from:"Asterisk Development Team"

[asterisk-dev] asterisk release 21.1.0

2024-01-25 Thread Asterisk Development Team

The Asterisk Development Team would like to announce
the release of asterisk-21.1.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/21.1.0
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!

Change Log for Release asterisk-21.1.0

Links:

- [Full
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)

- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
-
[Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)

- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)

Summary:

- logger: Fix linking regression.
- Revert "core & res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version >= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add

[asterisk-dev] asterisk release 20.6.0

2024-01-25 Thread Asterisk Development Team

The Asterisk Development Team would like to announce
the release of asterisk-20.6.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/20.6.0
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!

Change Log for Release asterisk-20.6.0

Links:

- [Full
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)

- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
-
[Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)

- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)

Summary:

- logger: Fix linking regression.
- Revert "core & res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version >= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/

[asterisk-dev] asterisk release 18.21.0

2024-01-25 Thread Asterisk Development Team

The Asterisk Development Team would like to announce
the release of asterisk-18.21.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/18.21.0
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!

Change Log for Release asterisk-18.21.0

Links:

- [Full
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)

- [GitHub
Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
-
[Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)

- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)

Summary:

- logger: Fix linking regression.
- Revert "core & res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version >= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fi

[asterisk-dev] asterisk release 21.1.0-rc2

2024-01-18 Thread Asterisk Development Team

The Asterisk Development Team would like to announce  
release candidate 2 of asterisk-21.1.0.

The release artifacts are available for immediate download at  
https://github.com/asterisk/asterisk/releases/tag/21.1.0-rc2
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community  
and would have not been possible without your participation.

Thank You!


Change Log for Release asterisk-21.1.0-rc2


Links:


 - [Full 
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0-rc2.md)
  
 - [GitHub 
Diff](https://github.com/asterisk/asterisk/compare/21.1.0-rc1...21.1.0-rc2)  
 - 
[Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0-rc2.tar.gz)
  
 - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)  

Summary:


- logger: Fix linking regression.

User Notes:



Upgrade Notes:



Closed Issues:


  - #539: [bug]: Existence of logger.xml causes linking failure

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To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

[asterisk-dev] asterisk release 20.6.0-rc2

2024-01-18 Thread Asterisk Development Team

The Asterisk Development Team would like to announce  
release candidate 2 of asterisk-20.6.0.

The release artifacts are available for immediate download at  
https://github.com/asterisk/asterisk/releases/tag/20.6.0-rc2
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community  
and would have not been possible without your participation.

Thank You!


Change Log for Release asterisk-20.6.0-rc2


Links:


 - [Full 
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0-rc2.md)
  
 - [GitHub 
Diff](https://github.com/asterisk/asterisk/compare/20.6.0-rc1...20.6.0-rc2)  
 - 
[Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0-rc2.tar.gz)
  
 - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)  

Summary:


- logger: Fix linking regression.

User Notes:



Upgrade Notes:



Closed Issues:


  - #539: [bug]: Existence of logger.xml causes linking failure

-- 
_
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asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

[asterisk-dev] asterisk release 18.21.0-rc2

2024-01-18 Thread Asterisk Development Team

The Asterisk Development Team would like to announce  
release candidate 2 of asterisk-18.21.0.

The release artifacts are available for immediate download at  
https://github.com/asterisk/asterisk/releases/tag/18.21.0-rc2
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community  
and would have not been possible without your participation.

Thank You!


Change Log for Release asterisk-18.21.0-rc2


Links:


 - [Full 
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0-rc2.md)
  
 - [GitHub 
Diff](https://github.com/asterisk/asterisk/compare/18.21.0-rc1...18.21.0-rc2)  
 - 
[Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0-rc2.tar.gz)
  
 - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)  

Summary:


- logger: Fix linking regression.

User Notes:



Upgrade Notes:



Closed Issues:


  - #539: [bug]: Existence of logger.xml causes linking failure

-- 
_
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asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

[asterisk-dev] asterisk release 21.1.0-rc1

2024-01-12 Thread Asterisk Development Team

The Asterisk Development Team would like to announce
release candidate 1 of asterisk-21.1.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/21.1.0-rc1
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!

Change Log for Release asterisk-21.1.0-rc1

Links:

- [Full
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0-rc1.md)

- [GitHub
Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0-rc1)
-
[Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0-rc1.tar.gz)

- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)

Summary:

- Revert "core & res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version >= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add

[asterisk-dev] asterisk release 20.6.0-rc1

2024-01-12 Thread Asterisk Development Team

The Asterisk Development Team would like to announce
release candidate 1 of asterisk-20.6.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/20.6.0-rc1
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!

Change Log for Release asterisk-20.6.0-rc1

Links:

- [Full
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0-rc1.md)

- [GitHub
Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0-rc1)
-
[Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0-rc1.tar.gz)

- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)

Summary:

- Revert "core & res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version >= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- fun

[asterisk-dev] asterisk release 18.21.0-rc1

2024-01-12 Thread Asterisk Development Team

The Asterisk Development Team would like to announce
release candidate 1 of asterisk-18.21.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/18.21.0-rc1
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!

Change Log for Release asterisk-18.21.0-rc1

Links:

- [Full
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0-rc1.md)

- [GitHub
Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0-rc1)
-
[Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0-rc1.tar.gz)

- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)

Summary:

- Revert "core & res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version >= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix cras

[asterisk-dev] asterisk release 21.0.2

2023-12-20 Thread Asterisk Development Team

The Asterisk Development Team would like to announce  
the release of asterisk-21.0.2.

The release artifacts are available for immediate download at  
https://github.com/asterisk/asterisk/releases/tag/21.0.2
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community  
and would have not been possible without your participation.

Thank You!


Change Log for Release asterisk-21.0.2


Links:


 - [Full 
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
  
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)  
 - 
[Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
  
 - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)  

Summary:


- res_rtp_asterisk: Fix regression issues with DTLS client check

User Notes:



Upgrade Notes:



Closed Issues:


  - #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't 
used
  - #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails 
when it shouldn't
  - #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs 
created by ast_sockaddr_from_pj_sockaddr()

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[asterisk-dev] asterisk release 20.5.2

2023-12-20 Thread Asterisk Development Team

The Asterisk Development Team would like to announce  
the release of asterisk-20.5.2.

The release artifacts are available for immediate download at  
https://github.com/asterisk/asterisk/releases/tag/20.5.2
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community  
and would have not been possible without your participation.

Thank You!


Change Log for Release asterisk-20.5.2


Links:


 - [Full 
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
  
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)  
 - 
[Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
  
 - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)  

Summary:


- res_rtp_asterisk: Fix regression issues with DTLS client check

User Notes:



Upgrade Notes:



Closed Issues:


  - #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't 
used
  - #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails 
when it shouldn't
  - #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs 
created by ast_sockaddr_from_pj_sockaddr()

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[asterisk-dev] asterisk release 18.20.2

2023-12-20 Thread Asterisk Development Team

The Asterisk Development Team would like to announce  
the release of asterisk-18.20.2.

The release artifacts are available for immediate download at  
https://github.com/asterisk/asterisk/releases/tag/18.20.2
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community  
and would have not been possible without your participation.

Thank You!


Change Log for Release asterisk-18.20.2


Links:


 - [Full 
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
  
 - [GitHub 
Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)  
 - 
[Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
  
 - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)  

Summary:


- res_rtp_asterisk: Fix regression issues with DTLS client check

User Notes:



Upgrade Notes:



Closed Issues:


  - #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't 
used
  - #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails 
when it shouldn't
  - #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs 
created by ast_sockaddr_from_pj_sockaddr()

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[asterisk-dev] asterisk release certified-18.9-cert7

2023-12-20 Thread Asterisk Development Team

The Asterisk Development Team would like to announce  
the release of Certified asterisk-18.9-cert7.

The release artifacts are available for immediate download at  
https://github.com/asterisk/asterisk/releases/tag/certified-18.9-cert7
and
https://downloads.asterisk.org/pub/telephony/certified-asterisk

This release resolves issues reported by the community  
and would have not been possible without your participation.

Thank You!


Change Log for Release asterisk-certified-18.9-cert7


Links:


 - [Full 
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-certified-18.9-cert7.md)
  
 - [GitHub 
Diff](https://github.com/asterisk/asterisk/compare/certified-18.9-cert6...certified-18.9-cert7)
  
 - 
[Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-certified-18.9-cert7.tar.gz)
  
 - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)  

Summary:


- res_rtp_asterisk: Fix regression issues with DTLS client check

User Notes:



Upgrade Notes:



Closed Issues:


  - #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't 
used
  - #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails 
when it shouldn't
  - #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs 
created by ast_sockaddr_from_pj_sockaddr()

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[asterisk-dev] CORRECTED asterisk release 21.0.1

2023-12-14 Thread Asterisk Development Team

The earlier announcement should not have had any User or Upgrade notes.

The Asterisk Development Team would like to announce security release
Asterisk 21.0.1.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/21.0.1
and
https://downloads.asterisk.org/pub/telephony/asterisk

The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files](
https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f
)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during
call initiation](
https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq
)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ](
https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7
)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when
using 'update'](
https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh
)


Change Log for Release asterisk-21.0.1


Links:


 - [Full ChangeLog](
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)

 - [GitHub Diff](
https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
 - [Tarball](
https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)

 - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)

Summary:


- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.

User Notes:


Upgrade Notes:


Closed Issues:


None
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[asterisk-dev] CORRECTED asterisk release certified-18.9-cert6

2023-12-14 Thread Asterisk Development Team

The earlier release announcement should NOT have had any User or Upgrade
notes.

The Asterisk Development Team would like to announce security release
Certified Asterisk 18.9-cert6.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/certified-18.9-cert6
and
https://downloads.asterisk.org/pub/telephony/certified-asterisk

The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files](
https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f
)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during
call initiation](
https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq
)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ](
https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7
)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when
using 'update'](
https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh
)


Change Log for Release asterisk-certified-18.9-cert6


Links:


 - [Full ChangeLog](
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-certified-18.9-cert6.md)

 - [GitHub Diff](
https://github.com/asterisk/asterisk/compare/certified-18.9-cert5...certified-18.9-cert6)

 - [Tarball](
https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-certified-18.9-cert6.tar.gz)

 - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)

Summary:


- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
- res_pjsip: disable raw bad packet logging

User Notes:


Upgrade Notes:


Closed Issues:


None
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[asterisk-dev] asterisk release certified-18.9-cert6

2023-12-14 Thread Asterisk Development Team

The Asterisk Development Team would like to announce security release
Certified Asterisk 18.9-cert6.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/certified-18.9-cert6
and
https://downloads.asterisk.org/pub/telephony/certified-asterisk

The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside
files](https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries
](https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)

Change Log for Release asterisk-certified-18.9-cert6

Links:

- [Full
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-certified-18.9-cert6.md)

- [GitHub
Diff](https://github.com/asterisk/asterisk/compare/certified-18.9-cert5...certified-18.9-cert6)

-
[Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-certified-18.9-cert6.tar.gz)

- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)

Summary:

- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
- res_pjsip: disable raw bad packet logging

User Notes:

- ### app_read: Add an option to return terminator on empty digits.
A new option 'e' has been added to allow Read() to return the
terminator as the dialed digits in the case where only the terminator
is entered.

- ### format_sln: add .slin as supported file extension
format_sln now recognizes '.slin' as a valid
file extension in addition to the existing
'.sln' and '.raw'.

- ### app_directory: Add a 'skip call' option.
A new option 's' has been added to the Directory() application that
will skip calling the extension and instead set the extension as
DIRECTORY_EXTEN channel variable.

- ### app_senddtmf: Add option to answer target channel.
A new option has been added to SendDTMF() which will answer the
specified channel if it is not already up. If no channel is specified,
the current channel will be answered instead.

- ### cli: increase channel column width
This change increases the display width on 'core show channels'
amd 'core show channels verbose'
For 'core show channels', the Channel name field is increased to
64 characters and the Location name field is increased to 32
characters.
For 'core show channels verbose', the Channel name field is
increased to 80 characters, the Context is increased to 24
characters and the Extension is increased to 24 characters.

- ### bridge_builtin_features: add beep via touch variable
Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid
interval in seconds will result in a periodic beep being
played to the monitored channel upon MixMontior/Monitor
feature start.
If an interval less than 5 seconds is specified, the interval
will default to 5 seconds. If the value is set to an invalid
interval, the default of 15 seconds will be used.

- ### test.c: Fix counting of tests and add 2 new tests
The "tests" attribute of the "testsuite" element in the
output XML now reflects only the tests actually requested
to be executed instead of all the tests registered.
The "failures" attribute was added to the "testsuite"
element.
Also added two new unit tests that just pass and fail
to be used for testing CI itself.

- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
It is now possible to specify the MixMonitorID when calling
the manager action: MixMonitorMute. This will allow an
individual MixMonitor instance to be muted via ID.
The MixMonitorID can be stored as a channel variable using
the 'i' MixMonitor option and is returned upon creation if
this option is used.
As part of this change, if no MixMonitorID is specified in
the manager action MixMonitorMute, Asterisk will set the mute
flag on all MixMonitor audiohooks on the channel. Previous
behavior would set the flag on the first MixMonitor audiohook
found.

Upgrade Notes:

Closed Issues:
-

[asterisk-dev] asterisk release 21.0.1

2023-12-14 Thread Asterisk Development Team

The Asterisk Development Team would like to announce security release
Asterisk 21.0.1.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/21.0.1
and
https://downloads.asterisk.org/pub/telephony/asterisk

Change Log for Release asterisk-21.0.1

Links:

- [Full
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)

- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
-
[Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)

- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)

Summary:

- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.

User Notes:

- ### http.c: Minor simplification to HTTP status output.
For bound addresses, the HTTP status page now combines the bound
address and bound port in a single line. Additionally, the SSL bind
address has been renamed to TLS.

Upgrade Notes:

- ### chan_sip: Remove deprecated module.
This module was deprecated in Asterisk 17
and is now being removed in accordance with
the Asterisk Module Deprecation policy.

- ### res_monitor: Remove deprecated module.
This module was deprecated in Asterisk 16
and is now being removed in accordance with
the Asterisk Module Deprecation policy.
This also removes the 'w' and 'W' options
for app_queue.
MixMonitor should be default and only option
for all settings that previously used either
Monitor or MixMonitor.

- ### app_osplookup: Remove deprecated module.
This module was deprecated in Asterisk 19
and is now being removed in accordance with
the Asterisk Module Deprecation policy.

- ### app_cdr: Remove deprecated application and option.
The previously deprecated NoCDR application has been removed.
Additionally, the previously deprecated 'e' option to the ResetCDR
application has been removed.

- ### chan_skinny: Remove deprecated module.
This module was deprecated in Asterisk 19
and is now being removed in accordance with
the Asterisk Module Deprecation policy.

- ### chan_mgcp: Remove deprecated module.
This module was deprecated in Asterisk 19
and is now being removed in accordance with
the Asterisk Module Deprecation policy.

- ### translate.c: Prefer better codecs upon translate ties.
When setting up translation between two codecs the quality was not taken into
account,
resulting in suboptimal translation. The quality is now taken into account,
which can reduce the number of translation steps required, and improve the
resulting quality.

- ### app_macro: Remove deprecated module.
This module was deprecated in Asterisk 16
and is now being removed in accordance with
the Asterisk Module Deprecation policy.
For most modules that interacted with app_macro,
this change is limited to no longer looking for
the current context from the macrocontext when set.
The following modules have additional impacts:
app_dial - no longer supports M^ connected/redirecting macro
app_minivm - samples written using macro will no longer work.
The sample needs to be re-written
app_queue - can no longer call a macro on the called party's
channel. Use gosub which is currently supported
ccss - no callback macro, gosub only
app_voicemail - no macro support
channel - remove macrocontext and priority, no connected
line or redirection macro options
options - stdexten is deprecated to gosub as the default
and only options
pbx - removed macrolock
pbx_dundi - no longer look for macro
snmp - removed macro context, exten, and priority

- ### chan_alsa: Remove deprecated module.
This module was deprecated in Asterisk 19
and is now being removed in accordance with
the Asterisk Module Deprecation policy.

- ### pbx_builtins: Remove deprecated and

[asterisk-dev] asterisk release 20.5.1

2023-12-14 Thread Asterisk Development Team

The Asterisk Development Team would like to announce security release  
Asterisk 20.5.1.

The release artifacts are available for immediate download at  
https://github.com/asterisk/asterisk/releases/tag/20.5.1
and
https://downloads.asterisk.org/pub/telephony/asterisk

The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside 
files](https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call 
initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries 
](https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using 
'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)


Change Log for Release asterisk-20.5.1


Links:


 - [Full 
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
  
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)  
 - 
[Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
  
 - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)  

Summary:


- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.

User Notes:



Upgrade Notes:



Closed Issues:


None

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[asterisk-dev] asterisk release 18.20.1

2023-12-14 Thread Asterisk Development Team

The Asterisk Development Team would like to announce security release  
Asterisk 18.20.1.

The release artifacts are available for immediate download at  
https://github.com/asterisk/asterisk/releases/tag/18.20.1
and
https://downloads.asterisk.org/pub/telephony/asterisk

The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside 
files](https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call 
initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries 
](https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using 
'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)


Change Log for Release asterisk-18.20.1


Links:


 - [Full 
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
  
 - [GitHub 
Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)  
 - 
[Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
  
 - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)  

Summary:


- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.

User Notes:



Upgrade Notes:



Closed Issues:


None

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[asterisk-dev] asterisk release 21.0.0

2023-10-18 Thread Asterisk Development Team

The Asterisk Development Team would like to announce
the release of asterisk-21.0.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/21.0.0
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!

Change Log for Release asterisk-21.0.0

Links:

- [Full
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)

- [GitHub
Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
-
[Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)

- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)

Summary:

- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files

User Notes:

- ### sig_analog: Add Called Subscriber Held capability.
Called Subscriber Held is now supported for analog
FXS channels, using the calledsubscriberheld option. This allows
a station user to go on hook when receiving an incoming call
and resume from another phone on the same line by going on hook,
without disconnecting the call.

- ### res_pjsip_header_funcs: Make prefix argument optional.
The prefix argument to PJSIP_HEADERS is now
optional. If not specified, all header names will be
returned.

Upgrade Notes:

- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
ast_gethostbyname() has been deprecated and will be removed
in Asterisk 23. New code should use `ast_sockaddr_resolve()` and
`ast_sockaddr_resolve_first_af()`.

- ### app_sla: Migrate SLA applications out of app_meetme.
The SLAStation and SLATrunk applications have been moved
from app_meetme to app_sla. If you are using these applications and have
autoload=no, you will need to explicitly load this module in modules.conf.

- ### users.conf: Deprecate users.conf configuration.
The users.conf config is now deprecated
and will be removed in a future version of Asterisk.

- ### res_monitor: Remove deprecated module.
This module was deprecated in Asterisk 16
and is now being removed in accordance with
the Asterisk Module Deprecation policy.

[asterisk-dev] asterisk release 20.5.0

2023-10-18 Thread Asterisk Development Team

The Asterisk Development Team would like to announce
the release of asterisk-20.5.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/20.5.0
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!

Change Log for Release asterisk-20.5.0

Links:

- [Full
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)

- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
-
[Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)

- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)

Summary:

- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
particular mailbox folder. The forward command can be used to copy a message
within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp->themssrc_valid`
into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.

User Notes:

- ### res_pjsip_header_funcs: Make prefix argument optional.
The prefix argument to PJSIP_HEADERS is now
optional. If not specified, all header names will be
returned.

- ### core/ari/pjsip: Add refer mechanism
There is a new ARI endpoint `/endpoints/refer` for referring
an endpoint to some URI or endpoint.

- ### chan_dahdi: Allow autoreoriginating after hangup.
The autoreoriginate setting now allows for kewlstart FXS
channels to automatically reoriginate and provide dial tone to the
user again after all calls on the line have cleared. This saves users
from having to manually hang up and pick up the receiver again before
making another call.

- ### sig_analog: Allow three-way flash to time out to silence.
The threewaysilenthold option now allows the three-way
dial tone to time out to silence, rather than continuing forever.

- ### res_pjsip: Enable TLS v1.3 if present.
res_pjsip now allows TLS v1.3 to be enabled if supported by
the underlying PJSIP library. The bundled version of PJSIP supports
TLS v1.3.

- ### app_queue: Add support for app

[asterisk-dev] asterisk release 18.20.0

2023-10-18 Thread Asterisk Development Team

The Asterisk Development Team would like to announce
the release of asterisk-18.20.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/18.20.0
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!

Change Log for Release asterisk-18.20.0

Links:

- [Full
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)

- [GitHub
Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
-
[Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)

- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)

Summary:

User Notes:

- ### res_pjsip_header_funcs: Make prefix argument optional.
The prefix argument to PJSIP_HEADERS is now
optional. If not specified, all header names will be
returned.

- ### core/ari/pjsip: Add refer mechanism
There is a new ARI endpoint `/endpoints/refer` for referring
an endpoint to some URI or endpoint.

- ### sig_analog: Allow three-way flash to time out to silence.
The threewaysilenthold option now allows the three-way
dial tone to time out to silence, rather than continuing forever.

- ### res_pjsip: Enable TLS v1.3 if present.
res_pjsip now allows TLS v1.3 to be enabled if supported by
the underlying PJSIP library. The bundled version of PJSIP supports
TLS v1.3.

- ### app_queue: Add support for app

[asterisk-dev] asterisk release 21.0.0-rc1

2023-09-06 Thread Asterisk Development Team

The Asterisk Development Team would like to announce
release candidate 1 of asterisk-21.0.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/21.0.0-rc1
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!

Change Log for Release asterisk-21.0.0-rc1

Links:

- [Full
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0-rc1.md)

- [GitHub
Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0-rc1)
-
[Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0-rc1.tar.gz)

- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)

Summary:

- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- Update config.yml
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files

User Notes:

- ### res_pjsip_header_funcs: Make prefix argument optional.
The prefix argument to PJSIP_HEADERS is now
optional. If not specified, all header names will be
returned.

Upgrade Notes:

- ### users.conf: Deprecate users.conf configuration.
The users.conf config is now deprecated
and will be removed in a future version of Asterisk.

- ### app_osplookup: Remove deprecated module.
This module was deprecated in Asterisk 19
and is now being removed in accordance with
the Asterisk Module Deprecation policy.

- ### res_monit

[asterisk-dev] asterisk release 20.5.0-rc1

2023-09-06 Thread Asterisk Development Team

The Asterisk Development Team would like to announce
release candidate 1 of asterisk-20.5.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/20.5.0-rc1
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!

Change Log for Release asterisk-20.5.0-rc1

Links:

- [Full
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0-rc1.md)

- [GitHub
Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0-rc1)
-
[Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0-rc1.tar.gz)

- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)

Summary:

User Notes:

- ### res_pjsip_header_funcs: Make prefix argument optional.
The prefix argument to PJSIP_HEADERS is now
optional. If not specified, all header names will be
returned.

- ### core/ari/pjsip: Add refer mechanism
There is a new ARI endpoint `/endpoints/refer` for referring
an endpoint to some URI or endpoint.

- ### sig_analog: Allow three-way flash to time out to silence.
The threewaysilenthold option now allows the three-way
dial tone to time out to silence, rather than continuing forever.

- ### res_pjsip: Enable TLS v1.3 if present.
res_pjsip now allows TLS v1.3 to be enabled if supported by
the underlying PJSIP library. The bundled version of PJSIP supports
TLS v1.3.

- ### app_que

[asterisk-dev] asterisk release 18.20.0-rc1

2023-09-06 Thread Asterisk Development Team

The Asterisk Development Team would like to announce
release candidate 1 of asterisk-18.20.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/18.20.0-rc1
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!

Change Log for Release asterisk-18.20.0-rc1

Links:

- [Full
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0-rc1.md)

- [GitHub
Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0-rc1)
-
[Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0-rc1.tar.gz)

- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)

Summary:

User Notes:

- ### res_pjsip_header_funcs: Make prefix argument optional.
The prefix argument to PJSIP_HEADERS is now
optional. If not specified, all header names will be
returned.

- ### core/ari/pjsip: Add refer mechanism
There is a new ARI endpoint `/endpoints/refer` for referring
an endpoint to some URI or endpoint.

- ### sig_analog: Allow three-way flash to time out to silence.
The threewaysilenthold option now allows the three-way
dial tone to time out to silence, rather than continuing forever.

- ### res_pjsip: Enable TLS v1.3 if present.
res_pjsip now allows TLS v1.3 to be enabled if supported by
the underlying PJSIP library. The bundled version of PJSIP supports
TLS v1.3.

- ##

[asterisk-dev] libpri release 1.6.1

2023-08-16 Thread Asterisk Development Team

The Asterisk Development Team would like to announce  
the release of libpri-1.6.1.

The release artifacts are available for immediate download at  
https://github.com/asterisk/libpri/releases/tag/1.6.1
and
https://downloads.asterisk.org/pub/telephony/libpri

This release resolves issues reported by the community  
and would have not been possible without your participation.

Thank You!


Change Log for Release libpri-1.6.1


Links:


 - [Full 
ChangeLog](https://downloads.asterisk.org/pub/telephony/libpri/releases/ChangeLog-1.6.1.md)
  
 - [GitHub Diff](https://github.com/asterisk/libpri/compare/1.6.0...1.6.1)  
 - 
[Tarball](https://downloads.asterisk.org/pub/telephony/libpri/libpri-1.6.1.tar.gz)
  
 - [Downloads](https://downloads.asterisk.org/pub/telephony/libpri)  

Summary:


- .github: Add Releaser workflow
- Link README to README.md
- Makefile: Fix modern compiler errors.
- Makefile: Add the ability to build libpri on MacOS for Linux target.
- q931.c: Fix subaddress finding octet 4.

User Notes:



Upgrade Notes:



Closed Issues:


None

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[asterisk-dev] Asterisk Release 18.19.0

2023-07-20 Thread Asterisk Development Team

The Asterisk Development Team would like to announce
the release of Asterisk 18.19.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/18.19.0
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!

Change Log for Release 18.19.0

Links:

- [Full
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)

- [GitHub
Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
-
[Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)

- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)

Summary:

- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.

User Notes:

- ### AMI: Add parking position parameter to Park action
New ParkingSpace parameter has been added to AMI action Park.

- ### res_musiconhold: Add option to loop last file.
The loop_last option in musiconhold.conf now
allows the last file in the directory to be looped once reached.

- ### AMI: Add CoreShowChannelMap action.
New AMI action CoreShowChannelMap has been added.

- ### sig_analog: Add fuller Caller ID support.
Additional Caller ID properties are now supported on
incoming calls to FXS stations, namely the
redirecting reason and call qualifier.

- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
When creating a bridge using the ARI the 'type' argument now
accept

[asterisk-dev] Asterisk Release 20.4.0

2023-07-20 Thread Asterisk Development Team

The Asterisk Development Team would like to announce
the release of Asterisk 20.4.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/20.4.0
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!

Change Log for Release 20.4.0

Links:

- [Full
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)

- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
-
[Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)

- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)

Summary:

- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.

User Notes:

- ### AMI: Add parking position parameter to Park action
New ParkingSpace parameter has been added to AMI action Park.

- ### res_musiconhold: Add option to loop last file.
The loop_last option in musiconhold.conf now
allows the last file in the directory to be looped once reached.

- ### AMI: Add CoreShowChannelMap action.
New AMI action CoreShowChannelMap has been added.

- ### sig_analog: Add fuller Caller ID support.
Additional Caller ID properties are now supported on
incoming calls to FXS stations, namely the
redirecting reason and call qualifier.

- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
When creating a bridge using the ARI the 'type&#

[asterisk-dev] Asterisk Release 20.4.0-rc2

2023-07-13 Thread Asterisk Development Team

The Asterisk Development Team would like to announce  
release candidate 2 of Asterisk 20.4.0.

The release artifacts are available for immediate download at  
https://github.com/asterisk/asterisk/releases/tag/20.4.0-rc2
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community  
and would have not been possible without your participation.

Thank You!


Change Log for Release 20.4.0-rc2


Links:


 - [Full 
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0-rc2.md)
  
 - [GitHub 
Diff](https://github.com/asterisk/asterisk/compare/20.4.0-rc1...20.4.0-rc2)  
 - 
[Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0-rc2.tar.gz)
  
 - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)  

Summary:


- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging

User Notes:



Upgrade Notes:



Closed Issues:


  - #200: [bug]: Regression: In app.h an enum is used before its declaration.

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[asterisk-dev] Asterisk Release 18.19.0-rc2

2023-07-13 Thread Asterisk Development Team

The Asterisk Development Team would like to announce  
release candidate 2 of Asterisk 18.19.0.

The release artifacts are available for immediate download at  
https://github.com/asterisk/asterisk/releases/tag/18.19.0-rc2
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community  
and would have not been possible without your participation.

Thank You!


Change Log for Release 18.19.0-rc2


Links:


 - [Full 
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0-rc2.md)
  
 - [GitHub 
Diff](https://github.com/asterisk/asterisk/compare/18.19.0-rc1...18.19.0-rc2)  
 - 
[Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0-rc2.tar.gz)
  
 - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)  

Summary:


- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging

User Notes:



Upgrade Notes:



Closed Issues:


  - #200: [bug]: Regression: In app.h an enum is used before its declaration.

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[asterisk-dev] Asterisk Release 20.3.1

2023-07-07 Thread Asterisk Development Team

The Asterisk Development Team would like to announce security release
Asterisk 20.3.1.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/20.3.1
and
https://downloads.asterisk.org/pub/telephony/asterisk

The following security advisories were resolved in this release:
https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm

Change Log for Release 20.3.1

Links:

- [Full
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)

- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
-
[Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)

- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)

Summary:

- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1

User Notes:

- ### res_http_media_cache: Introduce options and customize
The res_http_media_cache module now attempts to load
configuration from the res_http_media_cache.conf file.
The following options were added:
* timeout_secs
* user_agent
* follow_location
* max_redirects
* protocols
* redirect_protocols
* dns_cache_timeout_secs

- ### format_sln: add .slin as supported file extension
format_sln now recognizes '.slin' as a valid
file extension in addition to the existing
'.sln' and '.raw'.

- ### app_senddtmf: Add SendFlash AMI action.
The SendFlash AMI action now allows sending
a hook flash event on a channel.

- ### pbx_dundi: Add PJSIP support.
DUNDi now supports chan_pjsip. Outgoing calls using
PJSIP require the pjsip_outgoing_endpoint option
to be set in dundi.conf.

Upgrade Notes:

Closed Issues:

- #193: [bug]: third-party/apply-patches doesn't sort the patch file list
before applying

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[asterisk-dev] Asterisk Release certified-18.9-cert5

2023-07-07 Thread Asterisk Development Team

The Asterisk Development Team would like to announce security release  
Certified Asterisk 18.9-cert5.

The release artifacts are available for immediate download at  
https://github.com/asterisk/asterisk/releases/tag/certified-18.9-cert5
and
https://downloads.asterisk.org/pub/telephony/certified-asterisk

The following security advisories were resolved in this release:
https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm


Change Log for Release certified-18.9-cert5


Links:


 - [Full 
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-certified-18.9-cert5.md)
  
 - [GitHub 
Diff](https://github.com/asterisk/asterisk/compare/certified-18.9-cert4...certified-18.9-cert5)
  
 - 
[Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-certified-18.9-cert5.tar.gz)
  
 - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)  

Summary:


- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- bundled_pjproject: Backport security fixes from pjproject 2.13.1
- .github: Updates for AsteriskReleaser
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- res_pjsip_session: Added new function calls to avoid ABI issues.
- test_statis_endpoints:  Fix channel_messages test again
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- AMI: Add CoreShowChannelMap action.
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- .github: Change title of AsteriskReleaser job
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- core: Cleanup gerrit and JIRA references. (#40) (#61)
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
- .github: Add AsteriskReleaser
- cel: add local optimization begin event
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- .github: Add cherry-pick test progress labels
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- test.c: Fix counting of tests and add 2 new tests
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- bridge_builtin_features: add beep via touch variable
- cli: increase channel column width
- app_senddtmf: Add option to answer target channel.
- app_directory: Add a 'skip call' option.
- app_read: Add an option to return terminator on empty digits.
- app_directory: add ability to specify configuration file

User Notes:


- ### AMI: Add CoreShowChannelMap action.
  New AMI action CoreShowChannelMap has been added.

- ### cel: add local optimization begin event
  The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used
  by itself or in conert with the existing
  AST_CEL_LOCAL_OPTIMIZE to book-end local channel optimizaion.

- ### app_read: Add an option to return terminator on empty digits.
  A new option 'e' has been added to allow Read() to return the
  terminator as the dialed digits in the case where only the terminator
  is entered.

- ### format_sln: add .slin as supported file extension
  format_sln now recognizes '.slin' as a valid
  file extension in addition to the existing
  '.sln' and '.raw'.

- ### bridge_builtin_features: add beep via touch variable
  Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
  Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid
  interval in seconds will result in a periodic beep being
  played to the monitored channel upon MixMontior/Monitor
  feature start.
  If an interval less than 5 seconds is specified, the interval
  will default to 5 seconds.  If the value is set to an invalid
  interval, the default of 15 seconds will be used.

- ### app_directory: Add a

[asterisk-dev] Asterisk Release 19.8.1

2023-07-07 Thread Asterisk Development Team

The Asterisk Development Team would like to announce security release  
Asterisk 19.8.1.

The release artifacts are available for immediate download at  
https://github.com/asterisk/asterisk/releases/tag/19.8.1
and
https://downloads.asterisk.org/pub/telephony/asterisk

The following security advisories were resolved in this release:
https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm


Change Log for Release 19.8.1


Links:


 - [Full 
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.8.1.md)
  
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/19.8.0...19.8.1)  
 - 
[Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-19.8.1.tar.gz)
  
 - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)  

Summary:


- apply_patches: Use globbing instead of file/sort.
- bundled_pjproject: Backport 2 SSL patches from upstream
- bundled_pjproject: Backport security fixes from pjproject 2.13.1
- apply_patches: Sort patch list before applying

User Notes:



Upgrade Notes:



Closed Issues:


  - #188: [improvement]:  pjsip: Upgrade bundled version to pjproject 2.13.1 
#187 
  - #193: [bug]: third-party/apply-patches doesn't sort the patch file list 
before applying
  - #194: [bug]: Segfault/double-free in bundled pjproject using TLS transport

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[asterisk-dev] Asterisk Release 18.18.1

2023-07-07 Thread Asterisk Development Team

The Asterisk Development Team would like to announce security release
Asterisk 18.18.1.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/18.18.1
and
https://downloads.asterisk.org/pub/telephony/asterisk

The following security advisories were resolved in this release:
https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm

Change Log for Release 18.18.1

Links:

- [Full
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)

- [GitHub
Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
-
[Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)

- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)

Summary:

- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1

User Notes:

- ### format_sln: add .slin as supported file extension
format_sln now recognizes '.slin' as a valid
file extension in addition to the existing
'.sln' and '.raw'.

- ### app_senddtmf: Add SendFlash AMI action.
The SendFlash AMI action now allows sending
a hook flash event on a channel.

- ### pbx_dundi: Add PJSIP support.
DUNDi now supports chan_pjsip. Outgoing calls using
PJSIP require the pjsip_outgoing_endpoint option
to be set in dundi.conf.

Upgrade Notes:

Closed Issues:

- #193: [bug]: third-party/apply-patches doesn't sort the patch file list
before applying

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[asterisk-dev] Asterisk Release 16.30.1

2023-07-07 Thread Asterisk Development Team

The Asterisk Development Team would like to announce security release  
Asterisk 16.30.1.

The release artifacts are available for immediate download at  
https://github.com/asterisk/asterisk/releases/tag/16.30.1
and
https://downloads.asterisk.org/pub/telephony/asterisk

The following security advisories were resolved in this release:
https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm


Change Log for Release 16.30.1


Links:


 - [Full 
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.30.1.md)
  
 - [GitHub 
Diff](https://github.com/asterisk/asterisk/compare/16.30.0...16.30.1)  
 - 
[Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-16.30.1.tar.gz)
  
 - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)  

Summary:


- apply_patches: Use globbing instead of file/sort.
- bundled_pjproject: Backport 2 SSL patches from upstream
- bundled_pjproject: Backport security fixes from pjproject 2.13.1
- apply_patches: Sort patch list before applying

User Notes:



Upgrade Notes:



Closed Issues:


  - #188: [improvement]:  pjsip: Upgrade bundled version to pjproject 2.13.1 
#187 
  - #193: [bug]: third-party/apply-patches doesn't sort the patch file list 
before applying
  - #194: [bug]: Segfault/double-free in bundled pjproject using TLS transport

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[asterisk-dev] Asterisk Release 20.4.0-rc1

2023-06-30 Thread Asterisk Development Team

The Asterisk Development Team would like to announce
release candidate 1 of Asterisk 20.4.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/20.4.0-rc1
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!

Change Log for Release 20.4.0-rc1

Links:

- [Full
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0-rc1.md)

- [GitHub
Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.4.0-rc1)
-
[Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0-rc1.tar.gz)

- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)

Summary:

- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.

User Notes:

- ### AMI: Add parking position parameter to Park action
New ParkingSpace parameter has been added to AMI action Park.

- ### res_musiconhold: Add option to loop last file.
The loop_last option in musiconhold.conf now
allows the last file in the directory to be looped once reached.

- ### AMI: Add CoreShowChannelMap action.
New AMI action CoreShowChannelMap has been added.

- ### sig_analog: Add fuller Caller ID support.
Additional Caller ID properties are now supported on
incoming calls to FXS stations, namely the
redirecting reason and call qualifier.

- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
When creating a bridge using the ARI the 'type' argument now
accepts a new value 'sdp_label' which will configure the bridge to ad

[asterisk-dev] Asterisk Release 18.19.0-rc1

2023-06-30 Thread Asterisk Development Team

The Asterisk Development Team would like to announce
release candidate 1 of Asterisk 18.19.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/18.19.0-rc1
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!

Change Log for Release 18.19.0-rc1

Links:

- [Full
ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0-rc1.md)

- [GitHub
Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.19.0-rc1)
-
[Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0-rc1.tar.gz)

- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)

Summary:

- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.

User Notes:

- ### AMI: Add parking position parameter to Park action
New ParkingSpace parameter has been added to AMI action Park.

- ### res_musiconhold: Add option to loop last file.
The loop_last option in musiconhold.conf now
allows the last file in the directory to be looped once reached.

- ### AMI: Add CoreShowChannelMap action.
New AMI action CoreShowChannelMap has been added.

- ### sig_analog: Add fuller Caller ID support.
Additional Caller ID properties are now supported on
incoming calls to FXS stations, namely the
redirecting reason and call qualifier.

- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
When creating a bridge using the ARI the 'type' argument now
accepts a new value 'sdp_label' which will configure the bridge to add
labels for each stream in

[asterisk-dev] Asterisk Release 20.3.0

2023-05-23 Thread Asterisk Development Team

The Asterisk Development Team would like to announce  
the release of Asterisk 20.3.0.

The release artifacts are available for immediate download at  
https://github.com/asterisk/asterisk/releases/tag/20.3.0
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community  
and would have not been possible without your participation.

Thank You!


Change Log for Release 20.3.0


Summary:


- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.

User Notes:


- ### cel: add local optimization begin event
  The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used
  by itself or in conert with the existing
  AST_CEL_LOCAL_OPTIMIZE to book-end local channel optimizaion.

- ### chan_dahdi: Add dialmode option for FXS lines.
  A "dialmode" option has been added which allows
  specifying, on a per-channel basis, what methods of
  subscriber dialing (pulse and/or tone) are permitted.
  Additionally, this can be changed on a channel
  at any point during a call using the CHANNEL
  function.

- ### pbx_dundi: Add PJSIP support.
  DUNDi now supports chan_pjsip. Outgoing calls using
  PJSIP require the pjsip_outgoing_endpoint option
  to be set in dundi.conf.

- ### cli: increase channel column width
  This change increases the display width on 'core show channels'
  amd 'core show channels verbose'
  For 'core show channels', the Channel name field is increased to
  64 characters and the Location name field is increased to 32
  characters.
  For 'core show channels verbose', the Channel name field is
  increased to 80 characters, the Context is increased to 24
  characters and the Extension is increased to 24 characters.

- ### app_senddtmf: Add SendFlash AMI action.
  The SendFlash AMI action now allows sending
  a hook flash event on a channel.

- ### res_http_media_cache: Introduce options and customize
  The res_http_media_cache module now attempts to load
  configuration from the res_http_media_cache.conf file.
  The following options were added:
* timeout_secs
* user_agent
* follow_location
* max_redirects
* protocols
* redirect_protocols
* dns_cache_timeout_secs

- ### test.c: Fix counting of tests and add 2 new tests
  The "tests" attribute of the "testsuite" element in the
  output XML now reflects only the tests actually requested
  to be executed instead of all the tests registered.
  The "failures" attribute was added to the "testsuite"
  element.
  Also added two new unit tests that just pass and fail
  to be used for testing CI itself.

- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
  It is now possible to specify the MixMoni

[asterisk-dev] Asterisk Release 18.18.0

2023-05-23 Thread Asterisk Development Team

The Asterisk Development Team would like to announce  
the release of Asterisk 18.18.0.

The release artifacts are available for immediate download at  
https://github.com/asterisk/asterisk/releases/tag/18.18.0
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community  
and would have not been possible without your participation.

Thank You!


Change Log for Release 18.18.0


Summary:


- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize

User Notes:


- ### cel: add local optimization begin event
  The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used
  by itself or in conert with the existing
  AST_CEL_LOCAL_OPTIMIZE to book-end local channel optimizaion.

- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
  A "dialmode" option has been added which allows
  specifying, on a per-channel basis, what methods of
  subscriber dialing (pulse and/or tone) are permitted.
  Additionally, this can be changed on a channel
  at any point during a call using the CHANNEL
  function.

- ### pbx_dundi: Add PJSIP support.
  DUNDi now supports chan_pjsip. Outgoing calls using
  PJSIP require the pjsip_outgoing_endpoint option
  to be set in dundi.conf.

- ### cli: increase channel column width
  This change increases the display width on 'core show channels'
  amd 'core show channels verbose'
  For 'core show channels', the Channel name field is increased to
  64 characters and the Location name field is increased to 32
  characters.
  For 'core show channels verbose', the Channel name field is
  increased to 80 characters, the Context is increased to 24
  characters and the Extension is increased to 24 characters.

- ### app_senddtmf: Add SendFlash AMI action.
  The SendFlash AMI action now allows sending
  a hook flash event on a channel.

- ### res_http_media_cache: Introduce options and customize
  The res_http_media_cache module now attempts to load
  configuration from the res_http_media_cache.conf file.
  The following options were added:
* timeout_secs
* user_agent
* follow_location
* max_redirects
* protocols
* redirect_protocols
* dns_cache_timeout_secs

- ### test.c: Fix counting of tests and add 2 new tests
  The "tests" attribute of the "testsuite" element in the
  output XML now reflects only the tests actually requested
  to be executed instead of all the tests registered.
  The "failures" attribute was added to the "testsuite"
  element.
  Also added two new unit tests that just pass and fail
  to be used for testing CI itself.

- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
  It is now possibl

[asterisk-dev] Test HTML version of...Asterisk Release 20.3.0-rc1

2023-05-18 Thread Asterisk Development Team

The Asterisk Development Team would like to announce
release candidate 1 of Asterisk 20.3.0.
The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/20.3.0-rc1
and
https://downloads.asterisk.org/pub/telephony/asterisk
This release resolves issues reported by the community
and would have not been possible without your participation.
Thank You!
Change Log for Release 20.3.0-rc1
Summary:

Set up new ChangeLogs directory
.github: Add AsteriskReleaser
chan_pjsip: also return all codecs on empty re-INVITE for late offers
cel: add local optimization begin event
core: Cleanup gerrit and JIRA references. (#57)
.github: Fix CherryPickTest to only run when it should
.github: Fix reference to CHERRYPICKTESTINGINPROGRESS
.github: Remove separate set labels step from new PR
.github: Refactor CP progress and add new PR test progress
res_pjsip: mediasec: Add Security-Client headers after 401
.github: Add cherry-pick test progress labels
LICENSE: Update link to trademark policy.
chan_dahdi: Add dialmode option for FXS lines.
.github: Update issue templates
.github: Remove unnecessary parameter in CherryPickTest
Initial GitHub PRs
Initial GitHub Issue Templates
pbx_dundi: Fix PJSIP endpoint configuration check.
Revert "app_queue: periodic announcement configurable start time."
respjsipstir_shaken: Fix JSON field ordering and disallowed TN characters.
pbx_dundi: Add PJSIP support.
install_prereq: Add Linux Mint support.
chan_pjsip: fix music on hold continues after INVITE with replaces
voicemail.conf: Fix incorrect comment about #include.
app_queue: Fix minor xmldoc duplication and vagueness.
test.c: Fix counting of tests and add 2 new tests
res_calendar: output busy state as part of show calendar.
loader.c: Minor module key check simplification.
ael: Regenerate lexers and parsers.
bridgebuiltinfeatures: add beep via touch variable
res_mixmonitor: MixMonitorMute by MixMonitor ID
format_sln: add .slin as supported file extension
res_agi: RECORD FILE plays 2 beeps.
func_json: Fix JSON parsing issues.
app_senddtmf: Add SendFlash AMI action.
app_dial: Fix DTMF not relayed to caller on unanswered calls.
configure: fix detection of re-entrant resolver functions
cli: increase channel column width
app_queue: periodic announcement configurable start time.
make_version: Strip svn stuff and suppress ref HEAD errors
reshttpmedia_cache: Introduce options and customize
main/iostream.c: fix build with libressl
contrib: rc.archlinux.asterisk uses invalid redirect.

User Notes:

cel: add local optimization begin event
The new ASTCELLOCALOPTIMIZEBEGIN can be used
by itself or in conert with the existing
ASTCELLOCAL_OPTIMIZE to book-end local channel optimizaion.
chan_dahdi: Add dialmode option for FXS lines.
A "dialmode" option has been added which allows
specifying, on a per-channel basis, what methods of
subscriber dialing (pulse and/or tone) are permitted.
Additionally, this can be changed on a channel
at any point during a call using the CHANNEL
function.
app_senddtmf: Add SendFlash AMI action.
The SendFlash AMI action now allows sending
a hook flash event on a channel.
res_mixmonitor: MixMonitorMute by MixMonitor ID
It is now possible to specify the MixMonitorID when calling
the manager action: MixMonitorMute.  This will allow an
individual MixMonitor instance to be muted via ID.
The MixMonitorID can be stored as a channel variable using
the 'i' MixMonitor option and is returned upon creation if
this option is used.
As part of this change, if no MixMonitorID is specified in
the manager action MixMonitorMute, Asterisk will set the mute
flag on all MixMonitor audiohooks on the channel.  Previous
behavior would set the flag on the first MixMonitor audiohook
found.
bridgebuiltinfeatures: add beep via touch variable
Add optional touch variable : TOUCHMIXMONITORBEEP(interval)
Setting TOUCHMIXMONITORBEEP/TOUCHMONITORBEEP to a valid
interval in seconds will result in a periodic beep being
played to the monitored channel upon MixMontior/Monitor
feature start.
If an interval less than 5 seconds is specified, the interval
will default to 5 seconds.  If the value is set to an invalid
interval, the default of 15 seconds will be used.
cli: increase channel column width
This change increases the display width on 'core show channels'
amd 'core show channels verbose'
For 'core show channels', the Channel name field is increased to
64 characters and the Location name field is increased to 32
characters.
For 'core show channels verbose', the Channel name field is
increased to 80 characters, the Context is increased to 24
characters and the Extension is increased to 24 characters.
pbx_dundi: Add PJSIP support.
DUNDi now supports chanpjsip. Outgoing calls using
PJSIP require the pjsipoutgoing_endpoint option
to be set in dundi.conf.
format_sln: add .slin as supported file extension
format_sln now recognizes '.s

[asterisk-dev] Asterisk Release 20.3.0-rc1

2023-05-18 Thread Asterisk Development Team

The Asterisk Development Team would like to announce  
release candidate 1 of Asterisk 20.3.0.

The release artifacts are available for immediate download at  
https://github.com/asterisk/asterisk/releases/tag/20.3.0-rc1
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community  
and would have not been possible without your participation.

Thank You!


Change Log for Release 20.3.0-rc1


Summary:


- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.

User Notes:


- ### cel: add local optimization begin event
  The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used
  by itself or in conert with the existing
  AST_CEL_LOCAL_OPTIMIZE to book-end local channel optimizaion.

- ### chan_dahdi: Add dialmode option for FXS lines.
  A "dialmode" option has been added which allows
  specifying, on a per-channel basis, what methods of
  subscriber dialing (pulse and/or tone) are permitted.
  Additionally, this can be changed on a channel
  at any point during a call using the CHANNEL
  function.

- ### app_senddtmf: Add SendFlash AMI action.
  The SendFlash AMI action now allows sending
  a hook flash event on a channel.

- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
  It is now possible to specify the MixMonitorID when calling
  the manager action: MixMonitorMute.  This will allow an
  individual MixMonitor instance to be muted via ID.
  The MixMonitorID can be stored as a channel variable using
  the 'i' MixMonitor option and is returned upon creation if
  this option is used.
  As part of this change, if no MixMonitorID is specified in
  the manager action MixMonitorMute, Asterisk will set the mute
  flag on all MixMonitor audiohooks on the channel.  Previous
  behavior would set the flag on the first MixMonitor audiohook
  found.

- ### bridge_builtin_features: add beep via touch variable
  Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
  Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid
  interval in seconds will result in a periodic beep being
  played to the monitored channel upon MixMontior/Monitor
  feature start.
  If an interval less than 5 seconds is specified, the interval
  will default to 5 seconds.  If the value is set to an invalid
  interval, the default of 15 seconds will be used.

- ### cli: increase channel column width
  This change increases the display width on 'core show channels'
  amd 'core show channels verbose'
  For 'core show channels', the Channel name field is increased to
  64 characters and the Location name field is increased to 32
  characters.
  For 'core show channels verbose', the Channel name field is
  increased

[asterisk-dev] Asterisk Release 18.18.0-rc1

2023-05-18 Thread Asterisk Development Team

The Asterisk Development Team would like to announce  
release candidate 1 of Asterisk 18.18.0.

The release artifacts are available for immediate download at  
https://github.com/asterisk/asterisk/releases/tag/18.18.0-rc1
and
https://downloads.asterisk.org/pub/telephony/asterisk

This release resolves issues reported by the community  
and would have not been possible without your participation.

Thank You!


Change Log for Release 18.18.0-rc1


Summary:


- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize

User Notes:


- ### cel: add local optimization begin event
  The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used
  by itself or in conert with the existing
  AST_CEL_LOCAL_OPTIMIZE to book-end local channel optimizaion.

- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
  A "dialmode" option has been added which allows
  specifying, on a per-channel basis, what methods of
  subscriber dialing (pulse and/or tone) are permitted.
  Additionally, this can be changed on a channel
  at any point during a call using the CHANNEL
  function.

- ### app_senddtmf: Add SendFlash AMI action.
  The SendFlash AMI action now allows sending
  a hook flash event on a channel.

- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
  It is now possible to specify the MixMonitorID when calling
  the manager action: MixMonitorMute.  This will allow an
  individual MixMonitor instance to be muted via ID.
  The MixMonitorID can be stored as a channel variable using
  the 'i' MixMonitor option and is returned upon creation if
  this option is used.
  As part of this change, if no MixMonitorID is specified in
  the manager action MixMonitorMute, Asterisk will set the mute
  flag on all MixMonitor audiohooks on the channel.  Previous
  behavior would set the flag on the first MixMonitor audiohook
  found.

- ### bridge_builtin_features: add beep via touch variable
  Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
  Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid
  interval in seconds will result in a periodic beep being
  played to the monitored channel upon MixMontior/Monitor
  feature start.
  If an interval less than 5 seconds is specified, the interval
  will default to 5 seconds.  If the value is set to an invalid
  interval, the default of 15 seconds will be used.

- ### cli: increase channel column width
  This change increases the display width on 'core show channels'
  amd 'core show channels verbose'
  For 'core show channels', the Channel name field is increased to
  64 characters and the Location name field is increased to 32
  characters.
  For 'core show channels verbose', the Channel name

[asterisk-dev] Test Email 2

2023-05-18 Thread Asterisk Development Team

Test email hopefully without the phishing warning

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[asterisk-dev] Test Email

2023-05-18 Thread Asterisk Development Team

Test email

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[asterisk-dev] Asterisk issue reporting is now live on GitHub

2023-04-28 Thread Asterisk Development Team

All Asterisk issues should now be reported at
https://github.com/asterisk/asterisk/issues

The previous issue system at https://issues.asterisk.org remains in
read-only mode for reference but will eventually be replaced with a
searchable archive.
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[asterisk-dev] Reminder: Issues and Code Contribution move to GitHub

2023-04-27 Thread Asterisk Development Team

Issues and Code Contribution are moving to GitHub this weekend!!

Both issues.asterisk.org and gerrit.asterisk.org will be going read-only at
noon EDT (UTC-4:00) Friday April 28th.Within a few hours, the
capability to create issues in GitHub at
https://github.com/asterisk/asterisk should be available.   The ability to
accept pull requests may not be available until Monday morning because we
have to make sure the repositories are in sync and get workflows merged
into the appropriate branches.

We'll post status updates as things become available.
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[asterisk-dev] Asterisk 20.2.1 Now Available

2023-04-03 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
20.2.1.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 20.2.1 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Bugs fixed in this release:
---
 * ASTERISK-30469 - res_pjsip_pubsub: Regression for
  subscription shutdowns
  (Reported by N A)
 * ASTERISK-30472 - pbx_ael: Literal usage for variables broken

  (Reported by isrl)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1

Thank you for your continued support of Asterisk!
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[asterisk-dev] Asterisk 18.17.1 Now Available

2023-04-03 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
18.17.1.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.17.1 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Bugs fixed in this release:
---
 * ASTERISK-30469 - res_pjsip_pubsub: Regression for
  subscription shutdowns
  (Reported by N A)
 * ASTERISK-30472 - pbx_ael: Literal usage for variables broken

  (Reported by isrl)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1

Thank you for your continued support of Asterisk!
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[asterisk-dev] Asterisk 20.2.0 Now Available

2023-03-09 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
20.2.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 20.2.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

New Features made in this release:
---
 * ASTERISK-29810 - app_signal: Add channel signaling
  applications
  (Reported by N A)
 * ASTERISK-30262 - res_pjsip_session: Allow a context to be
  specified for overlap dialing
  (Reported by N A)
 * ASTERISK-30319 - Add BYE Reason support for SIP
 
  (Reported by Igor Goncharovsky)
 * ASTERISK-30180 - app_broadcast: Add a channel audio
  multicasting application
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-27830 - Asterisk crashes on Invalid UTF-8 string
   
  (Reported by AvayaXAsterisk)
 * ASTERISK-30354 - chan_iax2: Lack of formats prior to
  receiving voice frames causes jitterbuffer to stall
 
  (Reported by N A)
 * ASTERISK-30162 - when chan_iax is used to relay calls, no
  ringing indication is played
  (Reported by Jaco Kroon)
 * ASTERISK-30424 - pjproject_bundled: cross-compilation broken
  when ssl autodetected
  (Reported by Nick French)
 * ASTERISK-30388 - res_phoneprov: Stale SERVER variable when
  multi-homed
  (Reported by cmaj)
 * ASTERISK-30419 - pjsip: Crash when sending NOTIFY in PJSIP
  2.13
  (Reported by Ross Beer)
 * ASTERISK-30417 - Copy/Paste error in UnpauseQueueMember
 
  (Reported by Sean Bright)
 * ASTERISK-29604 - ari: Segfault with lots of calls
 
  (Reported by Danila Evgrafov)
 * ASTERISK-30406 - pbx_ael: Global variables are not expanded.

  (Reported by Sean Bright)
 * ASTERISK-30391 - res_rtp_asterisk: Issue with transcoding
  g722 after MES changes
  (Reported by George Joseph)
 * ASTERISK-30345 - loader.c: Modules that decline to load
  cannot be reloaded
  (Reported by N A)
 * ASTERISK-30351 - manager: Originate variables are not added
  when setvar used in manager.conf
  (Reported by Sebastian
  Gutierrez)
 * ASTERISK-30369 - res_pjsip: Websockets from same IP shut down
  when they shouldn't be
  (Reported by Joshua C. Colp)
 * ASTERISK-30379 - http: fix NULL pointer dereference while
  enable_status on TLS-only
  (Reported by Boris P. Korzun)
 * ASTERISK-30375 - res_http_media_cache: Crash when URL has no
  path component.
  (Reported by Sean Bright)
 * ASTERISK-30367 - pbx: Fix outdated channel snapshots with
  pbx_exec
  (Reported by N A)
 * ASTERISK-28767 - chan_pjsip: Caller ID not used when checking
  for extension, callerid supplement executed too late
 
  (Reported by Oleg)
 * ASTERISK-30350 - res_pjsip_sdp_rtp: rtp_timeout_hold is not
  used when moh_passthrough has call on hold
  (Reported by
  Benjamin Keith Ford)
 * ASTERISK-30240 - app voicemail odbc build error with gcc
  11.1
  (Reported by Michael Bradeen)
 * ASTERISK-30100 - res_pjsip: Path is ignored on INVITE to
  endpoint
  (Reported by Yury Kirsanov)
 * ASTERISK-30198 - Error `Too many open files` occurs after
  about ~8000 calls when using mixmonitor
  (Reported by
  Julien Alie)
 * ASTERISK-30347 - xmldocs: Remove references to removed
  applications
  (Reported by N A)

Improvements made in this release:
---
 * ASTERISK-30411 - app_read: add option to include terminating
  digit on empty, terminated strings
  (Reported by Michael
  Bradeen)
 * ASTERISK-30405 - app_directory: Add 's' option to skip
  channel call
  (Reported by Michael Bradeen)
 * ASTERISK-30422 - app_senddtmf: add the option for senddtmf to
  answer
  (Reported by Michael Bradeen)
 * ASTERISK-30325 - Upgrade Asterisk to bundled pjproject 2.13
 
  (Reported by Stanislav Abramenkov)
 * ASTERISK-30404 - app_directory: Add reading directory
  configuration from custom file
  (Reported by Michael
  Bradeen)
 * ASTERISK-29913 - func_json: Adds multi-level and array
  parsing to JSON_DECODE
  (Reported by N A)
 * ASTERISK-30353 - func_frame_trace: Print text for text
  frames
  (Reported by N A)
 * ASTERISK-30361 - json.h: Add missing
  ast_json_object_real_get
  (Reported by N A)
 * ASTERISK-30280 - Create capability to assign a Media
  Experience Score to RTP streams
  (Reported by George
  Joseph)
 * ASTERISK-30332 - func_callerid: Warn if invalid redirecting
  reason provided
  (Reported by N A)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0

Thank you for your conti

[asterisk-dev] Asterisk 18.17.0 Now Available

2023-03-09 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
18.17.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.17.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

New Features made in this release:
---
 * ASTERISK-29810 - app_signal: Add channel signaling
  applications
  (Reported by N A)
 * ASTERISK-30262 - res_pjsip_session: Allow a context to be
  specified for overlap dialing
  (Reported by N A)
 * ASTERISK-30319 - Add BYE Reason support for SIP
 
  (Reported by Igor Goncharovsky)
 * ASTERISK-30180 - app_broadcast: Add a channel audio
  multicasting application
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-27830 - Asterisk crashes on Invalid UTF-8 string
   
  (Reported by AvayaXAsterisk)
 * ASTERISK-30354 - chan_iax2: Lack of formats prior to
  receiving voice frames causes jitterbuffer to stall
 
  (Reported by N A)
 * ASTERISK-30162 - when chan_iax is used to relay calls, no
  ringing indication is played
  (Reported by Jaco Kroon)
 * ASTERISK-30424 - pjproject_bundled: cross-compilation broken
  when ssl autodetected
  (Reported by Nick French)
 * ASTERISK-30388 - res_phoneprov: Stale SERVER variable when
  multi-homed
  (Reported by cmaj)
 * ASTERISK-30419 - pjsip: Crash when sending NOTIFY in PJSIP
  2.13
  (Reported by Ross Beer)
 * ASTERISK-30417 - Copy/Paste error in UnpauseQueueMember
 
  (Reported by Sean Bright)
 * ASTERISK-30406 - pbx_ael: Global variables are not expanded.

  (Reported by Sean Bright)
 * ASTERISK-29604 - ari: Segfault with lots of calls
 
  (Reported by Danila Evgrafov)
 * ASTERISK-30391 - res_rtp_asterisk: Issue with transcoding
  g722 after MES changes
  (Reported by George Joseph)
 * ASTERISK-30345 - loader.c: Modules that decline to load
  cannot be reloaded
  (Reported by N A)
 * ASTERISK-30379 - http: fix NULL pointer dereference while
  enable_status on TLS-only
  (Reported by Boris P. Korzun)
 * ASTERISK-30375 - res_http_media_cache: Crash when URL has no
  path component.
  (Reported by Sean Bright)
 * ASTERISK-30351 - manager: Originate variables are not added
  when setvar used in manager.conf
  (Reported by Sebastian
  Gutierrez)
 * ASTERISK-30369 - res_pjsip: Websockets from same IP shut down
  when they shouldn't be
  (Reported by Joshua C. Colp)
 * ASTERISK-30367 - pbx: Fix outdated channel snapshots with
  pbx_exec
  (Reported by N A)
 * ASTERISK-28767 - chan_pjsip: Caller ID not used when checking
  for extension, callerid supplement executed too late
 
  (Reported by Oleg)
 * ASTERISK-30350 - res_pjsip_sdp_rtp: rtp_timeout_hold is not
  used when moh_passthrough has call on hold
  (Reported by
  Benjamin Keith Ford)
 * ASTERISK-30240 - app voicemail odbc build error with gcc
  11.1
  (Reported by Michael Bradeen)
 * ASTERISK-30100 - res_pjsip: Path is ignored on INVITE to
  endpoint
  (Reported by Yury Kirsanov)
 * ASTERISK-30198 - Error `Too many open files` occurs after
  about ~8000 calls when using mixmonitor
  (Reported by
  Julien Alie)

Improvements made in this release:
---
 * ASTERISK-30411 - app_read: add option to include terminating
  digit on empty, terminated strings
  (Reported by Michael
  Bradeen)
 * ASTERISK-30405 - app_directory: Add 's' option to skip
  channel call
  (Reported by Michael Bradeen)
 * ASTERISK-30422 - app_senddtmf: add the option for senddtmf to
  answer
  (Reported by Michael Bradeen)
 * ASTERISK-30325 - Upgrade Asterisk to bundled pjproject 2.13
 
  (Reported by Stanislav Abramenkov)
 * ASTERISK-30404 - app_directory: Add reading directory
  configuration from custom file
  (Reported by Michael
  Bradeen)
 * ASTERISK-29913 - func_json: Adds multi-level and array
  parsing to JSON_DECODE
  (Reported by N A)
 * ASTERISK-30353 - func_frame_trace: Print text for text
  frames
  (Reported by N A)
 * ASTERISK-30361 - json.h: Add missing
  ast_json_object_real_get
  (Reported by N A)
 * ASTERISK-30280 - Create capability to assign a Media
  Experience Score to RTP streams
  (Reported by George
  Joseph)
 * ASTERISK-30332 - func_callerid: Warn if invalid redirecting
  reason provided
  (Reported by N A)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0

Thank you for your continued support of Asterisk!
-- 
___

[asterisk-dev] Asterisk 20.2.0-rc1 Now Available

2023-03-02 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the first
release candidate of Asterisk 20.2.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 20.2.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release candidate:

New Features made in this release:
---
 * ASTERISK-29810 - app_signal: Add channel signaling
  applications
  (Reported by N A)
 * ASTERISK-30262 - res_pjsip_session: Allow a context to be
  specified for overlap dialing
  (Reported by N A)
 * ASTERISK-30319 - Add BYE Reason support for SIP
 
  (Reported by Igor Goncharovsky)
 * ASTERISK-30180 - app_broadcast: Add a channel audio
  multicasting application
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-27830 - Asterisk crashes on Invalid UTF-8 string
   
  (Reported by AvayaXAsterisk)
 * ASTERISK-30354 - chan_iax2: Lack of formats prior to
  receiving voice frames causes jitterbuffer to stall
 
  (Reported by N A)
 * ASTERISK-30162 - when chan_iax is used to relay calls, no
  ringing indication is played
  (Reported by Jaco Kroon)
 * ASTERISK-30424 - pjproject_bundled: cross-compilation broken
  when ssl autodetected
  (Reported by Nick French)
 * ASTERISK-30388 - res_phoneprov: Stale SERVER variable when
  multi-homed
  (Reported by cmaj)
 * ASTERISK-30419 - pjsip: Crash when sending NOTIFY in PJSIP
  2.13
  (Reported by Ross Beer)
 * ASTERISK-30417 - Copy/Paste error in UnpauseQueueMember
 
  (Reported by Sean Bright)
 * ASTERISK-29604 - ari: Segfault with lots of calls
 
  (Reported by Danila Evgrafov)
 * ASTERISK-30406 - pbx_ael: Global variables are not expanded.

  (Reported by Sean Bright)
 * ASTERISK-30391 - res_rtp_asterisk: Issue with transcoding
  g722 after MES changes
  (Reported by George Joseph)
 * ASTERISK-30345 - loader.c: Modules that decline to load
  cannot be reloaded
  (Reported by N A)
 * ASTERISK-30351 - manager: Originate variables are not added
  when setvar used in manager.conf
  (Reported by Sebastian
  Gutierrez)
 * ASTERISK-30369 - res_pjsip: Websockets from same IP shut down
  when they shouldn't be
  (Reported by Joshua C. Colp)
 * ASTERISK-30379 - http: fix NULL pointer dereference while
  enable_status on TLS-only
  (Reported by Boris P. Korzun)
 * ASTERISK-30375 - res_http_media_cache: Crash when URL has no
  path component.
  (Reported by Sean Bright)
 * ASTERISK-30367 - pbx: Fix outdated channel snapshots with
  pbx_exec
  (Reported by N A)
 * ASTERISK-28767 - chan_pjsip: Caller ID not used when checking
  for extension, callerid supplement executed too late
 
  (Reported by Oleg)
 * ASTERISK-30350 - res_pjsip_sdp_rtp: rtp_timeout_hold is not
  used when moh_passthrough has call on hold
  (Reported by
  Benjamin Keith Ford)
 * ASTERISK-30240 - app voicemail odbc build error with gcc
  11.1
  (Reported by Michael Bradeen)
 * ASTERISK-30100 - res_pjsip: Path is ignored on INVITE to
  endpoint
  (Reported by Yury Kirsanov)
 * ASTERISK-30198 - Error `Too many open files` occurs after
  about ~8000 calls when using mixmonitor
  (Reported by
  Julien Alie)
 * ASTERISK-30347 - xmldocs: Remove references to removed
  applications
  (Reported by N A)

Improvements made in this release:
---
 * ASTERISK-30411 - app_read: add option to include terminating
  digit on empty, terminated strings
  (Reported by Michael
  Bradeen)
 * ASTERISK-30405 - app_directory: Add 's' option to skip
  channel call
  (Reported by Michael Bradeen)
 * ASTERISK-30422 - app_senddtmf: add the option for senddtmf to
  answer
  (Reported by Michael Bradeen)
 * ASTERISK-30325 - Upgrade Asterisk to bundled pjproject 2.13
 
  (Reported by Stanislav Abramenkov)
 * ASTERISK-30404 - app_directory: Add reading directory
  configuration from custom file
  (Reported by Michael
  Bradeen)
 * ASTERISK-29913 - func_json: Adds multi-level and array
  parsing to JSON_DECODE
  (Reported by N A)
 * ASTERISK-30353 - func_frame_trace: Print text for text
  frames
  (Reported by N A)
 * ASTERISK-30361 - json.h: Add missing
  ast_json_object_real_get
  (Reported by N A)
 * ASTERISK-30280 - Create capability to assign a Media
  Experience Score to RTP streams
  (Reported by George
  Joseph)
 * ASTERISK-30332 - func_callerid: Warn if invalid redirecting
  reason provided
  (Reported by N A)

For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/ast

[asterisk-dev] Asterisk 18.17.0-rc1 Now Available

2023-03-02 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the first
release candidate of Asterisk 18.17.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.17.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release candidate:

New Features made in this release:
---
 * ASTERISK-29810 - app_signal: Add channel signaling
  applications
  (Reported by N A)
 * ASTERISK-30262 - res_pjsip_session: Allow a context to be
  specified for overlap dialing
  (Reported by N A)
 * ASTERISK-30319 - Add BYE Reason support for SIP
 
  (Reported by Igor Goncharovsky)
 * ASTERISK-30180 - app_broadcast: Add a channel audio
  multicasting application
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-27830 - Asterisk crashes on Invalid UTF-8 string
   
  (Reported by AvayaXAsterisk)
 * ASTERISK-30354 - chan_iax2: Lack of formats prior to
  receiving voice frames causes jitterbuffer to stall
 
  (Reported by N A)
 * ASTERISK-30162 - when chan_iax is used to relay calls, no
  ringing indication is played
  (Reported by Jaco Kroon)
 * ASTERISK-30424 - pjproject_bundled: cross-compilation broken
  when ssl autodetected
  (Reported by Nick French)
 * ASTERISK-30388 - res_phoneprov: Stale SERVER variable when
  multi-homed
  (Reported by cmaj)
 * ASTERISK-30419 - pjsip: Crash when sending NOTIFY in PJSIP
  2.13
  (Reported by Ross Beer)
 * ASTERISK-30417 - Copy/Paste error in UnpauseQueueMember
 
  (Reported by Sean Bright)
 * ASTERISK-30406 - pbx_ael: Global variables are not expanded.

  (Reported by Sean Bright)
 * ASTERISK-29604 - ari: Segfault with lots of calls
 
  (Reported by Danila Evgrafov)
 * ASTERISK-30391 - res_rtp_asterisk: Issue with transcoding
  g722 after MES changes
  (Reported by George Joseph)
 * ASTERISK-30345 - loader.c: Modules that decline to load
  cannot be reloaded
  (Reported by N A)
 * ASTERISK-30379 - http: fix NULL pointer dereference while
  enable_status on TLS-only
  (Reported by Boris P. Korzun)
 * ASTERISK-30375 - res_http_media_cache: Crash when URL has no
  path component.
  (Reported by Sean Bright)
 * ASTERISK-30351 - manager: Originate variables are not added
  when setvar used in manager.conf
  (Reported by Sebastian
  Gutierrez)
 * ASTERISK-30369 - res_pjsip: Websockets from same IP shut down
  when they shouldn't be
  (Reported by Joshua C. Colp)
 * ASTERISK-30367 - pbx: Fix outdated channel snapshots with
  pbx_exec
  (Reported by N A)
 * ASTERISK-28767 - chan_pjsip: Caller ID not used when checking
  for extension, callerid supplement executed too late
 
  (Reported by Oleg)
 * ASTERISK-30350 - res_pjsip_sdp_rtp: rtp_timeout_hold is not
  used when moh_passthrough has call on hold
  (Reported by
  Benjamin Keith Ford)
 * ASTERISK-30240 - app voicemail odbc build error with gcc
  11.1
  (Reported by Michael Bradeen)
 * ASTERISK-30100 - res_pjsip: Path is ignored on INVITE to
  endpoint
  (Reported by Yury Kirsanov)
 * ASTERISK-30198 - Error `Too many open files` occurs after
  about ~8000 calls when using mixmonitor
  (Reported by
  Julien Alie)

Improvements made in this release:
---
 * ASTERISK-30411 - app_read: add option to include terminating
  digit on empty, terminated strings
  (Reported by Michael
  Bradeen)
 * ASTERISK-30405 - app_directory: Add 's' option to skip
  channel call
  (Reported by Michael Bradeen)
 * ASTERISK-30422 - app_senddtmf: add the option for senddtmf to
  answer
  (Reported by Michael Bradeen)
 * ASTERISK-30325 - Upgrade Asterisk to bundled pjproject 2.13
 
  (Reported by Stanislav Abramenkov)
 * ASTERISK-30404 - app_directory: Add reading directory
  configuration from custom file
  (Reported by Michael
  Bradeen)
 * ASTERISK-29913 - func_json: Adds multi-level and array
  parsing to JSON_DECODE
  (Reported by N A)
 * ASTERISK-30353 - func_frame_trace: Print text for text
  frames
  (Reported by N A)
 * ASTERISK-30361 - json.h: Add missing
  ast_json_object_real_get
  (Reported by N A)
 * ASTERISK-30280 - Create capability to assign a Media
  Experience Score to RTP streams
  (Reported by George
  Joseph)
 * ASTERISK-30332 - func_callerid: Warn if invalid redirecting
  reason provided
  (Reported by N A)

For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0-rc1

Thank you for your continued supp

[asterisk-dev] Certified Asterisk 18.9-cert4 Now Available

2023-01-24 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Certified 
Asterisk 18.9-cert4.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/certified-asterisk

The release of Certified Asterisk 18.9-cert4 resolves several issues reported 
by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Bugs fixed in this release:
---
 * ASTERISK-30369 - res_pjsip: Websockets from same IP shut down
  when they shouldn't be
  (Reported by Joshua C. Colp)
 * ASTERISK-30065 - pjsip: Open Websocket connection is not
  reused for outgoing requests
  (Reported by LA)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-18.9-cert4

Thank you for your continued support of Asterisk!
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[asterisk-dev] Asterisk 20.1.0-rc1 Now Available

2022-12-15 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the first
release candidate of Asterisk 20.1.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 20.1.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release candidate:

Security bugs fixed in this release:
---
 * ASTERISK-30338 - pjproject: Backport security fixes from
  2.13
  (Reported by Benjamin Keith Ford)
 * ASTERISK-30176 - manager: GetConfig can read files outside of
  Asterisk
  (Reported by shawty)
 * ASTERISK-30103 - chan_ooh323 Vulnerability in calling/called
  party IE
  (Reported by Michael Bradeen)

Improvements made in this release:
---
 * ASTERISK-30328 - Typo in from_domain description on res_pjsip
  configuration documentation
  (Reported by Marcel Wagner)
 * ASTERISK-30316 - res_pjsip: Documentation should point out
  default if contact_user is not being set for outbound
  registrations
  (Reported by Marcel Wagner)
 * ASTERISK-30289 - xmldoc: Allow XML docs to be reloaded
 
  (Reported by N A)
 * ASTERISK-30327 - rtp_engine.h: Remove obsolete example usage

  (Reported by N A)
 * ASTERISK-30286 - app_mixmonitor: Add option to use real
  Caller ID for Caller ID
  (Reported by N A)
 * ASTERISK-30308 - pbx_builtins: Allow Answer to return
  immediately
  (Reported by N A)
 * ASTERISK-30295 - test_json: Remove duplicated static
  function
  (Reported by N A)
 * ASTERISK-30290 - file.c: Don't emit warnings on winks.
 
  (Reported by N A)
 * ASTERISK-30241 - res_pjsip_gelocation: Downgrade some NOTICE
  scope trace debugs to DEBUG level
  (Reported by N A)
 * ASTERISK-30223 - features: add no-answer option to Bridge
  application
  (Reported by N A)
 * ASTERISK-30158 - PJSIP: Add new 100rel option
  "peer_supported"
  (Reported by Maximilian Fridrich)

Bugs fixed in this release:
---
 * ASTERISK-30349 - app_if:  Format truncation error
 
  (Reported by George Joseph)
 * ASTERISK-30344 - ari: Memory leak in create when specifying
  JSON
  (Reported by Saken)
 * ASTERISK-30283 - app_voicemail: Fix msg_create_from_file not
  sending email to user
  (Reported by N A)
 * ASTERISK-30265 - res_pjsip_session: Fix missing PLAR support
  on INVITEs
  (Reported by N A)
 * ASTERISK-29793 - adsi: CAS is malformed
  (Reported by N
  A)
 * ASTERISK-30311 - func_presencestate: Fix invalid memory
  access.
  (Reported by N A)
 * ASTERISK-30336 - sig_analog: Fix no timeout duration
 
  (Reported by N A)
 * ASTERISK-30244 - res_pjsip_pubsub: Occasional crash when
  TCP/TLS connection terminated and subscription persistence is
  removed
  (Reported by nappsoft)
 * ASTERISK-30184 - res_pjsip_session: re-INVITE after answering
  results in wrong stream direction of first call leg
 
  (Reported by Maximilian Fridrich)
 * ASTERISK-29998 - sla: deadlock when calling SLAStation
  application
  (Reported by N A)
 * ASTERISK-30321 - Build:  Embedded blobs have executable
  stacks
  (Reported by George Joseph)
 * ASTERISK-30293 - Memory leak in JSON_DECODE
  (Reported
  by David Uczen)
 * ASTERISK-30314 - res_agi: RECORD FILE doesn't respect
  "transmit_silence" asterisk.conf option
  (Reported by
  Joshua C. Colp)
 * ASTERISK-30285 - manager.c: Remove outdated documentation
   
  (Reported by N A)
 * ASTERISK-30282 - CI: Coredump output isn't saved when running
  unittests
  (Reported by George Joseph)
 * ASTERISK-30076 - app_stack: Incorrect exit location in
  predial handlers logged
  (Reported by N A)
 * ASTERISK-30281 - chan_rtp: Local address being used before
  being set
  (Reported by George Joseph)
 * ASTERISK-28689 - res_pjsip: Crash when locking group lock
  when sending stateful response
  (Reported by Jesse Ross)
 * ASTERISK-30278 - tcptls: Abort occurs if SSL error is logged
  if MALLOC_DEBUG is enabled
  (Reported by N A)
 * ASTERISK-30217 - Registration do not allow multiple proxies
 
  (Reported by Igor Goncharovsky)
 * ASTERISK-30273 - test_mwi: compilation fails on 32-bit
  Debian
  (Reported by N A)
 * ASTERISK-30193 - chan_pjsip should return all codecs on a
  re-INVITE without SDP
  (Reported by Henning Westerholt)
 * ASTERISK-30258 - Dialing API: Cancel a running async thread,
  does not always cancel all calls
  (Reported by Frederic LE
  FOLL)
 * ASTERISK-30274 - chan_dahdi: Unavailable channels are BUSY
  
  (Reported by N A)
 * ASTERISK-30264 - res_pjsip: Subscription handlers do not get

[asterisk-dev] Asterisk 19.8.0-rc1 Now Available

2022-12-15 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the first
release candidate of Asterisk 19.8.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 19.8.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release candidate:

Security bugs fixed in this release:
---
 * ASTERISK-30338 - pjproject: Backport security fixes from
  2.13
  (Reported by Benjamin Keith Ford)
 * ASTERISK-30176 - manager: GetConfig can read files outside of
  Asterisk
  (Reported by shawty)
 * ASTERISK-30103 - chan_ooh323 Vulnerability in calling/called
  party IE
  (Reported by Michael Bradeen)

Improvements made in this release:
---
 * ASTERISK-30241 - res_pjsip_gelocation: Downgrade some NOTICE
  scope trace debugs to DEBUG level
  (Reported by N A)
 * ASTERISK-30223 - features: add no-answer option to Bridge
  application
  (Reported by N A)
 * ASTERISK-30158 - PJSIP: Add new 100rel option
  "peer_supported"
  (Reported by Maximilian Fridrich)

Bugs fixed in this release:
---
 * ASTERISK-30244 - res_pjsip_pubsub: Occasional crash when
  TCP/TLS connection terminated and subscription persistence is
  removed
  (Reported by nappsoft)
 * ASTERISK-30278 - tcptls: Abort occurs if SSL error is logged
  if MALLOC_DEBUG is enabled
  (Reported by N A)
 * ASTERISK-28689 - res_pjsip: Crash when locking group lock
  when sending stateful response
  (Reported by Jesse Ross)
 * ASTERISK-30217 - Registration do not allow multiple proxies
 
  (Reported by Igor Goncharovsky)
 * ASTERISK-30273 - test_mwi: compilation fails on 32-bit
  Debian
  (Reported by N A)
 * ASTERISK-30193 - chan_pjsip should return all codecs on a
  re-INVITE without SDP
  (Reported by Henning Westerholt)
 * ASTERISK-30258 - Dialing API: Cancel a running async thread,
  does not always cancel all calls
  (Reported by Frederic LE
  FOLL)
 * ASTERISK-30274 - chan_dahdi: Unavailable channels are BUSY
  
  (Reported by N A)
 * ASTERISK-30264 - res_pjsip: Subscription handlers do not get
  cleanly unregistered, causing crash
  (Reported by N A)
 * ASTERISK-30248 - ast_get_digit_str adds bogus initial
  delimiter if first character not to be spoken
  (Reported by
  David Woolley)
 * ASTERISK-30213 - Make crypto_load() reentrant and handle
  symlinks correctly
  (Reported by Philip Prindeville)
 * ASTERISK-30256 - chan_dahdi: Fix format truncation warnings
 
  (Reported by N A)
 * ASTERISK-30239 - Prometheus plugin crashes Asterisk when
  using local channel
  (Reported by Joeran Vinzens)
 * ASTERISK-30237 - res_prometheus: Crash when scraping bridges

  (Reported by Igor Yeroshev)
 * ASTERISK-30245 - db: ListItems is incorrect
  (Reported
  by N A)
 * ASTERISK-30243 - func_logic: IF function complains if both
  branches are empty
  (Reported by N A)
 * ASTERISK-30232 - Initialize stack-based ast_test_capture
  structures correctly
  (Reported by Philip Prindeville)
 * ASTERISK-30220 - func_scramble: Fix segfault due to null
  pointer deref
  (Reported by N A)
 * ASTERISK-30235 - res_crypto and tests:  Memory issues and and
  uninitialized variable error
  (Reported by George Joseph)
 * ASTERISK-30234 - res_geolocation: ...may be used
  uninitialized error in geoloc_config.c
  (Reported by George
  Joseph)
 * ASTERISK-30226 - REGRESSION: res_crypto complains about the
  stir_shaken directory in /var/lib/asterisk/keys
  (Reported
  by George Joseph)

New Features made in this release:
---
 * ASTERISK-30146 - res_pjsip_logger: Add method-based log
  filtering
  (Reported by N A)
 * ASTERISK-30263 - res_pjsip_notify: Allow using
  pjsip_notify.conf from AMI
  (Reported by N A)
 * ASTERISK-30091 - cdr: Allow CDRs to ignore call state
  changes
  (Reported by N A)
 * ASTERISK-30254 - res_tonedetect: Add audible ringback
  detection to TONE_DETECT
  (Reported by N A)
 * ASTERISK-30032 - Support of mediasec SIP headers and SDP
  attributes
  (Reported by Maximilian Fridrich)
 * ASTERISK-30216 - app_bridgewait: Add option for BridgeWait to
  not answer
  (Reported by N A)
 * ASTERISK-30179 - app_amd: Allow audio to be played while AMD
  is running
  (Reported by N A)
 * ASTERISK-29432 - New function to allow access to any channel

  (Reported by N A)
 * ASTERISK-30222 - func_strings: Add trim functions
 
  (Reported by N A)

For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk

[asterisk-dev] Asterisk 18.16.0-rc1 Now Available

2022-12-15 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the first
release candidate of Asterisk 18.16.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.16.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release candidate:

Security bugs fixed in this release:
---
 * ASTERISK-30338 - pjproject: Backport security fixes from
  2.13
  (Reported by Benjamin Keith Ford)
 * ASTERISK-30176 - manager: GetConfig can read files outside of
  Asterisk
  (Reported by shawty)
 * ASTERISK-30103 - chan_ooh323 Vulnerability in calling/called
  party IE
  (Reported by Michael Bradeen)

Improvements made in this release:
---
 * ASTERISK-30328 - Typo in from_domain description on res_pjsip
  configuration documentation
  (Reported by Marcel Wagner)
 * ASTERISK-30316 - res_pjsip: Documentation should point out
  default if contact_user is not being set for outbound
  registrations
  (Reported by Marcel Wagner)
 * ASTERISK-30289 - xmldoc: Allow XML docs to be reloaded
 
  (Reported by N A)
 * ASTERISK-30327 - rtp_engine.h: Remove obsolete example usage

  (Reported by N A)
 * ASTERISK-30286 - app_mixmonitor: Add option to use real
  Caller ID for Caller ID
  (Reported by N A)
 * ASTERISK-30308 - pbx_builtins: Allow Answer to return
  immediately
  (Reported by N A)
 * ASTERISK-30295 - test_json: Remove duplicated static
  function
  (Reported by N A)
 * ASTERISK-30290 - file.c: Don't emit warnings on winks.
 
  (Reported by N A)
 * ASTERISK-30241 - res_pjsip_gelocation: Downgrade some NOTICE
  scope trace debugs to DEBUG level
  (Reported by N A)
 * ASTERISK-30223 - features: add no-answer option to Bridge
  application
  (Reported by N A)
 * ASTERISK-30158 - PJSIP: Add new 100rel option
  "peer_supported"
  (Reported by Maximilian Fridrich)

Bugs fixed in this release:
---
 * ASTERISK-30349 - app_if:  Format truncation error
 
  (Reported by George Joseph)
 * ASTERISK-30265 - res_pjsip_session: Fix missing PLAR support
  on INVITEs
  (Reported by N A)
 * ASTERISK-30283 - app_voicemail: Fix msg_create_from_file not
  sending email to user
  (Reported by N A)
 * ASTERISK-29793 - adsi: CAS is malformed
  (Reported by N
  A)
 * ASTERISK-30344 - ari: Memory leak in create when specifying
  JSON
  (Reported by Saken)
 * ASTERISK-30311 - func_presencestate: Fix invalid memory
  access.
  (Reported by N A)
 * ASTERISK-30336 - sig_analog: Fix no timeout duration
 
  (Reported by N A)
 * ASTERISK-30244 - res_pjsip_pubsub: Occasional crash when
  TCP/TLS connection terminated and subscription persistence is
  removed
  (Reported by nappsoft)
 * ASTERISK-30184 - res_pjsip_session: re-INVITE after answering
  results in wrong stream direction of first call leg
 
  (Reported by Maximilian Fridrich)
 * ASTERISK-29998 - sla: deadlock when calling SLAStation
  application
  (Reported by N A)
 * ASTERISK-30321 - Build:  Embedded blobs have executable
  stacks
  (Reported by George Joseph)
 * ASTERISK-30293 - Memory leak in JSON_DECODE
  (Reported
  by David Uczen)
 * ASTERISK-30314 - res_agi: RECORD FILE doesn't respect
  "transmit_silence" asterisk.conf option
  (Reported by
  Joshua C. Colp)
 * ASTERISK-30285 - manager.c: Remove outdated documentation
   
  (Reported by N A)
 * ASTERISK-30076 - app_stack: Incorrect exit location in
  predial handlers logged
  (Reported by N A)
 * ASTERISK-30282 - CI: Coredump output isn't saved when running
  unittests
  (Reported by George Joseph)
 * ASTERISK-30281 - chan_rtp: Local address being used before
  being set
  (Reported by George Joseph)
 * ASTERISK-28689 - res_pjsip: Crash when locking group lock
  when sending stateful response
  (Reported by Jesse Ross)
 * ASTERISK-30217 - Registration do not allow multiple proxies
 
  (Reported by Igor Goncharovsky)
 * ASTERISK-30278 - tcptls: Abort occurs if SSL error is logged
  if MALLOC_DEBUG is enabled
  (Reported by N A)
 * ASTERISK-30273 - test_mwi: compilation fails on 32-bit
  Debian
  (Reported by N A)
 * ASTERISK-30193 - chan_pjsip should return all codecs on a
  re-INVITE without SDP
  (Reported by Henning Westerholt)
 * ASTERISK-30258 - Dialing API: Cancel a running async thread,
  does not always cancel all calls
  (Reported by Frederic LE
  FOLL)
 * ASTERISK-30274 - chan_dahdi: Unavailable channels are BUSY
  
  (Reported by N A)
 * ASTERISK-30248 - ast_get_digit_str adds bogus initial
  del

[asterisk-dev] Asterisk 16.30.0-rc1 Now Available

2022-12-15 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the first
release candidate of Asterisk 16.30.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.30.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release candidate:

Security bugs fixed in this release:
---
 * ASTERISK-30338 - pjproject: Backport security fixes from
  2.13
  (Reported by Benjamin Keith Ford)
 * ASTERISK-30176 - manager: GetConfig can read files outside of
  Asterisk
  (Reported by shawty)
 * ASTERISK-30103 - chan_ooh323 Vulnerability in calling/called
  party IE
  (Reported by Michael Bradeen)

Improvements made in this release:
---
 * ASTERISK-30241 - res_pjsip_gelocation: Downgrade some NOTICE
  scope trace debugs to DEBUG level
  (Reported by N A)
 * ASTERISK-30223 - features: add no-answer option to Bridge
  application
  (Reported by N A)
 * ASTERISK-30158 - PJSIP: Add new 100rel option
  "peer_supported"
  (Reported by Maximilian Fridrich)

Bugs fixed in this release:
---
 * ASTERISK-30244 - res_pjsip_pubsub: Occasional crash when
  TCP/TLS connection terminated and subscription persistence is
  removed
  (Reported by nappsoft)
 * ASTERISK-28689 - res_pjsip: Crash when locking group lock
  when sending stateful response
  (Reported by Jesse Ross)
 * ASTERISK-30217 - Registration do not allow multiple proxies
 
  (Reported by Igor Goncharovsky)
 * ASTERISK-30193 - chan_pjsip should return all codecs on a
  re-INVITE without SDP
  (Reported by Henning Westerholt)
 * ASTERISK-30213 - Make crypto_load() reentrant and handle
  symlinks correctly
  (Reported by Philip Prindeville)
 * ASTERISK-30256 - chan_dahdi: Fix format truncation warnings
 
  (Reported by N A)
 * ASTERISK-30245 - db: ListItems is incorrect
  (Reported
  by N A)
 * ASTERISK-30243 - func_logic: IF function complains if both
  branches are empty
  (Reported by N A)
 * ASTERISK-30232 - Initialize stack-based ast_test_capture
  structures correctly
  (Reported by Philip Prindeville)
 * ASTERISK-30220 - func_scramble: Fix segfault due to null
  pointer deref
  (Reported by N A)
 * ASTERISK-30235 - res_crypto and tests:  Memory issues and and
  uninitialized variable error
  (Reported by George Joseph)
 * ASTERISK-30234 - res_geolocation: ...may be used
  uninitialized error in geoloc_config.c
  (Reported by George
  Joseph)
 * ASTERISK-30226 - REGRESSION: res_crypto complains about the
  stir_shaken directory in /var/lib/asterisk/keys
  (Reported
  by George Joseph)

New Features made in this release:
---
 * ASTERISK-30091 - cdr: Allow CDRs to ignore call state
  changes
  (Reported by N A)
 * ASTERISK-30254 - res_tonedetect: Add audible ringback
  detection to TONE_DETECT
  (Reported by N A)
 * ASTERISK-30032 - Support of mediasec SIP headers and SDP
  attributes
  (Reported by Maximilian Fridrich)
 * ASTERISK-30216 - app_bridgewait: Add option for BridgeWait to
  not answer
  (Reported by N A)
 * ASTERISK-30179 - app_amd: Allow audio to be played while AMD
  is running
  (Reported by N A)
 * ASTERISK-29432 - New function to allow access to any channel

  (Reported by N A)
 * ASTERISK-30222 - func_strings: Add trim functions
 
  (Reported by N A)

For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.30.0-rc1

Thank you for your continued support of Asterisk!
-- 
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[asterisk-dev] Asterisk 16.29.1, 18.15.1, 19.7.1, 20.0.1 Now Available

2022-12-01 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of
Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves
issues reported by the
community and would have not been possible without your participation.Thank
you!

The following issue is resolved in this release:

Bugs fixed in this release:
———–

[ASTERISK-30103 <https://issues.asterisk.org/jira/browse/ASTERISK-30103>]
chan_ooh323 vulnerability in calling/called party IE (Reported By: Michael
Bradeen)

[ASTERISK-30176 <https://issues.asterisk.org/jira/browse/ASTERISK-30176>]
GetConfig can read files outside of Asterisk (Reported By: shawty)

[ASTERISK-30244 <https://issues.asterisk.org/jira/browse/ASTERISK-30244>]
Occasional crash when TCP/TLS connection terminated and subscription
persistence is removed (Reported By: nappsoft)

[ASTERISK-30338 <https://issues.asterisk.org/jira/browse/ASTERISK-30338>]
Backport 2.13 security fixes from pjproject

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1

https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1

https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1

https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1

Thank you for your continued support of Asterisk!
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[asterisk-dev] Asterisk 20.0.0 Now Available

2022-10-19 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
20.0.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 20.0.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Deprecations made in this release:
---
 * ASTERISK-29601 - moduleinfo: Add replacement module
  information
  (Reported by N A)
 * ASTERISK-29602 - res_monitor: Disable building by default.
  
  (Reported by Joshua C. Colp)
 * ASTERISK-29600 - muted: Remove deprecated application
 
  (Reported by Joshua C. Colp)
 * ASTERISK-29599 - conf2ael: Remove deprecated application

  (Reported by Joshua C. Colp)
 * ASTERISK-29598 - res_config_sqlite: Remove deprecated module

  (Reported by Joshua C. Colp)
 * ASTERISK-29597 - chan_vpb: Remove deprecated module
 
  (Reported by Joshua C. Colp)
 * ASTERISK-29596 - chan_misdn: Remove deprecated module
 
  (Reported by Joshua C. Colp)
 * ASTERISK-29595 - chan_nbs: Remove deprecated module
 
  (Reported by Joshua C. Colp)
 * ASTERISK-29594 - chan_phone: Remove deprecated module
 
  (Reported by Joshua C. Colp)
 * ASTERISK-29593 - chan_oss: Remove deprecated module
 
  (Reported by Joshua C. Colp)
 * ASTERISK-29592 - cdr_syslog: Remove deprecated module
 
  (Reported by Joshua C. Colp)
 * ASTERISK-29591 - app_dahdiras: Remove deprecated module
 
  (Reported by Joshua C. Colp)
 * ASTERISK-29590 - app_nbscat: Remove deprecated module
 
  (Reported by Joshua C. Colp)
 * ASTERISK-29589 - app_image: Remove deprecated module
 
  (Reported by Joshua C. Colp)
 * ASTERISK-29588 - app_url: Remove deprecated module
 
  (Reported by Joshua C. Colp)
 * ASTERISK-29587 - app_fax: Remove deprecated module
 
  (Reported by Joshua C. Colp)
 * ASTERISK-29586 - app_ices: Remove deprecated module
 
  (Reported by Joshua C. Colp)
 * ASTERISK-29585 - app_mysql: Remove deprecated module
 
  (Reported by Joshua C. Colp)
 * ASTERISK-29584 - cdr_mysql: Remove deprecated module
 
  (Reported by Joshua C. Colp)
 * ASTERISK-29548 - app_meetme: Deprecated in 19, to be removed
  in 21
  (Reported by Joshua C. Colp)
 * ASTERISK-29549 - app_osploop: Deprecated in 19, to be removed
  in 21
  (Reported by Joshua C. Colp)
 * ASTERISK-29550 - chan_alsa: Deprecated in 19, to be removed
  in 21
  (Reported by Joshua C. Colp)
 * ASTERISK-29551 - chan_mgcp: Deprecated in 19, to be removed
  in 21
  (Reported by Joshua C. Colp)
 * ASTERISK-29552 - chan_skinny: Deprecated in 19, to be removed
  in 21
  (Reported by Joshua C. Colp)
 * ASTERISK-29553 - res_pktccops: Deprecated in 19, to be
  removed in 21
  (Reported by Joshua C. Colp)
 * ASTERISK-29558 - app_macro: Deprecated in 16, to be removed
  in 21
  (Reported by Joshua C. Colp)
 * ASTERISK-29567 - chan_sip: Deprecated in 17, to be removed in
  21
  (Reported by Joshua C. Colp)
 * ASTERISK-29572 - res_monitor: Deprecated in 16, to be removed
  in 21
  (Reported by Joshua C. Colp)

Security bugs fixed in this release:
---
 * ASTERISK-29476 - res_stir_shaken: Blind SSRF vulnerabilities

  (Reported by Clint Ruoho)
 * ASTERISK-29872 - res_stir_shaken: Resource exhaustion with
  large files
  (Reported by Benjamin Keith Ford)
 * ASTERISK-29838 - ${SQL_ESC()} not correctly escaping a
  terminating \
  (Reported by Leandro Dardini)
 * ASTERISK-29415 - Crash in PJSIP TLS transport 
 
  (Reported by Andrew Yager)
 * ASTERISK-29381 - chan_pjsip: Remote denial of service by an
  authenticated user
  (Reported by Ivan Poddubny)

New Features made in this release:
---
 * ASTERISK-30037 - Add test support to calling external
  processes
  (Reported by Philip Prindeville)
 * ASTERISK-30161 - locks: add AMI event for deadlock
 
  (Reported by N A)
 * ASTERISK-30211 - app_confbridge: Add end_marked_any option
  
  (Reported by N A)
 * ASTERISK-30186 - res_pjsip: Add support for reloading TLS
  certificate and key information
  (Reported by Joshua C.
  Colp)
 * ASTERISK-29899 - features: Add advanced transfer initiation
  options
  (Reported by N A)
 * ASTERISK-30136 - db: Add AMI action to retrieve all keys
  beginning with a prefix
  (Reported by N A)
 * ASTERISK-3 - chan_dahdi: Add POLARITY function
 
  (Reported by N A)
 * ASTERISK-30062 - cli: Add CLI command to execute a dialplan
  app
  (Reported by N A)
 * ASTERISK-2 - pjsip: Get information from 200 OK INVITE
  reply headers
  (Reported by JosÃ© Lopes)
 * ASTERISK-30061 - pbx: Add pbx helper application

[asterisk-dev] Asterisk 19.7.0 Now Available

2022-10-19 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
19.7.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 19.7.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

New Features made in this release:
---
 * ASTERISK-30037 - Add test support to calling external
  processes
  (Reported by Philip Prindeville)
 * ASTERISK-30161 - locks: add AMI event for deadlock
 
  (Reported by N A)
 * ASTERISK-30211 - app_confbridge: Add end_marked_any option
  
  (Reported by N A)
 * ASTERISK-30186 - res_pjsip: Add support for reloading TLS
  certificate and key information
  (Reported by Joshua C.
  Colp)
 * ASTERISK-29899 - features: Add advanced transfer initiation
  options
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-30235 - res_crypto and tests:  Memory issues and and
  uninitialized variable error
  (Reported by George Joseph)
 * ASTERISK-30234 - res_geolocation: ...may be used
  uninitialized error in geoloc_config.c
  (Reported by George
  Joseph)
 * ASTERISK-30215 - Inbound SIP INVITE with Geo Location causing
  a Segmentation Fault
  (Reported by Dan Cropp)
 * ASTERISK-30135 - [res_musiconhold] Allows the moh only for
  the answered call
  (Reported by sungtae kim)
 * ASTERISK-26894 - pjsip should support tel uri scheme
 
  (Reported by Gergely DÃ¶msÃ¶di)
 * ASTERISK-30210 - func_frame_trace: Channel masquerade
  triggers assertion
  (Reported by N A)
 * ASTERISK-30190 - res_geolocation:  GEOLOC_PROFILE isn't
  returning correct values on incoming channel
  (Reported by
  George Joseph)
 * ASTERISK-29185 - chan_pjsip: Endpoint: allow = all is
  broken.
  (Reported by Alexander Traud)
 * ASTERISK-30192 - res_tonedetect: fix typo for frametype
 
  (Reported by N A)
 * ASTERISK-29453 - alembic: incoming_call_offer_pref and
  outgoing_call_offer_pref missing in "ps_endpoints" table
 
  (Reported by Daniel ThÃ¼men)
 * ASTERISK-26826 - testsuite: Add support for Python 3
 
  (Reported by Joshua C. Colp)
 * ASTERISK-30167 - res_geolocation:  Refactor for issues found
  by users
  (Reported by George Joseph)
 * ASTERISK-28422 - Memory Leak in Confbridge menu
 
  (Reported by Ted G)
 * ASTERISK-29917 - ami: FilterList action doesn't exist
 
  (Reported by N A)
 * ASTERISK-30020 - ConfbridgeListRooms Event Not Documented
   
  (Reported by Michael Cargile)
 * ASTERISK-30018 - app_meetme: MeetmeList AMI event not
  documented
  (Reported by Michael Cargile)
 * ASTERISK-30151 - Documentation doesn't include info about
  "field", a 3rd required parameter.
  (Reported by Chris
  Young)

Improvements made in this release:
---
 * ASTERISK-30241 - res_pjsip_gelocation: Downgrade some NOTICE
  scope trace debugs to DEBUG level
  (Reported by N A)
 * ASTERISK-30178 - extend user_eq_phone behavior to local
  uri's
  (Reported by Michael Bradeen)
 * ASTERISK-30046 - Reimplement res/res_crypto.c internals with
  EVP_PKEY interface to Openssl API's
  (Reported by Philip
  Prindeville)
 * ASTERISK-30045 - Add test coverage to res/res_crypto.c
  functionality
  (Reported by Philip Prindeville)
 * ASTERISK-30185 - res_geolocation: Allow location parameters
  to be specified in profiles
  (Reported by George Joseph)
 * ASTERISK-30177 - res_geolocation:  Add option to suppress
  empty elements
  (Reported by George Joseph)
 * ASTERISK-30182 - res_geolocation: Add built-in profiles to
  use in fully dynamic configurations
  (Reported by George
  Joseph)
 * ASTERISK-29906 - [patch] update RLS to reflect the changes to
  the lists
  (Reported by Alexei Gradinari)
 * ASTERISK-30163 - general: fix minor formatting issues
 
  (Reported by N A)
 * ASTERISK-30164 - chan_iax2: Add missing option documentation

  (Reported by N A)
 * ASTERISK-30153 - logger: Improve log levels
  (Reported
  by N A)
 * ASTERISK-30160 - cdr.conf: Remove obsolete app_mysql
  reference
  (Reported by N A)
 * ASTERISK-30159 - general: Remove obsolete SVN references

  (Reported by N A)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0

Thank you for your continued support of Asterisk!
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   http

[asterisk-dev] Asterisk 18.15.0 Now Available

2022-10-19 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
18.15.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.15.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

New Features made in this release:
---
 * ASTERISK-30037 - Add test support to calling external
  processes
  (Reported by Philip Prindeville)
 * ASTERISK-30161 - locks: add AMI event for deadlock
 
  (Reported by N A)
 * ASTERISK-30211 - app_confbridge: Add end_marked_any option
  
  (Reported by N A)
 * ASTERISK-30186 - res_pjsip: Add support for reloading TLS
  certificate and key information
  (Reported by Joshua C.
  Colp)
 * ASTERISK-29899 - features: Add advanced transfer initiation
  options
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-30235 - res_crypto and tests:  Memory issues and and
  uninitialized variable error
  (Reported by George Joseph)
 * ASTERISK-30234 - res_geolocation: ...may be used
  uninitialized error in geoloc_config.c
  (Reported by George
  Joseph)
 * ASTERISK-30215 - Inbound SIP INVITE with Geo Location causing
  a Segmentation Fault
  (Reported by Dan Cropp)
 * ASTERISK-30135 - [res_musiconhold] Allows the moh only for
  the answered call
  (Reported by sungtae kim)
 * ASTERISK-26894 - pjsip should support tel uri scheme
 
  (Reported by Gergely DÃ¶msÃ¶di)
 * ASTERISK-30210 - func_frame_trace: Channel masquerade
  triggers assertion
  (Reported by N A)
 * ASTERISK-30190 - res_geolocation:  GEOLOC_PROFILE isn't
  returning correct values on incoming channel
  (Reported by
  George Joseph)
 * ASTERISK-29185 - chan_pjsip: Endpoint: allow = all is
  broken.
  (Reported by Alexander Traud)
 * ASTERISK-30192 - res_tonedetect: fix typo for frametype
 
  (Reported by N A)
 * ASTERISK-29453 - alembic: incoming_call_offer_pref and
  outgoing_call_offer_pref missing in "ps_endpoints" table
 
  (Reported by Daniel ThÃ¼men)
 * ASTERISK-26826 - testsuite: Add support for Python 3
 
  (Reported by Joshua C. Colp)
 * ASTERISK-30167 - res_geolocation:  Refactor for issues found
  by users
  (Reported by George Joseph)
 * ASTERISK-28422 - Memory Leak in Confbridge menu
 
  (Reported by Ted G)
 * ASTERISK-29917 - ami: FilterList action doesn't exist
 
  (Reported by N A)
 * ASTERISK-30018 - app_meetme: MeetmeList AMI event not
  documented
  (Reported by Michael Cargile)
 * ASTERISK-30020 - ConfbridgeListRooms Event Not Documented
   
  (Reported by Michael Cargile)
 * ASTERISK-30151 - Documentation doesn't include info about
  "field", a 3rd required parameter.
  (Reported by Chris
  Young)

Improvements made in this release:
---
 * ASTERISK-30241 - res_pjsip_gelocation: Downgrade some NOTICE
  scope trace debugs to DEBUG level
  (Reported by N A)
 * ASTERISK-30178 - extend user_eq_phone behavior to local
  uri's
  (Reported by Michael Bradeen)
 * ASTERISK-30046 - Reimplement res/res_crypto.c internals with
  EVP_PKEY interface to Openssl API's
  (Reported by Philip
  Prindeville)
 * ASTERISK-30045 - Add test coverage to res/res_crypto.c
  functionality
  (Reported by Philip Prindeville)
 * ASTERISK-30185 - res_geolocation: Allow location parameters
  to be specified in profiles
  (Reported by George Joseph)
 * ASTERISK-30177 - res_geolocation:  Add option to suppress
  empty elements
  (Reported by George Joseph)
 * ASTERISK-30182 - res_geolocation: Add built-in profiles to
  use in fully dynamic configurations
  (Reported by George
  Joseph)
 * ASTERISK-29906 - [patch] update RLS to reflect the changes to
  the lists
  (Reported by Alexei Gradinari)
 * ASTERISK-30163 - general: fix minor formatting issues
 
  (Reported by N A)
 * ASTERISK-30164 - chan_iax2: Add missing option documentation

  (Reported by N A)
 * ASTERISK-30153 - logger: Improve log levels
  (Reported
  by N A)
 * ASTERISK-30160 - cdr.conf: Remove obsolete app_mysql
  reference
  (Reported by N A)
 * ASTERISK-30159 - general: Remove obsolete SVN references

  (Reported by N A)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0

Thank you for your continued support of Asterisk!
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[asterisk-dev] Asterisk 16.29.0 Now Available

2022-10-19 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
16.29.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.29.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

New Features made in this release:
---
 * ASTERISK-30037 - Add test support to calling external
  processes
  (Reported by Philip Prindeville)
 * ASTERISK-30161 - locks: add AMI event for deadlock
 
  (Reported by N A)
 * ASTERISK-30211 - app_confbridge: Add end_marked_any option
  
  (Reported by N A)
 * ASTERISK-30186 - res_pjsip: Add support for reloading TLS
  certificate and key information
  (Reported by Joshua C.
  Colp)
 * ASTERISK-29899 - features: Add advanced transfer initiation
  options
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-30235 - res_crypto and tests:  Memory issues and and
  uninitialized variable error
  (Reported by George Joseph)
 * ASTERISK-30234 - res_geolocation: ...may be used
  uninitialized error in geoloc_config.c
  (Reported by George
  Joseph)
 * ASTERISK-30215 - Inbound SIP INVITE with Geo Location causing
  a Segmentation Fault
  (Reported by Dan Cropp)
 * ASTERISK-30135 - [res_musiconhold] Allows the moh only for
  the answered call
  (Reported by sungtae kim)
 * ASTERISK-26894 - pjsip should support tel uri scheme
 
  (Reported by Gergely DÃ¶msÃ¶di)
 * ASTERISK-30210 - func_frame_trace: Channel masquerade
  triggers assertion
  (Reported by N A)
 * ASTERISK-30190 - res_geolocation:  GEOLOC_PROFILE isn't
  returning correct values on incoming channel
  (Reported by
  George Joseph)
 * ASTERISK-29185 - chan_pjsip: Endpoint: allow = all is
  broken.
  (Reported by Alexander Traud)
 * ASTERISK-30192 - res_tonedetect: fix typo for frametype
 
  (Reported by N A)
 * ASTERISK-29453 - alembic: incoming_call_offer_pref and
  outgoing_call_offer_pref missing in "ps_endpoints" table
 
  (Reported by Daniel ThÃ¼men)
 * ASTERISK-26826 - testsuite: Add support for Python 3
 
  (Reported by Joshua C. Colp)
 * ASTERISK-30167 - res_geolocation:  Refactor for issues found
  by users
  (Reported by George Joseph)
 * ASTERISK-28422 - Memory Leak in Confbridge menu
 
  (Reported by Ted G)
 * ASTERISK-29917 - ami: FilterList action doesn't exist
 
  (Reported by N A)
 * ASTERISK-30020 - ConfbridgeListRooms Event Not Documented
   
  (Reported by Michael Cargile)
 * ASTERISK-30018 - app_meetme: MeetmeList AMI event not
  documented
  (Reported by Michael Cargile)
 * ASTERISK-30151 - Documentation doesn't include info about
  "field", a 3rd required parameter.
  (Reported by Chris
  Young)

Improvements made in this release:
---
 * ASTERISK-30241 - res_pjsip_gelocation: Downgrade some NOTICE
  scope trace debugs to DEBUG level
  (Reported by N A)
 * ASTERISK-30178 - extend user_eq_phone behavior to local
  uri's
  (Reported by Michael Bradeen)
 * ASTERISK-30046 - Reimplement res/res_crypto.c internals with
  EVP_PKEY interface to Openssl API's
  (Reported by Philip
  Prindeville)
 * ASTERISK-30045 - Add test coverage to res/res_crypto.c
  functionality
  (Reported by Philip Prindeville)
 * ASTERISK-30185 - res_geolocation: Allow location parameters
  to be specified in profiles
  (Reported by George Joseph)
 * ASTERISK-30177 - res_geolocation:  Add option to suppress
  empty elements
  (Reported by George Joseph)
 * ASTERISK-30182 - res_geolocation: Add built-in profiles to
  use in fully dynamic configurations
  (Reported by George
  Joseph)
 * ASTERISK-29906 - [patch] update RLS to reflect the changes to
  the lists
  (Reported by Alexei Gradinari)
 * ASTERISK-30163 - general: fix minor formatting issues
 
  (Reported by N A)
 * ASTERISK-30164 - chan_iax2: Add missing option documentation

  (Reported by N A)
 * ASTERISK-30160 - cdr.conf: Remove obsolete app_mysql
  reference
  (Reported by N A)
 * ASTERISK-30159 - general: Remove obsolete SVN references

  (Reported by N A)
 * ASTERISK-30153 - logger: Improve log levels
  (Reported
  by N A)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.0

Thank you for your continued support of Asterisk!
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[asterisk-dev] Asterisk 20.0.0-rc2 Now Available

2022-09-28 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the second
release candidate of Asterisk 20.0.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 20.0.0-rc2 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release candidate:

Bugs fixed in this release:
---
 * ASTERISK-30235 - res_crypto and tests:  Memory issues and and
  uninitialized variable error
  (Reported by George Joseph)
 * ASTERISK-30234 - res_geolocation: ...may be used
  uninitialized error in geoloc_config.c
  (Reported by George
  Joseph)

Improvements made in this release:
---
 * ASTERISK-30241 - res_pjsip_gelocation: Downgrade some NOTICE
  scope trace debugs to DEBUG level
  (Reported by N A)

For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0-rc2

Thank you for your continued support of Asterisk!
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[asterisk-dev] Asterisk 19.7.0-rc2 Now Available

2022-09-28 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the second
release candidate of Asterisk 19.7.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 19.7.0-rc2 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release candidate:

Bugs fixed in this release:
---
 * ASTERISK-30235 - res_crypto and tests:  Memory issues and and
  uninitialized variable error
  (Reported by George Joseph)
 * ASTERISK-30234 - res_geolocation: ...may be used
  uninitialized error in geoloc_config.c
  (Reported by George
  Joseph)

Improvements made in this release:
---
 * ASTERISK-30241 - res_pjsip_gelocation: Downgrade some NOTICE
  scope trace debugs to DEBUG level
  (Reported by N A)

For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0-rc2

Thank you for your continued support of Asterisk!
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[asterisk-dev] Asterisk 18.15.0-rc2 Now Available

2022-09-28 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the second
release candidate of Asterisk 18.15.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.15.0-rc2 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release candidate:

Bugs fixed in this release:
---
 * ASTERISK-30235 - res_crypto and tests:  Memory issues and and
  uninitialized variable error
  (Reported by George Joseph)
 * ASTERISK-30234 - res_geolocation: ...may be used
  uninitialized error in geoloc_config.c
  (Reported by George
  Joseph)

Improvements made in this release:
---
 * ASTERISK-30241 - res_pjsip_gelocation: Downgrade some NOTICE
  scope trace debugs to DEBUG level
  (Reported by N A)

For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0-rc2

Thank you for your continued support of Asterisk!
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[asterisk-dev] Asterisk 16.29.0-rc2 Now Available

2022-09-28 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the second
release candidate of Asterisk 16.29.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.29.0-rc2 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release candidate:

Bugs fixed in this release:
---
 * ASTERISK-30235 - res_crypto and tests:  Memory issues and and
  uninitialized variable error
  (Reported by George Joseph)
 * ASTERISK-30234 - res_geolocation: ...may be used
  uninitialized error in geoloc_config.c
  (Reported by George
  Joseph)

Improvements made in this release:
---
 * ASTERISK-30241 - res_pjsip_gelocation: Downgrade some NOTICE
  scope trace debugs to DEBUG level
  (Reported by N A)

For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.0-rc2

Thank you for your continued support of Asterisk!
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[asterisk-dev] Asterisk 20.0.0-rc1 Now Available

2022-09-14 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the first
release candidate of Asterisk 20.0.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 20.0.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release candidate:

Deprecations made in this release:
---
 * ASTERISK-29601 - moduleinfo: Add replacement module
  information
  (Reported by N A)
 * ASTERISK-29602 - res_monitor: Disable building by default.
  
  (Reported by Joshua C. Colp)
 * ASTERISK-29600 - muted: Remove deprecated application
 
  (Reported by Joshua C. Colp)
 * ASTERISK-29599 - conf2ael: Remove deprecated application

  (Reported by Joshua C. Colp)
 * ASTERISK-29598 - res_config_sqlite: Remove deprecated module

  (Reported by Joshua C. Colp)
 * ASTERISK-29597 - chan_vpb: Remove deprecated module
 
  (Reported by Joshua C. Colp)
 * ASTERISK-29596 - chan_misdn: Remove deprecated module
 
  (Reported by Joshua C. Colp)
 * ASTERISK-29595 - chan_nbs: Remove deprecated module
 
  (Reported by Joshua C. Colp)
 * ASTERISK-29594 - chan_phone: Remove deprecated module
 
  (Reported by Joshua C. Colp)
 * ASTERISK-29593 - chan_oss: Remove deprecated module
 
  (Reported by Joshua C. Colp)
 * ASTERISK-29592 - cdr_syslog: Remove deprecated module
 
  (Reported by Joshua C. Colp)
 * ASTERISK-29591 - app_dahdiras: Remove deprecated module
 
  (Reported by Joshua C. Colp)
 * ASTERISK-29590 - app_nbscat: Remove deprecated module
 
  (Reported by Joshua C. Colp)
 * ASTERISK-29589 - app_image: Remove deprecated module
 
  (Reported by Joshua C. Colp)
 * ASTERISK-29588 - app_url: Remove deprecated module
 
  (Reported by Joshua C. Colp)
 * ASTERISK-29587 - app_fax: Remove deprecated module
 
  (Reported by Joshua C. Colp)
 * ASTERISK-29586 - app_ices: Remove deprecated module
 
  (Reported by Joshua C. Colp)
 * ASTERISK-29585 - app_mysql: Remove deprecated module
 
  (Reported by Joshua C. Colp)
 * ASTERISK-29584 - cdr_mysql: Remove deprecated module
 
  (Reported by Joshua C. Colp)
 * ASTERISK-29548 - app_meetme: Deprecated in 19, to be removed
  in 21
  (Reported by Joshua C. Colp)
 * ASTERISK-29549 - app_osploop: Deprecated in 19, to be removed
  in 21
  (Reported by Joshua C. Colp)
 * ASTERISK-29550 - chan_alsa: Deprecated in 19, to be removed
  in 21
  (Reported by Joshua C. Colp)
 * ASTERISK-29551 - chan_mgcp: Deprecated in 19, to be removed
  in 21
  (Reported by Joshua C. Colp)
 * ASTERISK-29552 - chan_skinny: Deprecated in 19, to be removed
  in 21
  (Reported by Joshua C. Colp)
 * ASTERISK-29553 - res_pktccops: Deprecated in 19, to be
  removed in 21
  (Reported by Joshua C. Colp)
 * ASTERISK-29558 - app_macro: Deprecated in 16, to be removed
  in 21
  (Reported by Joshua C. Colp)
 * ASTERISK-29567 - chan_sip: Deprecated in 17, to be removed in
  21
  (Reported by Joshua C. Colp)
 * ASTERISK-29572 - res_monitor: Deprecated in 16, to be removed
  in 21
  (Reported by Joshua C. Colp)

Security bugs fixed in this release:
---
 * ASTERISK-29476 - res_stir_shaken: Blind SSRF vulnerabilities

  (Reported by Clint Ruoho)
 * ASTERISK-29872 - res_stir_shaken: Resource exhaustion with
  large files
  (Reported by Benjamin Keith Ford)
 * ASTERISK-29838 - ${SQL_ESC()} not correctly escaping a
  terminating \
  (Reported by Leandro Dardini)
 * ASTERISK-29415 - Crash in PJSIP TLS transport 
 
  (Reported by Andrew Yager)
 * ASTERISK-29381 - chan_pjsip: Remote denial of service by an
  authenticated user
  (Reported by Ivan Poddubny)

New Features made in this release:
---
 * ASTERISK-30037 - Add test support to calling external
  processes
  (Reported by Philip Prindeville)
 * ASTERISK-30161 - locks: add AMI event for deadlock
 
  (Reported by N A)
 * ASTERISK-30211 - app_confbridge: Add end_marked_any option
  
  (Reported by N A)
 * ASTERISK-30186 - res_pjsip: Add support for reloading TLS
  certificate and key information
  (Reported by Joshua C.
  Colp)
 * ASTERISK-29899 - features: Add advanced transfer initiation
  options
  (Reported by N A)
 * ASTERISK-30136 - db: Add AMI action to retrieve all keys
  beginning with a prefix
  (Reported by N A)
 * ASTERISK-3 - chan_dahdi: Add POLARITY function
 
  (Reported by N A)
 * ASTERISK-30062 - cli: Add CLI command to execute a dialplan
  app
  (Reported by N A)
 * ASTERISK-2 - pjsip: Get information from 200 OK INVITE
  reply headers
  (Reported by JosÃ© Lopes)
 * ASTERISK

[asterisk-dev] Asterisk 19.7.0-rc1 Now Available

2022-09-14 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the first
release candidate of Asterisk 19.7.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 19.7.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release candidate:

New Features made in this release:
---
 * ASTERISK-30037 - Add test support to calling external
  processes
  (Reported by Philip Prindeville)
 * ASTERISK-30161 - locks: add AMI event for deadlock
 
  (Reported by N A)
 * ASTERISK-30211 - app_confbridge: Add end_marked_any option
  
  (Reported by N A)
 * ASTERISK-30186 - res_pjsip: Add support for reloading TLS
  certificate and key information
  (Reported by Joshua C.
  Colp)
 * ASTERISK-29899 - features: Add advanced transfer initiation
  options
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-30135 - [res_musiconhold] Allows the moh only for
  the answered call
  (Reported by sungtae kim)
 * ASTERISK-26894 - pjsip should support tel uri scheme
 
  (Reported by Gergely DÃ¶msÃ¶di)
 * ASTERISK-30210 - func_frame_trace: Channel masquerade
  triggers assertion
  (Reported by N A)
 * ASTERISK-30190 - res_geolocation:  GEOLOC_PROFILE isn't
  returning correct values on incoming channel
  (Reported by
  George Joseph)
 * ASTERISK-29185 - chan_pjsip: Endpoint: allow = all is
  broken.
  (Reported by Alexander Traud)
 * ASTERISK-30192 - res_tonedetect: fix typo for frametype
 
  (Reported by N A)
 * ASTERISK-29453 - alembic: incoming_call_offer_pref and
  outgoing_call_offer_pref missing in "ps_endpoints" table
 
  (Reported by Daniel ThÃ¼men)
 * ASTERISK-26826 - testsuite: Add support for Python 3
 
  (Reported by Joshua C. Colp)
 * ASTERISK-30167 - res_geolocation:  Refactor for issues found
  by users
  (Reported by George Joseph)
 * ASTERISK-28422 - Memory Leak in Confbridge menu
 
  (Reported by Ted G)
 * ASTERISK-29917 - ami: FilterList action doesn't exist
 
  (Reported by N A)
 * ASTERISK-30020 - ConfbridgeListRooms Event Not Documented
   
  (Reported by Michael Cargile)
 * ASTERISK-30018 - app_meetme: MeetmeList AMI event not
  documented
  (Reported by Michael Cargile)
 * ASTERISK-30151 - Documentation doesn't include info about
  "field", a 3rd required parameter.
  (Reported by Chris
  Young)

Improvements made in this release:
---
 * ASTERISK-30178 - extend user_eq_phone behavior to local
  uri's
  (Reported by Michael Bradeen)
 * ASTERISK-30046 - Reimplement res/res_crypto.c internals with
  EVP_PKEY interface to Openssl API's
  (Reported by Philip
  Prindeville)
 * ASTERISK-30045 - Add test coverage to res/res_crypto.c
  functionality
  (Reported by Philip Prindeville)
 * ASTERISK-30185 - res_geolocation: Allow location parameters
  to be specified in profiles
  (Reported by George Joseph)
 * ASTERISK-30177 - res_geolocation:  Add option to suppress
  empty elements
  (Reported by George Joseph)
 * ASTERISK-30182 - res_geolocation: Add built-in profiles to
  use in fully dynamic configurations
  (Reported by George
  Joseph)
 * ASTERISK-29906 - [patch] update RLS to reflect the changes to
  the lists
  (Reported by Alexei Gradinari)
 * ASTERISK-30163 - general: fix minor formatting issues
 
  (Reported by N A)
 * ASTERISK-30164 - chan_iax2: Add missing option documentation

  (Reported by N A)
 * ASTERISK-30153 - logger: Improve log levels
  (Reported
  by N A)
 * ASTERISK-30160 - cdr.conf: Remove obsolete app_mysql
  reference
  (Reported by N A)
 * ASTERISK-30159 - general: Remove obsolete SVN references

  (Reported by N A)

For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0-rc1

Thank you for your continued support of Asterisk!
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[asterisk-dev] Asterisk 18.15.0-rc1 Now Available

2022-09-14 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the first
release candidate of Asterisk 18.15.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.15.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release candidate:

New Features made in this release:
---
 * ASTERISK-30037 - Add test support to calling external
  processes
  (Reported by Philip Prindeville)
 * ASTERISK-30161 - locks: add AMI event for deadlock
 
  (Reported by N A)
 * ASTERISK-30211 - app_confbridge: Add end_marked_any option
  
  (Reported by N A)
 * ASTERISK-30186 - res_pjsip: Add support for reloading TLS
  certificate and key information
  (Reported by Joshua C.
  Colp)
 * ASTERISK-29899 - features: Add advanced transfer initiation
  options
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-30135 - [res_musiconhold] Allows the moh only for
  the answered call
  (Reported by sungtae kim)
 * ASTERISK-26894 - pjsip should support tel uri scheme
 
  (Reported by Gergely DÃ¶msÃ¶di)
 * ASTERISK-30210 - func_frame_trace: Channel masquerade
  triggers assertion
  (Reported by N A)
 * ASTERISK-30190 - res_geolocation:  GEOLOC_PROFILE isn't
  returning correct values on incoming channel
  (Reported by
  George Joseph)
 * ASTERISK-29185 - chan_pjsip: Endpoint: allow = all is
  broken.
  (Reported by Alexander Traud)
 * ASTERISK-30192 - res_tonedetect: fix typo for frametype
 
  (Reported by N A)
 * ASTERISK-29453 - alembic: incoming_call_offer_pref and
  outgoing_call_offer_pref missing in "ps_endpoints" table
 
  (Reported by Daniel ThÃ¼men)
 * ASTERISK-26826 - testsuite: Add support for Python 3
 
  (Reported by Joshua C. Colp)
 * ASTERISK-30167 - res_geolocation:  Refactor for issues found
  by users
  (Reported by George Joseph)
 * ASTERISK-28422 - Memory Leak in Confbridge menu
 
  (Reported by Ted G)
 * ASTERISK-29917 - ami: FilterList action doesn't exist
 
  (Reported by N A)
 * ASTERISK-30018 - app_meetme: MeetmeList AMI event not
  documented
  (Reported by Michael Cargile)
 * ASTERISK-30020 - ConfbridgeListRooms Event Not Documented
   
  (Reported by Michael Cargile)
 * ASTERISK-30151 - Documentation doesn't include info about
  "field", a 3rd required parameter.
  (Reported by Chris
  Young)

Improvements made in this release:
---
 * ASTERISK-30178 - extend user_eq_phone behavior to local
  uri's
  (Reported by Michael Bradeen)
 * ASTERISK-30046 - Reimplement res/res_crypto.c internals with
  EVP_PKEY interface to Openssl API's
  (Reported by Philip
  Prindeville)
 * ASTERISK-30045 - Add test coverage to res/res_crypto.c
  functionality
  (Reported by Philip Prindeville)
 * ASTERISK-30185 - res_geolocation: Allow location parameters
  to be specified in profiles
  (Reported by George Joseph)
 * ASTERISK-30177 - res_geolocation:  Add option to suppress
  empty elements
  (Reported by George Joseph)
 * ASTERISK-30182 - res_geolocation: Add built-in profiles to
  use in fully dynamic configurations
  (Reported by George
  Joseph)
 * ASTERISK-29906 - [patch] update RLS to reflect the changes to
  the lists
  (Reported by Alexei Gradinari)
 * ASTERISK-30163 - general: fix minor formatting issues
 
  (Reported by N A)
 * ASTERISK-30164 - chan_iax2: Add missing option documentation

  (Reported by N A)
 * ASTERISK-30153 - logger: Improve log levels
  (Reported
  by N A)
 * ASTERISK-30160 - cdr.conf: Remove obsolete app_mysql
  reference
  (Reported by N A)
 * ASTERISK-30159 - general: Remove obsolete SVN references

  (Reported by N A)

For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0-rc1

Thank you for your continued support of Asterisk!
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[asterisk-dev] Asterisk 16.29.0-rc1 Now Available

2022-09-14 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the first
release candidate of Asterisk 16.29.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.29.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release candidate:

New Features made in this release:
---
 * ASTERISK-30037 - Add test support to calling external
  processes
  (Reported by Philip Prindeville)
 * ASTERISK-30161 - locks: add AMI event for deadlock
 
  (Reported by N A)
 * ASTERISK-30211 - app_confbridge: Add end_marked_any option
  
  (Reported by N A)
 * ASTERISK-30186 - res_pjsip: Add support for reloading TLS
  certificate and key information
  (Reported by Joshua C.
  Colp)
 * ASTERISK-29899 - features: Add advanced transfer initiation
  options
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-30135 - [res_musiconhold] Allows the moh only for
  the answered call
  (Reported by sungtae kim)
 * ASTERISK-26894 - pjsip should support tel uri scheme
 
  (Reported by Gergely DÃ¶msÃ¶di)
 * ASTERISK-30210 - func_frame_trace: Channel masquerade
  triggers assertion
  (Reported by N A)
 * ASTERISK-30190 - res_geolocation:  GEOLOC_PROFILE isn't
  returning correct values on incoming channel
  (Reported by
  George Joseph)
 * ASTERISK-29185 - chan_pjsip: Endpoint: allow = all is
  broken.
  (Reported by Alexander Traud)
 * ASTERISK-30192 - res_tonedetect: fix typo for frametype
 
  (Reported by N A)
 * ASTERISK-29453 - alembic: incoming_call_offer_pref and
  outgoing_call_offer_pref missing in "ps_endpoints" table
 
  (Reported by Daniel ThÃ¼men)
 * ASTERISK-26826 - testsuite: Add support for Python 3
 
  (Reported by Joshua C. Colp)
 * ASTERISK-30167 - res_geolocation:  Refactor for issues found
  by users
  (Reported by George Joseph)
 * ASTERISK-28422 - Memory Leak in Confbridge menu
 
  (Reported by Ted G)
 * ASTERISK-29917 - ami: FilterList action doesn't exist
 
  (Reported by N A)
 * ASTERISK-30020 - ConfbridgeListRooms Event Not Documented
   
  (Reported by Michael Cargile)
 * ASTERISK-30018 - app_meetme: MeetmeList AMI event not
  documented
  (Reported by Michael Cargile)
 * ASTERISK-30151 - Documentation doesn't include info about
  "field", a 3rd required parameter.
  (Reported by Chris
  Young)

Improvements made in this release:
---
 * ASTERISK-30178 - extend user_eq_phone behavior to local
  uri's
  (Reported by Michael Bradeen)
 * ASTERISK-30046 - Reimplement res/res_crypto.c internals with
  EVP_PKEY interface to Openssl API's
  (Reported by Philip
  Prindeville)
 * ASTERISK-30045 - Add test coverage to res/res_crypto.c
  functionality
  (Reported by Philip Prindeville)
 * ASTERISK-30185 - res_geolocation: Allow location parameters
  to be specified in profiles
  (Reported by George Joseph)
 * ASTERISK-30177 - res_geolocation:  Add option to suppress
  empty elements
  (Reported by George Joseph)
 * ASTERISK-30182 - res_geolocation: Add built-in profiles to
  use in fully dynamic configurations
  (Reported by George
  Joseph)
 * ASTERISK-29906 - [patch] update RLS to reflect the changes to
  the lists
  (Reported by Alexei Gradinari)
 * ASTERISK-30163 - general: fix minor formatting issues
 
  (Reported by N A)
 * ASTERISK-30164 - chan_iax2: Add missing option documentation

  (Reported by N A)
 * ASTERISK-30160 - cdr.conf: Remove obsolete app_mysql
  reference
  (Reported by N A)
 * ASTERISK-30159 - general: Remove obsolete SVN references

  (Reported by N A)
 * ASTERISK-30153 - logger: Improve log levels
  (Reported
  by N A)

For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.0-rc1

Thank you for your continued support of Asterisk!
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[asterisk-dev] Asterisk 19.6.0 Now Available

2022-08-18 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
19.6.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 19.6.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Improvements made in this release:
---
 * ASTERISK-30128 - Create PJSIP interface module for
  Geolocation
  (Reported by George Joseph)
 * ASTERISK-30127 - Create core Geolocation capability for
  Asterisk
  (Reported by George Joseph)
 * ASTERISK-30089 - general: fix typos
  (Reported by N A)
 * ASTERISK-30050 - Upgrade Asterisk to bundled pjproject
  2.12.1
  (Reported by Stanislav Abramenkov)

Bugs fixed in this release:
---
 * ASTERISK-30167 - res_geolocation:  Refactor for issues found
  by users
  (Reported by George Joseph)
 * ASTERISK-29966 - pbx_variables: ast_str_strlen can be wrong
 
  (Reported by N A)
 * ASTERISK-29905 - OSX: bininstall launchd issue on
  cross-platfrom build
  (Reported by Sergey V. Lobanov)
 * ASTERISK-30137 - manager: Global disabled event filtered is
  incomplete
  (Reported by N A)
 * ASTERISK-30109 - res_pjsip: no contact-status AMI event on
  register of prune-on-boot contact that uses the same URI as
  before Asterisk restart
  (Reported by Michael Neuhauser)
 * ASTERISK-30126 - Spelling mistake in
  configs/samples/queues.conf.sample
  (Reported by Sam Banks)
 * ASTERISK-29991 - chan_dahdi, callerid: Caller ID does not
  honor presentation
  (Reported by N A)
 * ASTERISK-30029 - build: Git security vulnerability fix is sad
  with our accessing git as root during "make install"
 
  (Reported by Joshua C. Colp)
 * ASTERISK-29907 - res_pjsip, app_confbridge: Video call
  through ConfBridge with normal endpoints causes infinite
  loop/crash
  (Reported by N A)
 * ASTERISK-30138 - Compile failure in
  res_geolocation/geoloc_eprofile.c when optimization is enabled
 
  (Reported by George Joseph)
 * ASTERISK-30096 -  cel_odbc: Column type 9 (field
  'cdr:cel:eventtime') is unsupported at this time
  (Reported
  by Morvai Szabolcs)
 * ASTERISK-30083 - chan_iax2: Optional dependency on
  openssl/res_crypto is now mandatory
  (Reported by Dmitry
  Melekhov)
 * ASTERISK-30099 - test_aeap_transport: transport_connect_fail
  sporadically causes failure
  (Reported by Kevin Harwell)
 * ASTERISK-30123 - features: Update automixmon documentation to
  reflect reality
  (Reported by Trevor Peirce)
 * ASTERISK-30117 - pbx_lua: Remove compiler warnings
 
  (Reported by Boris P. Korzun)
 * ASTERISK-30101 - res_prometheus: Optional load
  res_pjsip_outbound_registration.so
  (Reported by Boris P.
  Korzun)
 * ASTERISK-29989 - app_dial, chan_dahdi: DIALSTATUS is
  inconsistent for busy
  (Reported by N A)
 * ASTERISK-30106 - res_calendar_icalendar: Microsoft online ICS
  calendars no longer work
  (Reported by N A)
 * ASTERISK-30001 - db: Removing nonexistent entries shows
  "Database entry removed"
  (Reported by N A)
 * ASTERISK-29822 - cli: Typing \? freezes the CLI permanently
  with remote console
  (Reported by N A)
 * ASTERISK-30115 - app_dial: Allow hook flashes to propogate on
  outbound dials
  (Reported by N A)
 * ASTERISK-30075 - say: Abort if channel hangs up during
  playback
  (Reported by N A)
 * ASTERISK-30072 - res_pjsip: allow TLS verification of
  wildcard cert-bearing servers
  (Reported by Kevin Harwell)

New Features made in this release:
---
 * ASTERISK-30136 - db: Add AMI action to retrieve all keys
  beginning with a prefix
  (Reported by N A)
 * ASTERISK-3 - chan_dahdi: Add POLARITY function
 
  (Reported by N A)
 * ASTERISK-30062 - cli: Add CLI command to execute a dialplan
  app
  (Reported by N A)
 * ASTERISK-2 - pjsip: Get information from 200 OK INVITE
  reply headers
  (Reported by JosÃ© Lopes)
 * ASTERISK-30061 - pbx: Add pbx helper application
 
  (Reported by N A)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0

Thank you for your continued support of Asterisk!
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[asterisk-dev] Asterisk 18.14.0 Now Available

2022-08-18 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
18.14.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.14.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Improvements made in this release:
---
 * ASTERISK-30128 - Create PJSIP interface module for
  Geolocation
  (Reported by George Joseph)
 * ASTERISK-30127 - Create core Geolocation capability for
  Asterisk
  (Reported by George Joseph)
 * ASTERISK-30089 - general: fix typos
  (Reported by N A)
 * ASTERISK-30050 - Upgrade Asterisk to bundled pjproject
  2.12.1
  (Reported by Stanislav Abramenkov)

Bugs fixed in this release:
---
 * ASTERISK-30167 - res_geolocation:  Refactor for issues found
  by users
  (Reported by George Joseph)
 * ASTERISK-29966 - pbx_variables: ast_str_strlen can be wrong
 
  (Reported by N A)
 * ASTERISK-29905 - OSX: bininstall launchd issue on
  cross-platfrom build
  (Reported by Sergey V. Lobanov)
 * ASTERISK-30137 - manager: Global disabled event filtered is
  incomplete
  (Reported by N A)
 * ASTERISK-30109 - res_pjsip: no contact-status AMI event on
  register of prune-on-boot contact that uses the same URI as
  before Asterisk restart
  (Reported by Michael Neuhauser)
 * ASTERISK-29991 - chan_dahdi, callerid: Caller ID does not
  honor presentation
  (Reported by N A)
 * ASTERISK-30126 - Spelling mistake in
  configs/samples/queues.conf.sample
  (Reported by Sam Banks)
 * ASTERISK-30029 - build: Git security vulnerability fix is sad
  with our accessing git as root during "make install"
 
  (Reported by Joshua C. Colp)
 * ASTERISK-29907 - res_pjsip, app_confbridge: Video call
  through ConfBridge with normal endpoints causes infinite
  loop/crash
  (Reported by N A)
 * ASTERISK-30138 - Compile failure in
  res_geolocation/geoloc_eprofile.c when optimization is enabled
 
  (Reported by George Joseph)
 * ASTERISK-30096 -  cel_odbc: Column type 9 (field
  'cdr:cel:eventtime') is unsupported at this time
  (Reported
  by Morvai Szabolcs)
 * ASTERISK-30083 - chan_iax2: Optional dependency on
  openssl/res_crypto is now mandatory
  (Reported by Dmitry
  Melekhov)
 * ASTERISK-30099 - test_aeap_transport: transport_connect_fail
  sporadically causes failure
  (Reported by Kevin Harwell)
 * ASTERISK-30123 - features: Update automixmon documentation to
  reflect reality
  (Reported by Trevor Peirce)
 * ASTERISK-30117 - pbx_lua: Remove compiler warnings
 
  (Reported by Boris P. Korzun)
 * ASTERISK-30101 - res_prometheus: Optional load
  res_pjsip_outbound_registration.so
  (Reported by Boris P.
  Korzun)
 * ASTERISK-29989 - app_dial, chan_dahdi: DIALSTATUS is
  inconsistent for busy
  (Reported by N A)
 * ASTERISK-30001 - db: Removing nonexistent entries shows
  "Database entry removed"
  (Reported by N A)
 * ASTERISK-30115 - app_dial: Allow hook flashes to propogate on
  outbound dials
  (Reported by N A)
 * ASTERISK-30106 - res_calendar_icalendar: Microsoft online ICS
  calendars no longer work
  (Reported by N A)
 * ASTERISK-29822 - cli: Typing \? freezes the CLI permanently
  with remote console
  (Reported by N A)
 * ASTERISK-30072 - res_pjsip: allow TLS verification of
  wildcard cert-bearing servers
  (Reported by Kevin Harwell)
 * ASTERISK-30075 - say: Abort if channel hangs up during
  playback
  (Reported by N A)

New Features made in this release:
---
 * ASTERISK-30136 - db: Add AMI action to retrieve all keys
  beginning with a prefix
  (Reported by N A)
 * ASTERISK-3 - chan_dahdi: Add POLARITY function
 
  (Reported by N A)
 * ASTERISK-30062 - cli: Add CLI command to execute a dialplan
  app
  (Reported by N A)
 * ASTERISK-2 - pjsip: Get information from 200 OK INVITE
  reply headers
  (Reported by JosÃ© Lopes)
 * ASTERISK-30061 - pbx: Add pbx helper application
 
  (Reported by N A)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0

Thank you for your continued support of Asterisk!
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[asterisk-dev] Asterisk 16.28.0 Now Available

2022-08-18 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
16.28.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.28.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Improvements made in this release:
---
 * ASTERISK-30128 - Create PJSIP interface module for
  Geolocation
  (Reported by George Joseph)
 * ASTERISK-30127 - Create core Geolocation capability for
  Asterisk
  (Reported by George Joseph)
 * ASTERISK-30089 - general: fix typos
  (Reported by N A)
 * ASTERISK-30050 - Upgrade Asterisk to bundled pjproject
  2.12.1
  (Reported by Stanislav Abramenkov)

Bugs fixed in this release:
---
 * ASTERISK-30167 - res_geolocation:  Refactor for issues found
  by users
  (Reported by George Joseph)
 * ASTERISK-29966 - pbx_variables: ast_str_strlen can be wrong
 
  (Reported by N A)
 * ASTERISK-29905 - OSX: bininstall launchd issue on
  cross-platfrom build
  (Reported by Sergey V. Lobanov)
 * ASTERISK-30137 - manager: Global disabled event filtered is
  incomplete
  (Reported by N A)
 * ASTERISK-30109 - res_pjsip: no contact-status AMI event on
  register of prune-on-boot contact that uses the same URI as
  before Asterisk restart
  (Reported by Michael Neuhauser)
 * ASTERISK-30126 - Spelling mistake in
  configs/samples/queues.conf.sample
  (Reported by Sam Banks)
 * ASTERISK-29991 - chan_dahdi, callerid: Caller ID does not
  honor presentation
  (Reported by N A)
 * ASTERISK-29907 - res_pjsip, app_confbridge: Video call
  through ConfBridge with normal endpoints causes infinite
  loop/crash
  (Reported by N A)
 * ASTERISK-30029 - build: Git security vulnerability fix is sad
  with our accessing git as root during "make install"
 
  (Reported by Joshua C. Colp)
 * ASTERISK-30138 - Compile failure in
  res_geolocation/geoloc_eprofile.c when optimization is enabled
 
  (Reported by George Joseph)
 * ASTERISK-30096 -  cel_odbc: Column type 9 (field
  'cdr:cel:eventtime') is unsupported at this time
  (Reported
  by Morvai Szabolcs)
 * ASTERISK-30083 - chan_iax2: Optional dependency on
  openssl/res_crypto is now mandatory
  (Reported by Dmitry
  Melekhov)
 * ASTERISK-30123 - features: Update automixmon documentation to
  reflect reality
  (Reported by Trevor Peirce)
 * ASTERISK-30117 - pbx_lua: Remove compiler warnings
 
  (Reported by Boris P. Korzun)
 * ASTERISK-30001 - db: Removing nonexistent entries shows
  "Database entry removed"
  (Reported by N A)
 * ASTERISK-29822 - cli: Typing \? freezes the CLI permanently
  with remote console
  (Reported by N A)
 * ASTERISK-30106 - res_calendar_icalendar: Microsoft online ICS
  calendars no longer work
  (Reported by N A)
 * ASTERISK-30115 - app_dial: Allow hook flashes to propogate on
  outbound dials
  (Reported by N A)
 * ASTERISK-29989 - app_dial, chan_dahdi: DIALSTATUS is
  inconsistent for busy
  (Reported by N A)
 * ASTERISK-30072 - res_pjsip: allow TLS verification of
  wildcard cert-bearing servers
  (Reported by Kevin Harwell)
 * ASTERISK-30075 - say: Abort if channel hangs up during
  playback
  (Reported by N A)

New Features made in this release:
---
 * ASTERISK-30136 - db: Add AMI action to retrieve all keys
  beginning with a prefix
  (Reported by N A)
 * ASTERISK-3 - chan_dahdi: Add POLARITY function
 
  (Reported by N A)
 * ASTERISK-30062 - cli: Add CLI command to execute a dialplan
  app
  (Reported by N A)
 * ASTERISK-2 - pjsip: Get information from 200 OK INVITE
  reply headers
  (Reported by JosÃ© Lopes)
 * ASTERISK-30061 - pbx: Add pbx helper application
 
  (Reported by N A)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0

Thank you for your continued support of Asterisk!
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[asterisk-dev] Asterisk 19.6.0-rc2 Now Available

2022-08-11 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the second
release candidate of Asterisk 19.6.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 19.6.0-rc2 resolves an issue reported by the
community and would have not been possible without your participation.

Thank you!

The following issue is resolved in this release candidate:

Bugs fixed in this release:
---
 * ASTERISK-30167 - res_geolocation:  Refactor for issues found
  by users
  (Reported by George Joseph)

For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0-rc2

Thank you for your continued support of Asterisk!
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[asterisk-dev] Asterisk 18.14.0-rc2 Now Available

2022-08-11 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the second
release candidate of Asterisk 18.14.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.14.0-rc2 resolves an issue reported by the
community and would have not been possible without your participation.

Thank you!

The following issue is resolved in this release candidate:

Bugs fixed in this release:
---
 * ASTERISK-30167 - res_geolocation:  Refactor for issues found
  by users
  (Reported by George Joseph)

For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0-rc2

Thank you for your continued support of Asterisk!
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[asterisk-dev] Asterisk 16.28.0-rc2 Now Available

2022-08-11 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the second
release candidate of Asterisk 16.28.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.28.0-rc2 resolves an issue reported by the
community and would have not been possible without your participation.

Thank you!

The following issue is resolved in this release candidate:

Bugs fixed in this release:
---
 * ASTERISK-30167 - res_geolocation:  Refactor for issues found
  by users
  (Reported by George Joseph)

For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0-rc2

Thank you for your continued support of Asterisk!
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[asterisk-dev] Asterisk 19.6.0-rc1 Now Available

2022-07-28 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the first
release candidate of Asterisk 19.6.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 19.6.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release candidate:

Improvements made in this release:
---
 * ASTERISK-30128 - Create PJSIP interface module for
  Geolocation
  (Reported by George Joseph)
 * ASTERISK-30127 - Create core Geolocation capability for
  Asterisk
  (Reported by George Joseph)
 * ASTERISK-30089 - general: fix typos
  (Reported by N A)
 * ASTERISK-30050 - Upgrade Asterisk to bundled pjproject
  2.12.1
  (Reported by Stanislav Abramenkov)

Bugs fixed in this release:
---
 * ASTERISK-29966 - pbx_variables: ast_str_strlen can be wrong
 
  (Reported by N A)
 * ASTERISK-29905 - OSX: bininstall launchd issue on
  cross-platfrom build
  (Reported by Sergey V. Lobanov)
 * ASTERISK-30137 - manager: Global disabled event filtered is
  incomplete
  (Reported by N A)
 * ASTERISK-30109 - res_pjsip: no contact-status AMI event on
  register of prune-on-boot contact that uses the same URI as
  before Asterisk restart
  (Reported by Michael Neuhauser)
 * ASTERISK-30126 - Spelling mistake in
  configs/samples/queues.conf.sample
  (Reported by Sam Banks)
 * ASTERISK-29991 - chan_dahdi, callerid: Caller ID does not
  honor presentation
  (Reported by N A)
 * ASTERISK-30029 - build: Git security vulnerability fix is sad
  with our accessing git as root during "make install"
 
  (Reported by Joshua C. Colp)
 * ASTERISK-29907 - res_pjsip, app_confbridge: Video call
  through ConfBridge with normal endpoints causes infinite
  loop/crash
  (Reported by N A)
 * ASTERISK-30138 - Compile failure in
  res_geolocation/geoloc_eprofile.c when optimization is enabled
 
  (Reported by George Joseph)
 * ASTERISK-30096 -  cel_odbc: Column type 9 (field
  'cdr:cel:eventtime') is unsupported at this time
  (Reported
  by Morvai Szabolcs)
 * ASTERISK-30083 - chan_iax2: Optional dependency on
  openssl/res_crypto is now mandatory
  (Reported by Dmitry
  Melekhov)
 * ASTERISK-30099 - test_aeap_transport: transport_connect_fail
  sporadically causes failure
  (Reported by Kevin Harwell)
 * ASTERISK-30123 - features: Update automixmon documentation to
  reflect reality
  (Reported by Trevor Peirce)
 * ASTERISK-30117 - pbx_lua: Remove compiler warnings
 
  (Reported by Boris P. Korzun)
 * ASTERISK-30101 - res_prometheus: Optional load
  res_pjsip_outbound_registration.so
  (Reported by Boris P.
  Korzun)
 * ASTERISK-29989 - app_dial, chan_dahdi: DIALSTATUS is
  inconsistent for busy
  (Reported by N A)
 * ASTERISK-30106 - res_calendar_icalendar: Microsoft online ICS
  calendars no longer work
  (Reported by N A)
 * ASTERISK-30001 - db: Removing nonexistent entries shows
  "Database entry removed"
  (Reported by N A)
 * ASTERISK-29822 - cli: Typing \? freezes the CLI permanently
  with remote console
  (Reported by N A)
 * ASTERISK-30115 - app_dial: Allow hook flashes to propogate on
  outbound dials
  (Reported by N A)
 * ASTERISK-30075 - say: Abort if channel hangs up during
  playback
  (Reported by N A)
 * ASTERISK-30072 - res_pjsip: allow TLS verification of
  wildcard cert-bearing servers
  (Reported by Kevin Harwell)

New Features made in this release:
---
 * ASTERISK-30136 - db: Add AMI action to retrieve all keys
  beginning with a prefix
  (Reported by N A)
 * ASTERISK-3 - chan_dahdi: Add POLARITY function
 
  (Reported by N A)
 * ASTERISK-30062 - cli: Add CLI command to execute a dialplan
  app
  (Reported by N A)
 * ASTERISK-2 - pjsip: Get information from 200 OK INVITE
  reply headers
  (Reported by JosÃ© Lopes)
 * ASTERISK-30061 - pbx: Add pbx helper application
 
  (Reported by N A)

For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0-rc1

Thank you for your continued support of Asterisk!
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[asterisk-dev] Asterisk 18.14.0-rc1 Now Available

2022-07-28 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the first
release candidate of Asterisk 18.14.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.14.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release candidate:

Improvements made in this release:
---
 * ASTERISK-30128 - Create PJSIP interface module for
  Geolocation
  (Reported by George Joseph)
 * ASTERISK-30127 - Create core Geolocation capability for
  Asterisk
  (Reported by George Joseph)
 * ASTERISK-30089 - general: fix typos
  (Reported by N A)
 * ASTERISK-30050 - Upgrade Asterisk to bundled pjproject
  2.12.1
  (Reported by Stanislav Abramenkov)

Bugs fixed in this release:
---
 * ASTERISK-29966 - pbx_variables: ast_str_strlen can be wrong
 
  (Reported by N A)
 * ASTERISK-29905 - OSX: bininstall launchd issue on
  cross-platfrom build
  (Reported by Sergey V. Lobanov)
 * ASTERISK-30137 - manager: Global disabled event filtered is
  incomplete
  (Reported by N A)
 * ASTERISK-30109 - res_pjsip: no contact-status AMI event on
  register of prune-on-boot contact that uses the same URI as
  before Asterisk restart
  (Reported by Michael Neuhauser)
 * ASTERISK-29991 - chan_dahdi, callerid: Caller ID does not
  honor presentation
  (Reported by N A)
 * ASTERISK-30126 - Spelling mistake in
  configs/samples/queues.conf.sample
  (Reported by Sam Banks)
 * ASTERISK-30029 - build: Git security vulnerability fix is sad
  with our accessing git as root during "make install"
 
  (Reported by Joshua C. Colp)
 * ASTERISK-29907 - res_pjsip, app_confbridge: Video call
  through ConfBridge with normal endpoints causes infinite
  loop/crash
  (Reported by N A)
 * ASTERISK-30138 - Compile failure in
  res_geolocation/geoloc_eprofile.c when optimization is enabled
 
  (Reported by George Joseph)
 * ASTERISK-30096 -  cel_odbc: Column type 9 (field
  'cdr:cel:eventtime') is unsupported at this time
  (Reported
  by Morvai Szabolcs)
 * ASTERISK-30083 - chan_iax2: Optional dependency on
  openssl/res_crypto is now mandatory
  (Reported by Dmitry
  Melekhov)
 * ASTERISK-30099 - test_aeap_transport: transport_connect_fail
  sporadically causes failure
  (Reported by Kevin Harwell)
 * ASTERISK-30123 - features: Update automixmon documentation to
  reflect reality
  (Reported by Trevor Peirce)
 * ASTERISK-30117 - pbx_lua: Remove compiler warnings
 
  (Reported by Boris P. Korzun)
 * ASTERISK-30101 - res_prometheus: Optional load
  res_pjsip_outbound_registration.so
  (Reported by Boris P.
  Korzun)
 * ASTERISK-29989 - app_dial, chan_dahdi: DIALSTATUS is
  inconsistent for busy
  (Reported by N A)
 * ASTERISK-30001 - db: Removing nonexistent entries shows
  "Database entry removed"
  (Reported by N A)
 * ASTERISK-30115 - app_dial: Allow hook flashes to propogate on
  outbound dials
  (Reported by N A)
 * ASTERISK-30106 - res_calendar_icalendar: Microsoft online ICS
  calendars no longer work
  (Reported by N A)
 * ASTERISK-29822 - cli: Typing \? freezes the CLI permanently
  with remote console
  (Reported by N A)
 * ASTERISK-30072 - res_pjsip: allow TLS verification of
  wildcard cert-bearing servers
  (Reported by Kevin Harwell)
 * ASTERISK-30075 - say: Abort if channel hangs up during
  playback
  (Reported by N A)

New Features made in this release:
---
 * ASTERISK-30136 - db: Add AMI action to retrieve all keys
  beginning with a prefix
  (Reported by N A)
 * ASTERISK-3 - chan_dahdi: Add POLARITY function
 
  (Reported by N A)
 * ASTERISK-30062 - cli: Add CLI command to execute a dialplan
  app
  (Reported by N A)
 * ASTERISK-2 - pjsip: Get information from 200 OK INVITE
  reply headers
  (Reported by JosÃ© Lopes)
 * ASTERISK-30061 - pbx: Add pbx helper application
 
  (Reported by N A)

For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0-rc1

Thank you for your continued support of Asterisk!
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[asterisk-dev] Asterisk 16.28.0-rc1 Now Available

2022-07-28 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the first
release candidate of Asterisk 16.28.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.28.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release candidate:

Improvements made in this release:
---
 * ASTERISK-30128 - Create PJSIP interface module for
  Geolocation
  (Reported by George Joseph)
 * ASTERISK-30127 - Create core Geolocation capability for
  Asterisk
  (Reported by George Joseph)
 * ASTERISK-30089 - general: fix typos
  (Reported by N A)
 * ASTERISK-30050 - Upgrade Asterisk to bundled pjproject
  2.12.1
  (Reported by Stanislav Abramenkov)

Bugs fixed in this release:
---
 * ASTERISK-29966 - pbx_variables: ast_str_strlen can be wrong
 
  (Reported by N A)
 * ASTERISK-29905 - OSX: bininstall launchd issue on
  cross-platfrom build
  (Reported by Sergey V. Lobanov)
 * ASTERISK-30137 - manager: Global disabled event filtered is
  incomplete
  (Reported by N A)
 * ASTERISK-30109 - res_pjsip: no contact-status AMI event on
  register of prune-on-boot contact that uses the same URI as
  before Asterisk restart
  (Reported by Michael Neuhauser)
 * ASTERISK-30126 - Spelling mistake in
  configs/samples/queues.conf.sample
  (Reported by Sam Banks)
 * ASTERISK-29991 - chan_dahdi, callerid: Caller ID does not
  honor presentation
  (Reported by N A)
 * ASTERISK-29907 - res_pjsip, app_confbridge: Video call
  through ConfBridge with normal endpoints causes infinite
  loop/crash
  (Reported by N A)
 * ASTERISK-30029 - build: Git security vulnerability fix is sad
  with our accessing git as root during "make install"
 
  (Reported by Joshua C. Colp)
 * ASTERISK-30138 - Compile failure in
  res_geolocation/geoloc_eprofile.c when optimization is enabled
 
  (Reported by George Joseph)
 * ASTERISK-30096 -  cel_odbc: Column type 9 (field
  'cdr:cel:eventtime') is unsupported at this time
  (Reported
  by Morvai Szabolcs)
 * ASTERISK-30083 - chan_iax2: Optional dependency on
  openssl/res_crypto is now mandatory
  (Reported by Dmitry
  Melekhov)
 * ASTERISK-30123 - features: Update automixmon documentation to
  reflect reality
  (Reported by Trevor Peirce)
 * ASTERISK-30117 - pbx_lua: Remove compiler warnings
 
  (Reported by Boris P. Korzun)
 * ASTERISK-30001 - db: Removing nonexistent entries shows
  "Database entry removed"
  (Reported by N A)
 * ASTERISK-29822 - cli: Typing \? freezes the CLI permanently
  with remote console
  (Reported by N A)
 * ASTERISK-30106 - res_calendar_icalendar: Microsoft online ICS
  calendars no longer work
  (Reported by N A)
 * ASTERISK-30115 - app_dial: Allow hook flashes to propogate on
  outbound dials
  (Reported by N A)
 * ASTERISK-29989 - app_dial, chan_dahdi: DIALSTATUS is
  inconsistent for busy
  (Reported by N A)
 * ASTERISK-30072 - res_pjsip: allow TLS verification of
  wildcard cert-bearing servers
  (Reported by Kevin Harwell)
 * ASTERISK-30075 - say: Abort if channel hangs up during
  playback
  (Reported by N A)

New Features made in this release:
---
 * ASTERISK-30136 - db: Add AMI action to retrieve all keys
  beginning with a prefix
  (Reported by N A)
 * ASTERISK-3 - chan_dahdi: Add POLARITY function
 
  (Reported by N A)
 * ASTERISK-30062 - cli: Add CLI command to execute a dialplan
  app
  (Reported by N A)
 * ASTERISK-2 - pjsip: Get information from 200 OK INVITE
  reply headers
  (Reported by JosÃ© Lopes)
 * ASTERISK-30061 - pbx: Add pbx helper application
 
  (Reported by N A)

For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0-rc1

Thank you for your continued support of Asterisk!
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[asterisk-dev] Asterisk 19.5.0 Now Available

2022-06-23 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
19.5.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 19.5.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Improvements made in this release:
---
 * ASTERISK-29906 - [patch] update RLS to reflect the changes to
  the lists
  (Reported by Alexei Gradinari)
 * ASTERISK-29891 - [patch] provide a display name for RLS
  subscriptions
  (Reported by Alexei Gradinari)
 * ASTERISK-30090 - xmldocs: Use example tags for examples
 
  (Reported by N A)
 * ASTERISK-30086 - res_parking: Warn when invalid parking space
  requested
  (Reported by N A)
 * ASTERISK-30058 - Evaluate dialplan functions and variables in
  agi exec
  (Reported by Shloime Rosenblum)
 * ASTERISK-30027 - ari: expose channel driver's unique id (i.e.
  Call-ID for chan_sip/chan_pjsip) in ARI channel resource
 
  (Reported by Moritz Fain)
 * ASTERISK-29845 - res_pjsip_outbound_registration: Show time
  remaining until registration lapses
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-30097 - console: Recent documentation changes for
  connecting to remote console are inconsistent
  (Reported by
  Matthias Hensler)
 * ASTERISK-30043 - Wrong party is disconnected when
  hook-flashing on 3-way bridge
  (Reported by Josh Alberts)
 * ASTERISK-29603 - res_pjsip: UPDATE/re-INVITE not sent when
  "timers=always" is specified in pjsip.conf
  (Reported by
  Ray Crumrine)
 * ASTERISK-30092 - DateTime application: wrong inflection for
  one o'clock in German
  (Reported by Christof Efkemann)
 * ASTERISK-29981 - res_calendar: Asterisk crashes when
  starting, and will not run
  (Reported by N A)
 * ASTERISK-30064 - pbx: iax2 switch causes crash due to
  deadlock and assertion
  (Reported by N A)
 * ASTERISK-30039 - cli: Targeted debug on startup deadlocks and
  creates unstable system
  (Reported by N A)
 * ASTERISK-30051 - res_pjsip: No video after un-hold with
  moh_passthrough=yes
  (Reported by Maximilian Fridrich)
 * ASTERISK-24601 - [patch]Missing RFC4235 tags and attributes
  in PJSIP NOTIFY event: dialog  XML body
  (Reported by Marco
  Paland)
 * ASTERISK-30059 - menuselect: libxml include fails under
  Gentoo
  (Reported by waltermoeller)
 * ASTERISK-30060 - loader: format warnings in dev mode
 
  (Reported by N A)
 * ASTERISK-30065 - pjsip: Open Websocket connection is not
  reused for outgoing requests
  (Reported by LA)
 * ASTERISK-30042 - res_pjsip_transport_websocket: Registration
  over websocket returns a rewritten contact
  (Reported by
  Thomas Guebels)
 * ASTERISK-29993 - chan_dahdi: Operator control option borks
  both lines involved on callee disconnect
  (Reported by N A)
 * ASTERISK-30044 - GCC 12 issues
  (Reported by George
  Joseph)

New Features made in this release:
---
 * ASTERISK-30063 - app_voicemail: Add option to prevent
  deletion of messages
  (Reported by N A)
 * ASTERISK-29965 - res_pjsip_outbound_registration: Make max
  registration delay configurable
  (Reported by N A)
 * ASTERISK-30087 - res_parking: Add music on hold override
  option
  (Reported by N A)
 * ASTERISK-30036 - app_confbridge: Add CONFBRIDGE_CHANNELS
  function
  (Reported by N A)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0

Thank you for your continued support of Asterisk!
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[asterisk-dev] Asterisk 18.13.0 Now Available

2022-06-23 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
18.13.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.13.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Improvements made in this release:
---
 * ASTERISK-29906 - [patch] update RLS to reflect the changes to
  the lists
  (Reported by Alexei Gradinari)
 * ASTERISK-29891 - [patch] provide a display name for RLS
  subscriptions
  (Reported by Alexei Gradinari)
 * ASTERISK-30090 - xmldocs: Use example tags for examples
 
  (Reported by N A)
 * ASTERISK-30086 - res_parking: Warn when invalid parking space
  requested
  (Reported by N A)
 * ASTERISK-30058 - Evaluate dialplan functions and variables in
  agi exec
  (Reported by Shloime Rosenblum)
 * ASTERISK-30027 - ari: expose channel driver's unique id (i.e.
  Call-ID for chan_sip/chan_pjsip) in ARI channel resource
 
  (Reported by Moritz Fain)
 * ASTERISK-29845 - res_pjsip_outbound_registration: Show time
  remaining until registration lapses
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-30097 - console: Recent documentation changes for
  connecting to remote console are inconsistent
  (Reported by
  Matthias Hensler)
 * ASTERISK-30043 - Wrong party is disconnected when
  hook-flashing on 3-way bridge
  (Reported by Josh Alberts)
 * ASTERISK-29603 - res_pjsip: UPDATE/re-INVITE not sent when
  "timers=always" is specified in pjsip.conf
  (Reported by
  Ray Crumrine)
 * ASTERISK-30092 - DateTime application: wrong inflection for
  one o'clock in German
  (Reported by Christof Efkemann)
 * ASTERISK-30064 - pbx: iax2 switch causes crash due to
  deadlock and assertion
  (Reported by N A)
 * ASTERISK-29981 - res_calendar: Asterisk crashes when
  starting, and will not run
  (Reported by N A)
 * ASTERISK-30039 - cli: Targeted debug on startup deadlocks and
  creates unstable system
  (Reported by N A)
 * ASTERISK-30051 - res_pjsip: No video after un-hold with
  moh_passthrough=yes
  (Reported by Maximilian Fridrich)
 * ASTERISK-24601 - [patch]Missing RFC4235 tags and attributes
  in PJSIP NOTIFY event: dialog  XML body
  (Reported by Marco
  Paland)
 * ASTERISK-30059 - menuselect: libxml include fails under
  Gentoo
  (Reported by waltermoeller)
 * ASTERISK-30060 - loader: format warnings in dev mode
 
  (Reported by N A)
 * ASTERISK-30065 - pjsip: Open Websocket connection is not
  reused for outgoing requests
  (Reported by LA)
 * ASTERISK-30042 - res_pjsip_transport_websocket: Registration
  over websocket returns a rewritten contact
  (Reported by
  Thomas Guebels)
 * ASTERISK-29993 - chan_dahdi: Operator control option borks
  both lines involved on callee disconnect
  (Reported by N A)
 * ASTERISK-30044 - GCC 12 issues
  (Reported by George
  Joseph)

New Features made in this release:
---
 * ASTERISK-30063 - app_voicemail: Add option to prevent
  deletion of messages
  (Reported by N A)
 * ASTERISK-29965 - res_pjsip_outbound_registration: Make max
  registration delay configurable
  (Reported by N A)
 * ASTERISK-30087 - res_parking: Add music on hold override
  option
  (Reported by N A)
 * ASTERISK-30036 - app_confbridge: Add CONFBRIDGE_CHANNELS
  function
  (Reported by N A)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0

Thank you for your continued support of Asterisk!
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[asterisk-dev] Asterisk 16.27.0 Now Available

2022-06-23 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
16.27.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.27.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Improvements made in this release:
---
 * ASTERISK-30090 - xmldocs: Use example tags for examples
 
  (Reported by N A)
 * ASTERISK-29906 - [patch] update RLS to reflect the changes to
  the lists
  (Reported by Alexei Gradinari)
 * ASTERISK-29891 - [patch] provide a display name for RLS
  subscriptions
  (Reported by Alexei Gradinari)
 * ASTERISK-30086 - res_parking: Warn when invalid parking space
  requested
  (Reported by N A)
 * ASTERISK-30058 - Evaluate dialplan functions and variables in
  agi exec
  (Reported by Shloime Rosenblum)
 * ASTERISK-30027 - ari: expose channel driver's unique id (i.e.
  Call-ID for chan_sip/chan_pjsip) in ARI channel resource
 
  (Reported by Moritz Fain)
 * ASTERISK-29845 - res_pjsip_outbound_registration: Show time
  remaining until registration lapses
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-30097 - console: Recent documentation changes for
  connecting to remote console are inconsistent
  (Reported by
  Matthias Hensler)
 * ASTERISK-30043 - Wrong party is disconnected when
  hook-flashing on 3-way bridge
  (Reported by Josh Alberts)
 * ASTERISK-29603 - res_pjsip: UPDATE/re-INVITE not sent when
  "timers=always" is specified in pjsip.conf
  (Reported by
  Ray Crumrine)
 * ASTERISK-30092 - DateTime application: wrong inflection for
  one o'clock in German
  (Reported by Christof Efkemann)
 * ASTERISK-30064 - pbx: iax2 switch causes crash due to
  deadlock and assertion
  (Reported by N A)
 * ASTERISK-29981 - res_calendar: Asterisk crashes when
  starting, and will not run
  (Reported by N A)
 * ASTERISK-30039 - cli: Targeted debug on startup deadlocks and
  creates unstable system
  (Reported by N A)
 * ASTERISK-30051 - res_pjsip: No video after un-hold with
  moh_passthrough=yes
  (Reported by Maximilian Fridrich)
 * ASTERISK-24601 - [patch]Missing RFC4235 tags and attributes
  in PJSIP NOTIFY event: dialog  XML body
  (Reported by Marco
  Paland)
 * ASTERISK-30060 - loader: format warnings in dev mode
 
  (Reported by N A)
 * ASTERISK-30059 - menuselect: libxml include fails under
  Gentoo
  (Reported by waltermoeller)
 * ASTERISK-30065 - pjsip: Open Websocket connection is not
  reused for outgoing requests
  (Reported by LA)
 * ASTERISK-30042 - res_pjsip_transport_websocket: Registration
  over websocket returns a rewritten contact
  (Reported by
  Thomas Guebels)
 * ASTERISK-29993 - chan_dahdi: Operator control option borks
  both lines involved on callee disconnect
  (Reported by N A)
 * ASTERISK-30044 - GCC 12 issues
  (Reported by George
  Joseph)

New Features made in this release:
---
 * ASTERISK-30063 - app_voicemail: Add option to prevent
  deletion of messages
  (Reported by N A)
 * ASTERISK-30087 - res_parking: Add music on hold override
  option
  (Reported by N A)
 * ASTERISK-29965 - res_pjsip_outbound_registration: Make max
  registration delay configurable
  (Reported by N A)
 * ASTERISK-30036 - app_confbridge: Add CONFBRIDGE_CHANNELS
  function
  (Reported by N A)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0

Thank you for your continued support of Asterisk!
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[asterisk-dev] Asterisk 18.13.0-rc1 Now Available

2022-06-16 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the first
release candidate of Asterisk 18.13.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.13.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release candidate:

Improvements made in this release:
---
 * ASTERISK-29906 - [patch] update RLS to reflect the changes to
  the lists
  (Reported by Alexei Gradinari)
 * ASTERISK-29891 - [patch] provide a display name for RLS
  subscriptions
  (Reported by Alexei Gradinari)
 * ASTERISK-30090 - xmldocs: Use example tags for examples
 
  (Reported by N A)
 * ASTERISK-30086 - res_parking: Warn when invalid parking space
  requested
  (Reported by N A)
 * ASTERISK-30058 - Evaluate dialplan functions and variables in
  agi exec
  (Reported by Shloime Rosenblum)
 * ASTERISK-30027 - ari: expose channel driver's unique id (i.e.
  Call-ID for chan_sip/chan_pjsip) in ARI channel resource
 
  (Reported by Moritz Fain)
 * ASTERISK-29845 - res_pjsip_outbound_registration: Show time
  remaining until registration lapses
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-30097 - console: Recent documentation changes for
  connecting to remote console are inconsistent
  (Reported by
  Matthias Hensler)
 * ASTERISK-30043 - Wrong party is disconnected when
  hook-flashing on 3-way bridge
  (Reported by Josh Alberts)
 * ASTERISK-29603 - res_pjsip: UPDATE/re-INVITE not sent when
  "timers=always" is specified in pjsip.conf
  (Reported by
  Ray Crumrine)
 * ASTERISK-30092 - DateTime application: wrong inflection for
  one o'clock in German
  (Reported by Christof Efkemann)
 * ASTERISK-30064 - pbx: iax2 switch causes crash due to
  deadlock and assertion
  (Reported by N A)
 * ASTERISK-29981 - res_calendar: Asterisk crashes when
  starting, and will not run
  (Reported by N A)
 * ASTERISK-30039 - cli: Targeted debug on startup deadlocks and
  creates unstable system
  (Reported by N A)
 * ASTERISK-30051 - res_pjsip: No video after un-hold with
  moh_passthrough=yes
  (Reported by Maximilian Fridrich)
 * ASTERISK-24601 - [patch]Missing RFC4235 tags and attributes
  in PJSIP NOTIFY event: dialog  XML body
  (Reported by Marco
  Paland)
 * ASTERISK-30059 - menuselect: libxml include fails under
  Gentoo
  (Reported by waltermoeller)
 * ASTERISK-30060 - loader: format warnings in dev mode
 
  (Reported by N A)
 * ASTERISK-30065 - pjsip: Open Websocket connection is not
  reused for outgoing requests
  (Reported by LA)
 * ASTERISK-30042 - res_pjsip_transport_websocket: Registration
  over websocket returns a rewritten contact
  (Reported by
  Thomas Guebels)
 * ASTERISK-29993 - chan_dahdi: Operator control option borks
  both lines involved on callee disconnect
  (Reported by N A)
 * ASTERISK-30044 - GCC 12 issues
  (Reported by George
  Joseph)

New Features made in this release:
---
 * ASTERISK-30063 - app_voicemail: Add option to prevent
  deletion of messages
  (Reported by N A)
 * ASTERISK-29965 - res_pjsip_outbound_registration: Make max
  registration delay configurable
  (Reported by N A)
 * ASTERISK-30087 - res_parking: Add music on hold override
  option
  (Reported by N A)
 * ASTERISK-30036 - app_confbridge: Add CONFBRIDGE_CHANNELS
  function
  (Reported by N A)

For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0-rc1

Thank you for your continued support of Asterisk!
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[asterisk-dev] Asterisk 19.5.0-rc1 Now Available

2022-06-16 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the first
release candidate of Asterisk 19.5.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 19.5.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release candidate:

Improvements made in this release:
---
 * ASTERISK-29906 - [patch] update RLS to reflect the changes to
  the lists
  (Reported by Alexei Gradinari)
 * ASTERISK-29891 - [patch] provide a display name for RLS
  subscriptions
  (Reported by Alexei Gradinari)
 * ASTERISK-30090 - xmldocs: Use example tags for examples
 
  (Reported by N A)
 * ASTERISK-30086 - res_parking: Warn when invalid parking space
  requested
  (Reported by N A)
 * ASTERISK-30058 - Evaluate dialplan functions and variables in
  agi exec
  (Reported by Shloime Rosenblum)
 * ASTERISK-30027 - ari: expose channel driver's unique id (i.e.
  Call-ID for chan_sip/chan_pjsip) in ARI channel resource
 
  (Reported by Moritz Fain)
 * ASTERISK-29845 - res_pjsip_outbound_registration: Show time
  remaining until registration lapses
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-30097 - console: Recent documentation changes for
  connecting to remote console are inconsistent
  (Reported by
  Matthias Hensler)
 * ASTERISK-30043 - Wrong party is disconnected when
  hook-flashing on 3-way bridge
  (Reported by Josh Alberts)
 * ASTERISK-29603 - res_pjsip: UPDATE/re-INVITE not sent when
  "timers=always" is specified in pjsip.conf
  (Reported by
  Ray Crumrine)
 * ASTERISK-30092 - DateTime application: wrong inflection for
  one o'clock in German
  (Reported by Christof Efkemann)
 * ASTERISK-29981 - res_calendar: Asterisk crashes when
  starting, and will not run
  (Reported by N A)
 * ASTERISK-30064 - pbx: iax2 switch causes crash due to
  deadlock and assertion
  (Reported by N A)
 * ASTERISK-30039 - cli: Targeted debug on startup deadlocks and
  creates unstable system
  (Reported by N A)
 * ASTERISK-30051 - res_pjsip: No video after un-hold with
  moh_passthrough=yes
  (Reported by Maximilian Fridrich)
 * ASTERISK-24601 - [patch]Missing RFC4235 tags and attributes
  in PJSIP NOTIFY event: dialog  XML body
  (Reported by Marco
  Paland)
 * ASTERISK-30059 - menuselect: libxml include fails under
  Gentoo
  (Reported by waltermoeller)
 * ASTERISK-30060 - loader: format warnings in dev mode
 
  (Reported by N A)
 * ASTERISK-30065 - pjsip: Open Websocket connection is not
  reused for outgoing requests
  (Reported by LA)
 * ASTERISK-30042 - res_pjsip_transport_websocket: Registration
  over websocket returns a rewritten contact
  (Reported by
  Thomas Guebels)
 * ASTERISK-29993 - chan_dahdi: Operator control option borks
  both lines involved on callee disconnect
  (Reported by N A)
 * ASTERISK-30044 - GCC 12 issues
  (Reported by George
  Joseph)

New Features made in this release:
---
 * ASTERISK-30063 - app_voicemail: Add option to prevent
  deletion of messages
  (Reported by N A)
 * ASTERISK-29965 - res_pjsip_outbound_registration: Make max
  registration delay configurable
  (Reported by N A)
 * ASTERISK-30087 - res_parking: Add music on hold override
  option
  (Reported by N A)
 * ASTERISK-30036 - app_confbridge: Add CONFBRIDGE_CHANNELS
  function
  (Reported by N A)

For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0-rc1

Thank you for your continued support of Asterisk!
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[asterisk-dev] Asterisk 16.27.0-rc1 Now Available

2022-06-16 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the first
release candidate of Asterisk 16.27.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.27.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release candidate:

Improvements made in this release:
---
 * ASTERISK-30090 - xmldocs: Use example tags for examples
 
  (Reported by N A)
 * ASTERISK-29906 - [patch] update RLS to reflect the changes to
  the lists
  (Reported by Alexei Gradinari)
 * ASTERISK-29891 - [patch] provide a display name for RLS
  subscriptions
  (Reported by Alexei Gradinari)
 * ASTERISK-30086 - res_parking: Warn when invalid parking space
  requested
  (Reported by N A)
 * ASTERISK-30058 - Evaluate dialplan functions and variables in
  agi exec
  (Reported by Shloime Rosenblum)
 * ASTERISK-30027 - ari: expose channel driver's unique id (i.e.
  Call-ID for chan_sip/chan_pjsip) in ARI channel resource
 
  (Reported by Moritz Fain)
 * ASTERISK-29845 - res_pjsip_outbound_registration: Show time
  remaining until registration lapses
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-30097 - console: Recent documentation changes for
  connecting to remote console are inconsistent
  (Reported by
  Matthias Hensler)
 * ASTERISK-30043 - Wrong party is disconnected when
  hook-flashing on 3-way bridge
  (Reported by Josh Alberts)
 * ASTERISK-29603 - res_pjsip: UPDATE/re-INVITE not sent when
  "timers=always" is specified in pjsip.conf
  (Reported by
  Ray Crumrine)
 * ASTERISK-30092 - DateTime application: wrong inflection for
  one o'clock in German
  (Reported by Christof Efkemann)
 * ASTERISK-30064 - pbx: iax2 switch causes crash due to
  deadlock and assertion
  (Reported by N A)
 * ASTERISK-29981 - res_calendar: Asterisk crashes when
  starting, and will not run
  (Reported by N A)
 * ASTERISK-30039 - cli: Targeted debug on startup deadlocks and
  creates unstable system
  (Reported by N A)
 * ASTERISK-30051 - res_pjsip: No video after un-hold with
  moh_passthrough=yes
  (Reported by Maximilian Fridrich)
 * ASTERISK-24601 - [patch]Missing RFC4235 tags and attributes
  in PJSIP NOTIFY event: dialog  XML body
  (Reported by Marco
  Paland)
 * ASTERISK-30060 - loader: format warnings in dev mode
 
  (Reported by N A)
 * ASTERISK-30059 - menuselect: libxml include fails under
  Gentoo
  (Reported by waltermoeller)
 * ASTERISK-30065 - pjsip: Open Websocket connection is not
  reused for outgoing requests
  (Reported by LA)
 * ASTERISK-30042 - res_pjsip_transport_websocket: Registration
  over websocket returns a rewritten contact
  (Reported by
  Thomas Guebels)
 * ASTERISK-29993 - chan_dahdi: Operator control option borks
  both lines involved on callee disconnect
  (Reported by N A)
 * ASTERISK-30044 - GCC 12 issues
  (Reported by George
  Joseph)

New Features made in this release:
---
 * ASTERISK-30063 - app_voicemail: Add option to prevent
  deletion of messages
  (Reported by N A)
 * ASTERISK-30087 - res_parking: Add music on hold override
  option
  (Reported by N A)
 * ASTERISK-29965 - res_pjsip_outbound_registration: Make max
  registration delay configurable
  (Reported by N A)
 * ASTERISK-30036 - app_confbridge: Add CONFBRIDGE_CHANNELS
  function
  (Reported by N A)

For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0-rc1

Thank you for your continued support of Asterisk!
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[asterisk-dev] Asterisk 19.4.1 Now Available

2022-05-19 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
19.4.1.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 19.4.1 resolves an issue reported by the
community and would have not been possible without your participation.

Thank you!

The following issue is resolved in this release:

Bugs fixed in this release:
---
 * ASTERISK-30065 - pjsip: Open Websocket connection is not
  reused for outgoing requests
  (Reported by LA)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1

Thank you for your continued support of Asterisk!
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[asterisk-dev] Asterisk 18.12.1 Now Available

2022-05-19 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
18.12.1.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.12.1 resolves an issue reported by the
community and would have not been possible without your participation.

Thank you!

The following issue is resolved in this release:

Bugs fixed in this release:
---
 * ASTERISK-30065 - pjsip: Open Websocket connection is not
  reused for outgoing requests
  (Reported by LA)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1

Thank you for your continued support of Asterisk!
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[asterisk-dev] Asterisk 16.26.1 Now Available

2022-05-19 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
16.26.1.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.26.1 resolves an issue reported by the
community and would have not been possible without your participation.

Thank you!

The following issue is resolved in this release:

Bugs fixed in this release:
---
 * ASTERISK-30065 - pjsip: Open Websocket connection is not
  reused for outgoing requests
  (Reported by LA)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1

Thank you for your continued support of Asterisk!
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[asterisk-dev] Asterisk 19.4.0 Now Available

2022-05-12 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
19.4.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 19.4.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Security bugs fixed in this release:
---
 * ASTERISK-29476 - res_stir_shaken: Blind SSRF vulnerabilities

  (Reported by Clint Ruoho)
 * ASTERISK-29838 - ${SQL_ESC()} not correctly escaping a
  terminating \
  (Reported by Leandro Dardini)
 * ASTERISK-29872 - res_stir_shaken: Resource exhaustion with
  large files
  (Reported by Benjamin Keith Ford)

New Features made in this release:
---
 * ASTERISK-29931 - Option to allow a user to not hear the join
  sound on enter but everyone else can
  (Reported by Michael
  Cargile)
 * ASTERISK-29968 - func_db: Add a function to return
  cardinality of keys at prefix
  (Reported by N A)
 * ASTERISK-29486 - Hint-like extension value lookup function
  without device state
  (Reported by N A)
 * ASTERISK-29820 - cli: Add command to evaluate a function

  (Reported by N A)
 * ASTERISK-29941 - chan_pjsip: Add ability to send flash
  events
  (Reported by N A)
 * ASTERISK-29876 - app_queue: Add music on hold option
 
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-29655 - res_pjsip_session: No video to caller if no
  camera available
  (Reported by Michael Auracher)
 * ASTERISK-29638 - res_pjsip_session: No video after early
  media
  (Reported by Michael Auracher)
 * ASTERISK-28518 - chan_dahdi: Caller ID FSK Erroneously Sent
  when Picking Up Dahdi Call On Hold
  (Reported by Josh
  Alberts)
 * ASTERISK-30007 - chan_iax2: Prevent crashes due to attempted
  encryption with missing secrets
  (Reported by N A)
 * ASTERISK-29990 - chan_dahdi: adding ring cadences is not
  idempotent on dahdi restart
  (Reported by N A)
 * ASTERISK-29728 - menuselect: Disabled by default modules that
  are enabled are always recompiled
  (Reported by N A)
 * ASTERISK-30002 - app_meetme: Don't erroneously set global
  variables when channel is NULL
  (Reported by N A)
 * ASTERISK-22246 - Asterisk's "T" flag is ignored when used
  with "r" or "R" flags. (documentation bug)
  (Reported by
  Rusty Newton)
 * ASTERISK-26582 - Asterisk seems to ignore the "n" parameter
  for "disable console colorization"
  (Reported by Sebastian
  Gutierrez)
 * ASTERISK-29994 - chan_dahdi: Round robin array size is too
  small for max number of groups
  (Reported by N A)
 * ASTERISK-29943 - file.c: seeking to negative file offset is
  not prevented
  (Reported by N A)
 * ASTERISK-29843 - Session timers get removed on UPDATE
 
  (Reported by Mark Petersen)
 * ASTERISK-29842 - Do not change 180 Ringing to 183 Progress
  even if early_media already enabled
  (Reported by Mark
  Petersen)
 * ASTERISK-29955 - chan_sip: SIP route header is missing on
  UPDATE
  (Reported by Mark Petersen)
 * ASTERISK-29253 - Incorrect bridging on transfer
 
  (Reported by Yury Kirsanov)
 * ASTERISK-29948 - iostream: Infinite TCP timeout writing data

  (Reported by N A)
 * ASTERISK-30024 - Failed to sign STIR/SHAKEN payload with
  functionality not enabled
  (Reported by Claude Diderich)
 * ASTERISK-30006 - res_pjsip: UDP transport does not work when
  async_operations is greater than 1
  (Reported by Ross Beer)
 * ASTERISK-30021 - ast_variable_list_replace_variable uses
  variable with new keyword
  (Reported by Jasper
  Hafkenscheid)
 * ASTERISK-30015 - pjsip / WebRTC: Chrome creating large number
  of SDP attributes
  (Reported by Josh Hogan)
 * ASTERISK-30023 - cdr_adaptive_odbc: does not support DATETIME
  database columns
  (Reported by Gregory Massel)
 * ASTERISK-26689 - res_pjsip_sdp_rtp: 183 Session in Progress.
  Disconnecting channel for lack of RTP activity
  (Reported
  by Dmitriy Serov)
 * ASTERISK-29929 - res_pjsip_sdp_rtp: Disconnecting channel for
  lack of RTP activity in one way sessions
  (Reported by
  Boris P. Korzun)
 * ASTERISK-29411 - Crash in pjsip_msg_find_hdr_by_name
 
  (Reported by LA)
 * ASTERISK-29535 - Segmentation fault in libasteriskpj.so.2
   
  (Reported by Daniel Bonazzi)
 * ASTERISK-26719 - pbx: Only up to 127 includes in a dialplan
  context (AST_PBX_MAX_STACK - 1)
  (Reported by Tzafrir
  Cohen)
 * ASTERISK-29986 - build: Asterisk 18.11.0 doesn't compile when
  wget isn't available
  (Reported by Stefan Ruijs

[asterisk-dev] Asterisk 18.12.0 Now Available

2022-05-12 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
18.12.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.12.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Security bugs fixed in this release:
---
 * ASTERISK-29476 - res_stir_shaken: Blind SSRF vulnerabilities

  (Reported by Clint Ruoho)
 * ASTERISK-29838 - ${SQL_ESC()} not correctly escaping a
  terminating \
  (Reported by Leandro Dardini)
 * ASTERISK-29872 - res_stir_shaken: Resource exhaustion with
  large files
  (Reported by Benjamin Keith Ford)

New Features made in this release:
---
 * ASTERISK-29931 - Option to allow a user to not hear the join
  sound on enter but everyone else can
  (Reported by Michael
  Cargile)
 * ASTERISK-29968 - func_db: Add a function to return
  cardinality of keys at prefix
  (Reported by N A)
 * ASTERISK-29486 - Hint-like extension value lookup function
  without device state
  (Reported by N A)
 * ASTERISK-29941 - chan_pjsip: Add ability to send flash
  events
  (Reported by N A)
 * ASTERISK-29820 - cli: Add command to evaluate a function

  (Reported by N A)
 * ASTERISK-29876 - app_queue: Add music on hold option
 
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-29655 - res_pjsip_session: No video to caller if no
  camera available
  (Reported by Michael Auracher)
 * ASTERISK-29638 - res_pjsip_session: No video after early
  media
  (Reported by Michael Auracher)
 * ASTERISK-28518 - chan_dahdi: Caller ID FSK Erroneously Sent
  when Picking Up Dahdi Call On Hold
  (Reported by Josh
  Alberts)
 * ASTERISK-29990 - chan_dahdi: adding ring cadences is not
  idempotent on dahdi restart
  (Reported by N A)
 * ASTERISK-30007 - chan_iax2: Prevent crashes due to attempted
  encryption with missing secrets
  (Reported by N A)
 * ASTERISK-29728 - menuselect: Disabled by default modules that
  are enabled are always recompiled
  (Reported by N A)
 * ASTERISK-30002 - app_meetme: Don't erroneously set global
  variables when channel is NULL
  (Reported by N A)
 * ASTERISK-29994 - chan_dahdi: Round robin array size is too
  small for max number of groups
  (Reported by N A)
 * ASTERISK-22246 - Asterisk's "T" flag is ignored when used
  with "r" or "R" flags. (documentation bug)
  (Reported by
  Rusty Newton)
 * ASTERISK-26582 - Asterisk seems to ignore the "n" parameter
  for "disable console colorization"
  (Reported by Sebastian
  Gutierrez)
 * ASTERISK-29843 - Session timers get removed on UPDATE
 
  (Reported by Mark Petersen)
 * ASTERISK-29943 - file.c: seeking to negative file offset is
  not prevented
  (Reported by N A)
 * ASTERISK-29955 - chan_sip: SIP route header is missing on
  UPDATE
  (Reported by Mark Petersen)
 * ASTERISK-29842 - Do not change 180 Ringing to 183 Progress
  even if early_media already enabled
  (Reported by Mark
  Petersen)
 * ASTERISK-29948 - iostream: Infinite TCP timeout writing data

  (Reported by N A)
 * ASTERISK-29253 - Incorrect bridging on transfer
 
  (Reported by Yury Kirsanov)
 * ASTERISK-30006 - res_pjsip: UDP transport does not work when
  async_operations is greater than 1
  (Reported by Ross Beer)
 * ASTERISK-30024 - Failed to sign STIR/SHAKEN payload with
  functionality not enabled
  (Reported by Claude Diderich)
 * ASTERISK-30021 - ast_variable_list_replace_variable uses
  variable with new keyword
  (Reported by Jasper
  Hafkenscheid)
 * ASTERISK-30023 - cdr_adaptive_odbc: does not support DATETIME
  database columns
  (Reported by Gregory Massel)
 * ASTERISK-30015 - pjsip / WebRTC: Chrome creating large number
  of SDP attributes
  (Reported by Josh Hogan)
 * ASTERISK-26689 - res_pjsip_sdp_rtp: 183 Session in Progress.
  Disconnecting channel for lack of RTP activity
  (Reported
  by Dmitriy Serov)
 * ASTERISK-29929 - res_pjsip_sdp_rtp: Disconnecting channel for
  lack of RTP activity in one way sessions
  (Reported by
  Boris P. Korzun)
 * ASTERISK-29411 - Crash in pjsip_msg_find_hdr_by_name
 
  (Reported by LA)
 * ASTERISK-29535 - Segmentation fault in libasteriskpj.so.2
   
  (Reported by Daniel Bonazzi)
 * ASTERISK-26719 - pbx: Only up to 127 includes in a dialplan
  context (AST_PBX_MAX_STACK - 1)
  (Reported by Tzafrir
  Cohen)
 * ASTERISK-29986 - build: Asterisk 18.11.0 doesn't compile when
  wget isn't available
  (Reported by Stefan Ruijs

[asterisk-dev] Asterisk 16.26.0 Now Available

2022-05-12 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
16.26.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.26.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Security bugs fixed in this release:
---
 * ASTERISK-29476 - res_stir_shaken: Blind SSRF vulnerabilities

  (Reported by Clint Ruoho)
 * ASTERISK-29838 - ${SQL_ESC()} not correctly escaping a
  terminating \
  (Reported by Leandro Dardini)
 * ASTERISK-29872 - res_stir_shaken: Resource exhaustion with
  large files
  (Reported by Benjamin Keith Ford)

New Features made in this release:
---
 * ASTERISK-29931 - Option to allow a user to not hear the join
  sound on enter but everyone else can
  (Reported by Michael
  Cargile)
 * ASTERISK-29968 - func_db: Add a function to return
  cardinality of keys at prefix
  (Reported by N A)
 * ASTERISK-29486 - Hint-like extension value lookup function
  without device state
  (Reported by N A)
 * ASTERISK-29941 - chan_pjsip: Add ability to send flash
  events
  (Reported by N A)
 * ASTERISK-29820 - cli: Add command to evaluate a function

  (Reported by N A)
 * ASTERISK-29876 - app_queue: Add music on hold option
 
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-28518 - chan_dahdi: Caller ID FSK Erroneously Sent
  when Picking Up Dahdi Call On Hold
  (Reported by Josh
  Alberts)
 * ASTERISK-29990 - chan_dahdi: adding ring cadences is not
  idempotent on dahdi restart
  (Reported by N A)
 * ASTERISK-30007 - chan_iax2: Prevent crashes due to attempted
  encryption with missing secrets
  (Reported by N A)
 * ASTERISK-29728 - menuselect: Disabled by default modules that
  are enabled are always recompiled
  (Reported by N A)
 * ASTERISK-30002 - app_meetme: Don't erroneously set global
  variables when channel is NULL
  (Reported by N A)
 * ASTERISK-29994 - chan_dahdi: Round robin array size is too
  small for max number of groups
  (Reported by N A)
 * ASTERISK-22246 - Asterisk's "T" flag is ignored when used
  with "r" or "R" flags. (documentation bug)
  (Reported by
  Rusty Newton)
 * ASTERISK-26582 - Asterisk seems to ignore the "n" parameter
  for "disable console colorization"
  (Reported by Sebastian
  Gutierrez)
 * ASTERISK-29843 - Session timers get removed on UPDATE
 
  (Reported by Mark Petersen)
 * ASTERISK-29943 - file.c: seeking to negative file offset is
  not prevented
  (Reported by N A)
 * ASTERISK-29955 - chan_sip: SIP route header is missing on
  UPDATE
  (Reported by Mark Petersen)
 * ASTERISK-29842 - Do not change 180 Ringing to 183 Progress
  even if early_media already enabled
  (Reported by Mark
  Petersen)
 * ASTERISK-29948 - iostream: Infinite TCP timeout writing data

  (Reported by N A)
 * ASTERISK-29253 - Incorrect bridging on transfer
 
  (Reported by Yury Kirsanov)
 * ASTERISK-30024 - Failed to sign STIR/SHAKEN payload with
  functionality not enabled
  (Reported by Claude Diderich)
 * ASTERISK-30006 - res_pjsip: UDP transport does not work when
  async_operations is greater than 1
  (Reported by Ross Beer)
 * ASTERISK-29655 - res_pjsip_session: No video to caller if no
  camera available
  (Reported by Michael Auracher)
 * ASTERISK-29638 - res_pjsip_session: No video after early
  media
  (Reported by Michael Auracher)
 * ASTERISK-30015 - pjsip / WebRTC: Chrome creating large number
  of SDP attributes
  (Reported by Josh Hogan)
 * ASTERISK-30021 - ast_variable_list_replace_variable uses
  variable with new keyword
  (Reported by Jasper
  Hafkenscheid)
 * ASTERISK-30023 - cdr_adaptive_odbc: does not support DATETIME
  database columns
  (Reported by Gregory Massel)
 * ASTERISK-29411 - Crash in pjsip_msg_find_hdr_by_name
 
  (Reported by LA)
 * ASTERISK-29535 - Segmentation fault in libasteriskpj.so.2
   
  (Reported by Daniel Bonazzi)
 * ASTERISK-26719 - pbx: Only up to 127 includes in a dialplan
  context (AST_PBX_MAX_STACK - 1)
  (Reported by Tzafrir
  Cohen)
 * ASTERISK-29988 - REGRESSION: The build process is requiring
  xmllint or xmlstarlet ro be installed when it shouldn't
 
  (Reported by George Joseph)
 * ASTERISK-29986 - build: Asterisk 18.11.0 doesn't compile when
  wget isn't available
  (Reported by Stefan Ruijsenaars)
 * ASTERISK-29895 - chan_iax2: Fix misaligned spacing in iax2
  show netstats printout
  (Reported by N A)
 * ASTERISK-29939 - a

[asterisk-dev] Asterisk 19.4.0-rc1 Now Available

2022-05-05 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the first
release candidate of Asterisk 19.4.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 19.4.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release candidate:

Security bugs fixed in this release:
---
 * ASTERISK-29476 - res_stir_shaken: Blind SSRF vulnerabilities

  (Reported by Clint Ruoho)
 * ASTERISK-29838 - ${SQL_ESC()} not correctly escaping a
  terminating \
  (Reported by Leandro Dardini)
 * ASTERISK-29872 - res_stir_shaken: Resource exhaustion with
  large files
  (Reported by Benjamin Keith Ford)

New Features made in this release:
---
 * ASTERISK-29931 - Option to allow a user to not hear the join
  sound on enter but everyone else can
  (Reported by Michael
  Cargile)
 * ASTERISK-29968 - func_db: Add a function to return
  cardinality of keys at prefix
  (Reported by N A)
 * ASTERISK-29486 - Hint-like extension value lookup function
  without device state
  (Reported by N A)
 * ASTERISK-29820 - cli: Add command to evaluate a function

  (Reported by N A)
 * ASTERISK-29941 - chan_pjsip: Add ability to send flash
  events
  (Reported by N A)
 * ASTERISK-29876 - app_queue: Add music on hold option
 
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-29655 - res_pjsip_session: No video to caller if no
  camera available
  (Reported by Michael Auracher)
 * ASTERISK-29638 - res_pjsip_session: No video after early
  media
  (Reported by Michael Auracher)
 * ASTERISK-28518 - chan_dahdi: Caller ID FSK Erroneously Sent
  when Picking Up Dahdi Call On Hold
  (Reported by Josh
  Alberts)
 * ASTERISK-30007 - chan_iax2: Prevent crashes due to attempted
  encryption with missing secrets
  (Reported by N A)
 * ASTERISK-29990 - chan_dahdi: adding ring cadences is not
  idempotent on dahdi restart
  (Reported by N A)
 * ASTERISK-29728 - menuselect: Disabled by default modules that
  are enabled are always recompiled
  (Reported by N A)
 * ASTERISK-30002 - app_meetme: Don't erroneously set global
  variables when channel is NULL
  (Reported by N A)
 * ASTERISK-22246 - Asterisk's "T" flag is ignored when used
  with "r" or "R" flags. (documentation bug)
  (Reported by
  Rusty Newton)
 * ASTERISK-26582 - Asterisk seems to ignore the "n" parameter
  for "disable console colorization"
  (Reported by Sebastian
  Gutierrez)
 * ASTERISK-29994 - chan_dahdi: Round robin array size is too
  small for max number of groups
  (Reported by N A)
 * ASTERISK-29943 - file.c: seeking to negative file offset is
  not prevented
  (Reported by N A)
 * ASTERISK-29843 - Session timers get removed on UPDATE
 
  (Reported by Mark Petersen)
 * ASTERISK-29842 - Do not change 180 Ringing to 183 Progress
  even if early_media already enabled
  (Reported by Mark
  Petersen)
 * ASTERISK-29955 - chan_sip: SIP route header is missing on
  UPDATE
  (Reported by Mark Petersen)
 * ASTERISK-29253 - Incorrect bridging on transfer
 
  (Reported by Yury Kirsanov)
 * ASTERISK-29948 - iostream: Infinite TCP timeout writing data

  (Reported by N A)
 * ASTERISK-30024 - Failed to sign STIR/SHAKEN payload with
  functionality not enabled
  (Reported by Claude Diderich)
 * ASTERISK-30006 - res_pjsip: UDP transport does not work when
  async_operations is greater than 1
  (Reported by Ross Beer)
 * ASTERISK-30021 - ast_variable_list_replace_variable uses
  variable with new keyword
  (Reported by Jasper
  Hafkenscheid)
 * ASTERISK-30015 - pjsip / WebRTC: Chrome creating large number
  of SDP attributes
  (Reported by Josh Hogan)
 * ASTERISK-30023 - cdr_adaptive_odbc: does not support DATETIME
  database columns
  (Reported by Gregory Massel)
 * ASTERISK-26689 - res_pjsip_sdp_rtp: 183 Session in Progress.
  Disconnecting channel for lack of RTP activity
  (Reported
  by Dmitriy Serov)
 * ASTERISK-29929 - res_pjsip_sdp_rtp: Disconnecting channel for
  lack of RTP activity in one way sessions
  (Reported by
  Boris P. Korzun)
 * ASTERISK-29411 - Crash in pjsip_msg_find_hdr_by_name
 
  (Reported by LA)
 * ASTERISK-29535 - Segmentation fault in libasteriskpj.so.2
   
  (Reported by Daniel Bonazzi)
 * ASTERISK-26719 - pbx: Only up to 127 includes in a dialplan
  context (AST_PBX_MAX_STACK - 1)
  (Reported by Tzafrir
  Cohen)
 * ASTERISK-29986 - build: Asterisk 18.11.0 doesn't compile when
  wget isn't available

[asterisk-dev] Asterisk 18.12.0-rc1 Now Available

2022-05-05 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the first
release candidate of Asterisk 18.12.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.12.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release candidate:

Security bugs fixed in this release:
---
 * ASTERISK-29476 - res_stir_shaken: Blind SSRF vulnerabilities

  (Reported by Clint Ruoho)
 * ASTERISK-29838 - ${SQL_ESC()} not correctly escaping a
  terminating \
  (Reported by Leandro Dardini)
 * ASTERISK-29872 - res_stir_shaken: Resource exhaustion with
  large files
  (Reported by Benjamin Keith Ford)

New Features made in this release:
---
 * ASTERISK-29931 - Option to allow a user to not hear the join
  sound on enter but everyone else can
  (Reported by Michael
  Cargile)
 * ASTERISK-29968 - func_db: Add a function to return
  cardinality of keys at prefix
  (Reported by N A)
 * ASTERISK-29486 - Hint-like extension value lookup function
  without device state
  (Reported by N A)
 * ASTERISK-29941 - chan_pjsip: Add ability to send flash
  events
  (Reported by N A)
 * ASTERISK-29820 - cli: Add command to evaluate a function

  (Reported by N A)
 * ASTERISK-29876 - app_queue: Add music on hold option
 
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-29655 - res_pjsip_session: No video to caller if no
  camera available
  (Reported by Michael Auracher)
 * ASTERISK-29638 - res_pjsip_session: No video after early
  media
  (Reported by Michael Auracher)
 * ASTERISK-28518 - chan_dahdi: Caller ID FSK Erroneously Sent
  when Picking Up Dahdi Call On Hold
  (Reported by Josh
  Alberts)
 * ASTERISK-29990 - chan_dahdi: adding ring cadences is not
  idempotent on dahdi restart
  (Reported by N A)
 * ASTERISK-30007 - chan_iax2: Prevent crashes due to attempted
  encryption with missing secrets
  (Reported by N A)
 * ASTERISK-29728 - menuselect: Disabled by default modules that
  are enabled are always recompiled
  (Reported by N A)
 * ASTERISK-30002 - app_meetme: Don't erroneously set global
  variables when channel is NULL
  (Reported by N A)
 * ASTERISK-29994 - chan_dahdi: Round robin array size is too
  small for max number of groups
  (Reported by N A)
 * ASTERISK-22246 - Asterisk's "T" flag is ignored when used
  with "r" or "R" flags. (documentation bug)
  (Reported by
  Rusty Newton)
 * ASTERISK-26582 - Asterisk seems to ignore the "n" parameter
  for "disable console colorization"
  (Reported by Sebastian
  Gutierrez)
 * ASTERISK-29843 - Session timers get removed on UPDATE
 
  (Reported by Mark Petersen)
 * ASTERISK-29943 - file.c: seeking to negative file offset is
  not prevented
  (Reported by N A)
 * ASTERISK-29955 - chan_sip: SIP route header is missing on
  UPDATE
  (Reported by Mark Petersen)
 * ASTERISK-29842 - Do not change 180 Ringing to 183 Progress
  even if early_media already enabled
  (Reported by Mark
  Petersen)
 * ASTERISK-29948 - iostream: Infinite TCP timeout writing data

  (Reported by N A)
 * ASTERISK-29253 - Incorrect bridging on transfer
 
  (Reported by Yury Kirsanov)
 * ASTERISK-30006 - res_pjsip: UDP transport does not work when
  async_operations is greater than 1
  (Reported by Ross Beer)
 * ASTERISK-30024 - Failed to sign STIR/SHAKEN payload with
  functionality not enabled
  (Reported by Claude Diderich)
 * ASTERISK-30021 - ast_variable_list_replace_variable uses
  variable with new keyword
  (Reported by Jasper
  Hafkenscheid)
 * ASTERISK-30023 - cdr_adaptive_odbc: does not support DATETIME
  database columns
  (Reported by Gregory Massel)
 * ASTERISK-30015 - pjsip / WebRTC: Chrome creating large number
  of SDP attributes
  (Reported by Josh Hogan)
 * ASTERISK-26689 - res_pjsip_sdp_rtp: 183 Session in Progress.
  Disconnecting channel for lack of RTP activity
  (Reported
  by Dmitriy Serov)
 * ASTERISK-29929 - res_pjsip_sdp_rtp: Disconnecting channel for
  lack of RTP activity in one way sessions
  (Reported by
  Boris P. Korzun)
 * ASTERISK-29411 - Crash in pjsip_msg_find_hdr_by_name
 
  (Reported by LA)
 * ASTERISK-29535 - Segmentation fault in libasteriskpj.so.2
   
  (Reported by Daniel Bonazzi)
 * ASTERISK-26719 - pbx: Only up to 127 includes in a dialplan
  context (AST_PBX_MAX_STACK - 1)
  (Reported by Tzafrir
  Cohen)
 * ASTERISK-29986 - build: Asterisk 18.11.0 doesn't compile when
  wget isn

[asterisk-dev] Asterisk 16.26.0-rc1 Now Available

2022-05-05 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the first
release candidate of Asterisk 16.26.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.26.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release candidate:

Security bugs fixed in this release:
---
 * ASTERISK-29476 - res_stir_shaken: Blind SSRF vulnerabilities

  (Reported by Clint Ruoho)
 * ASTERISK-29838 - ${SQL_ESC()} not correctly escaping a
  terminating \
  (Reported by Leandro Dardini)
 * ASTERISK-29872 - res_stir_shaken: Resource exhaustion with
  large files
  (Reported by Benjamin Keith Ford)

New Features made in this release:
---
 * ASTERISK-29931 - Option to allow a user to not hear the join
  sound on enter but everyone else can
  (Reported by Michael
  Cargile)
 * ASTERISK-29968 - func_db: Add a function to return
  cardinality of keys at prefix
  (Reported by N A)
 * ASTERISK-29486 - Hint-like extension value lookup function
  without device state
  (Reported by N A)
 * ASTERISK-29941 - chan_pjsip: Add ability to send flash
  events
  (Reported by N A)
 * ASTERISK-29820 - cli: Add command to evaluate a function

  (Reported by N A)
 * ASTERISK-29876 - app_queue: Add music on hold option
 
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-28518 - chan_dahdi: Caller ID FSK Erroneously Sent
  when Picking Up Dahdi Call On Hold
  (Reported by Josh
  Alberts)
 * ASTERISK-29990 - chan_dahdi: adding ring cadences is not
  idempotent on dahdi restart
  (Reported by N A)
 * ASTERISK-30007 - chan_iax2: Prevent crashes due to attempted
  encryption with missing secrets
  (Reported by N A)
 * ASTERISK-29728 - menuselect: Disabled by default modules that
  are enabled are always recompiled
  (Reported by N A)
 * ASTERISK-30002 - app_meetme: Don't erroneously set global
  variables when channel is NULL
  (Reported by N A)
 * ASTERISK-29994 - chan_dahdi: Round robin array size is too
  small for max number of groups
  (Reported by N A)
 * ASTERISK-22246 - Asterisk's "T" flag is ignored when used
  with "r" or "R" flags. (documentation bug)
  (Reported by
  Rusty Newton)
 * ASTERISK-26582 - Asterisk seems to ignore the "n" parameter
  for "disable console colorization"
  (Reported by Sebastian
  Gutierrez)
 * ASTERISK-29843 - Session timers get removed on UPDATE
 
  (Reported by Mark Petersen)
 * ASTERISK-29943 - file.c: seeking to negative file offset is
  not prevented
  (Reported by N A)
 * ASTERISK-29955 - chan_sip: SIP route header is missing on
  UPDATE
  (Reported by Mark Petersen)
 * ASTERISK-29842 - Do not change 180 Ringing to 183 Progress
  even if early_media already enabled
  (Reported by Mark
  Petersen)
 * ASTERISK-29948 - iostream: Infinite TCP timeout writing data

  (Reported by N A)
 * ASTERISK-29253 - Incorrect bridging on transfer
 
  (Reported by Yury Kirsanov)
 * ASTERISK-30024 - Failed to sign STIR/SHAKEN payload with
  functionality not enabled
  (Reported by Claude Diderich)
 * ASTERISK-30006 - res_pjsip: UDP transport does not work when
  async_operations is greater than 1
  (Reported by Ross Beer)
 * ASTERISK-29655 - res_pjsip_session: No video to caller if no
  camera available
  (Reported by Michael Auracher)
 * ASTERISK-29638 - res_pjsip_session: No video after early
  media
  (Reported by Michael Auracher)
 * ASTERISK-30015 - pjsip / WebRTC: Chrome creating large number
  of SDP attributes
  (Reported by Josh Hogan)
 * ASTERISK-30021 - ast_variable_list_replace_variable uses
  variable with new keyword
  (Reported by Jasper
  Hafkenscheid)
 * ASTERISK-30023 - cdr_adaptive_odbc: does not support DATETIME
  database columns
  (Reported by Gregory Massel)
 * ASTERISK-29411 - Crash in pjsip_msg_find_hdr_by_name
 
  (Reported by LA)
 * ASTERISK-29535 - Segmentation fault in libasteriskpj.so.2
   
  (Reported by Daniel Bonazzi)
 * ASTERISK-26719 - pbx: Only up to 127 includes in a dialplan
  context (AST_PBX_MAX_STACK - 1)
  (Reported by Tzafrir
  Cohen)
 * ASTERISK-29988 - REGRESSION: The build process is requiring
  xmllint or xmlstarlet ro be installed when it shouldn't
 
  (Reported by George Joseph)
 * ASTERISK-29986 - build: Asterisk 18.11.0 doesn't compile when
  wget isn't available
  (Reported by Stefan Ruijsenaars)
 * ASTERISK-29895 - chan_iax2: Fix misaligned spacing in iax2
  show netstats printout

[asterisk-dev] Asterisk 19.3.3 Now Available

2022-04-26 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
19.3.3.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 19.3.3 resolves an issue reported by the
community and would have not been possible without your participation.

Thank you!

The following issue is resolved in this release:

Bugs fixed in this release:
---
 * ASTERISK-30024 - Failed to sign STIR/SHAKEN payload with
  functionality not enabled
  (Reported by Claude Diderich)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3

Thank you for your continued support of Asterisk!
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[asterisk-dev] Asterisk 18.11.3 Now Available

2022-04-26 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
18.11.3.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.11.3 resolves an issue reported by the
community and would have not been possible without your participation.

Thank you!

The following issue is resolved in this release:

Bugs fixed in this release:
---
 * ASTERISK-30024 - Failed to sign STIR/SHAKEN payload with
  functionality not enabled
  (Reported by Claude Diderich)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3

Thank you for your continued support of Asterisk!
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[asterisk-dev] Asterisk 16.25.3 Now Available

2022-04-26 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
16.25.3.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.25.3 resolves an issue reported by the
community and would have not been possible without your participation.

Thank you!

The following issue is resolved in this release:

Bugs fixed in this release:
---
 * ASTERISK-30024 - Failed to sign STIR/SHAKEN payload with
  functionality not enabled
  (Reported by Claude Diderich)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3

Thank you for your continued support of Asterisk!
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[asterisk-dev] Asterisk 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14 Now Available (Security)

2022-04-14 Thread Asterisk Development Team

The Asterisk Development Team would like to announce security releases for
Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases are
released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.

These releases are available for immediate download at

https://downloads.asterisk.org/pub/telephony/asterisk/releases
https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases

The following security vulnerabilities were resolved in these versions:

* AST-2022-001: res_stir_shaken: resource exhaustion with large files
  When using STIR/SHAKEN, itâs possible to download files that are not
  certificates. These files could be much larger than what you would expect to
  download.

* AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header
  When using STIR/SHAKEN, itâs possible to send arbitrary requests like GET to
  interfaces such as localhost using the Identity header.

* AST-2022-003: func_odbc: Possible SQL Injection
  Some databases can use backslashes to escape certain characters, such as
  backticks. If input is provided to func_odbc which includes backslashes it is
  possible for func_odbc to construct a broken SQL query and the SQL query to
  fail.

For a full list of changes in the current releases, please see the ChangeLogs:

https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14

The security advisories are available at:

https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
https://downloads.asterisk.org/pub/security/AST-2022-003.pdf

Thank you for your continued support of Asterisk!-- 
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[asterisk-dev] Asterisk 19.3.1 Now Available

2022-03-30 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
19.3.1.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 19.3.1 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Bugs fixed in this release:
---
 * ASTERISK-29986 - build: Asterisk 18.11.0 doesn't compile when
  wget isn't available
  (Reported by Stefan Ruijsenaars)
 * ASTERISK-29988 - REGRESSION: The build process is requiring
  xmllint or xmlstarlet ro be installed when it shouldn't
 
  (Reported by George Joseph)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1

Thank you for your continued support of Asterisk!
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[asterisk-dev] Asterisk 18.11.1 Now Available

2022-03-29 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
18.11.1.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.11.1 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Bugs fixed in this release:
---
 * ASTERISK-29986 - build: Asterisk 18.11.0 doesn't compile when
  wget isn't available
  (Reported by Stefan Ruijsenaars)
 * ASTERISK-29988 - REGRESSION: The build process is requiring
  xmllint or xmlstarlet ro be installed when it shouldn't
 
  (Reported by George Joseph)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1

Thank you for your continued support of Asterisk!
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