET version_num='1c55c341360f' WHERE alembic_version.version_num = '39428242f7f5';
-- Running upgrade 1c55c341360f -> 64fae6bbe7fb
DROP INDEX voicemail_messages_dir ON voicemail_messages;
UPDATE alembic_version SET version_num='64fae6bbe7fb' WHERE alembic_version.version_num = '1c55c341360f';

12
contrib/realtime/postgresql/postgresql_config.sql
View File

@@ -1787,5 +1787,17 @@ ALTER TABLE ps_endpoints ADD COLUMN tenantid VARCHAR(80);
UPDATE alembic_version SET version_num='655054a68ad5' WHERE alembic_version.version_num = '6c475a93f48a';
-- Running upgrade 655054a68ad5 -> 801b9fced8b7
ALTER TABLE ps_subscription_persistence ADD COLUMN generator_data TEXT;
UPDATE alembic_version SET version_num='801b9fced8b7' WHERE alembic_version.version_num = '655054a68ad5';
-- Running upgrade 801b9fced8b7 -> 4f91fc18c979
ALTER TABLE ps_endpoints ADD COLUMN suppress_moh_on_sendonly ast_bool_values;
UPDATE alembic_version SET version_num='4f91fc18c979' WHERE alembic_version.version_num = '801b9fced8b7';
COMMIT;

6
contrib/realtime/postgresql/postgresql_voicemail.sql
View File

@@ -39,5 +39,11 @@ UPDATE alembic_version SET version_num='39428242f7f5' WHERE alembic_version.vers
UPDATE alembic_version SET version_num='1c55c341360f' WHERE alembic_version.version_num = '39428242f7f5';
-- Running upgrade 1c55c341360f -> 64fae6bbe7fb
DROP INDEX voicemail_messages_dir;
UPDATE alembic_version SET version_num='64fae6bbe7fb' WHERE alembic_version.version_num = '1c55c341360f';
COMMIT;