from:"Asterisk Development Team"

[asterisk-users] 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 libjwt to third-part

[asterisk-users] 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/pickupgroup.
-

[asterisk-users] 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: Fix crashes for s

[asterisk-users] 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()

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] 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()

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] 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()

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] 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()

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] 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
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] 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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Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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[asterisk-users] 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:

None

--
_
-- B

[asterisk-users] 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 defunct functionality.
The previously deprecated

[asterisk-users] 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

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] 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

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] 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-users] 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 applying caller pr

[asterisk-users] 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 applying caller pr

[asterisk-users] 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

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

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  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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[asterisk-users] 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
accepts a new value 'sdp_label' which will configure

[asterisk-users] 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' argument now
accepts a new value 'sdp_label

[asterisk-users] 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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Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
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To UNSUBSCRIBE or update options visit:
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[asterisk-users] 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 'skip call' option.
  A new option 's' has been added to the Directory

[asterisk-users] 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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Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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[asterisk-users] 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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Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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[asterisk-users] 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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-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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[asterisk-users] 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 MixMonitorID when calling
  the manager action:

[asterisk-users] 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 possible to specify the MixMonitorID when ca

[asterisk-users] 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-users] 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-users] 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-users] 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-users] 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 continued support

[asterisk-users] 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!
-- 
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[asterisk-users] 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-users] 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-users] 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-users] 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!
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Sta

[asterisk-users] 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!
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to

[asterisk-users] 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!
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to

[asterisk-users] 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!
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] 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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Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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[asterisk-users] 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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Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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[asterisk-users] 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!
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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[asterisk-users] 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!
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
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[asterisk-users] 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!
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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[asterisk-users] 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!
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] 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!
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] 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!
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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[asterisk-users] 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 Ruijsenaars)
 * ASTERISK-

[asterisk-users] 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 Ruijsenaars)
 * ASTERISK-

[asterisk-users] 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 - agi: Fix xmldoc bug with s

[asterisk-users] 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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Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] 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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-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] 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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-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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[asterisk-users] 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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-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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[asterisk-users] 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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-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] 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!
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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[asterisk-users] Asterisk 16.25.1 Now Available

2022-03-29 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
16.25.1.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.25.1 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-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)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1

Thank you for your continued support of Asterisk!
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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[asterisk-users] Asterisk 19.3.0 Now Available

2022-03-24 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
19.3.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 19.3.0 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-29945 - pjproject: Security fixes for things
 
  (Reported by Kevin Harwell)

New Features made in this release:
---
 * ASTERISK-29853 - ami: Allow events to be globally disabled
  
  (Reported by N A)
 * ASTERISK-29840 - func_channel: Add LASTCONTEXT and LASTEXTEN
  fields
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-29924 - res_config_pgsql: omit "unsupported column
  type 'text'" error
  (Reported by Boris P. Korzun)
 * ASTERISK-29923 - docs, LICENSE: pbx.digium.com no longer
  exists
  (Reported by N A)
 * ASTERISK-29904 - RLS: Batched Notifications stop working

  (Reported by Alexei Gradinari)
 * ASTERISK-29365 - taskprocessor: Can cause assert at shutdown

  (Reported by Joshua C. Colp)
 * ASTERISK-29873 - [patch] Queue Realtime load
  (Reported
  by Alexei Gradinari)
 * ASTERISK-18416 - [patch] Realtime queue agents unavailable
  via AMI before a call event.
  (Reported by kwk)
 * ASTERISK-27597 - AMI Queuestatus not working (with realtime
  queue)
  (Reported by cagdas kopuz)
 * ASTERISK-29871 - res_prometheus: Failure to load causes
  FRACKs
  (Reported by Mark Petersen)
 * ASTERISK-29886 - Asterisk AMI sends not-valid XML
 
  (Reported by Napadailo Yaroslav)

Improvements made in this release:
---
 * ASTERISK-29906 - [patch] update RLS to reflect the changes to
  the lists
  (Reported by Alexei Gradinari)
 * ASTERISK-29909 - app_queue: Add support for withdrawing a
  call
  (Reported by Kfir Itzhak)
 * ASTERISK-29353 - Qualify jansson 2.14 for asterisk
 
  (Reported by George Joseph)
 * ASTERISK-29897 - channels: Increase core debug levels for
  chatty debugs
  (Reported by N A)
 * ASTERISK-29861 - asterisk.h: add macro for curl user agent
  
  (Reported by N A)
 * ASTERISK-29896 - xmldocs: Add since tag
  (Reported by N
  A)
 * ASTERISK-29809 - curl, stir_shaken: refactor curl code
 
  (Reported by N A)
 * ASTERISK-29920 - app_voicemail: Warn if trying to manage
  nonexistent mailbox
  (Reported by N A)
 * ASTERISK-29925 - func_db: Warn about malformed key names

  (Reported by N A)
 * ASTERISK-29891 - [patch] provide a display name for RLS
  subscriptions
  (Reported by Alexei Gradinari)
 * ASTERISK-29898 - documentation: Add default attributes to
  documentation
  (Reported by N A)
 * ASTERISK-29866 - cli: add core dump information to core show
  settings
  (Reported by N A)
 * ASTERISK-29900 - app_mp3: Document and warn about https
  incompatibility
  (Reported by N A)
 * ASTERISK-29877 - app_mf: Allow reading a maximum number of
  digits
  (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-19.3.0

Thank you for your continued support of Asterisk!
-- 
_
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Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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To UNSUBSCRIBE or update options visit:
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[asterisk-users] Asterisk 18.11.0 Now Available

2022-03-24 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
18.11.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.11.0 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-29945 - pjproject: Security fixes for things
 
  (Reported by Kevin Harwell)

New Features made in this release:
---
 * ASTERISK-29853 - ami: Allow events to be globally disabled
  
  (Reported by N A)
 * ASTERISK-29840 - func_channel: Add LASTCONTEXT and LASTEXTEN
  fields
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-29924 - res_config_pgsql: omit "unsupported column
  type 'text'" error
  (Reported by Boris P. Korzun)
 * ASTERISK-29923 - docs, LICENSE: pbx.digium.com no longer
  exists
  (Reported by N A)
 * ASTERISK-29904 - RLS: Batched Notifications stop working

  (Reported by Alexei Gradinari)
 * ASTERISK-29365 - taskprocessor: Can cause assert at shutdown

  (Reported by Joshua C. Colp)
 * ASTERISK-29873 - [patch] Queue Realtime load
  (Reported
  by Alexei Gradinari)
 * ASTERISK-18416 - [patch] Realtime queue agents unavailable
  via AMI before a call event.
  (Reported by kwk)
 * ASTERISK-27597 - AMI Queuestatus not working (with realtime
  queue)
  (Reported by cagdas kopuz)
 * ASTERISK-29871 - res_prometheus: Failure to load causes
  FRACKs
  (Reported by Mark Petersen)
 * ASTERISK-29886 - Asterisk AMI sends not-valid XML
 
  (Reported by Napadailo Yaroslav)

Improvements made in this release:
---
 * ASTERISK-29909 - app_queue: Add support for withdrawing a
  call
  (Reported by Kfir Itzhak)
 * ASTERISK-29906 - [patch] update RLS to reflect the changes to
  the lists
  (Reported by Alexei Gradinari)
 * ASTERISK-29353 - Qualify jansson 2.14 for asterisk
 
  (Reported by George Joseph)
 * ASTERISK-29897 - channels: Increase core debug levels for
  chatty debugs
  (Reported by N A)
 * ASTERISK-29896 - xmldocs: Add since tag
  (Reported by N
  A)
 * ASTERISK-29861 - asterisk.h: add macro for curl user agent
  
  (Reported by N A)
 * ASTERISK-29809 - curl, stir_shaken: refactor curl code
 
  (Reported by N A)
 * ASTERISK-29920 - app_voicemail: Warn if trying to manage
  nonexistent mailbox
  (Reported by N A)
 * ASTERISK-29925 - func_db: Warn about malformed key names

  (Reported by N A)
 * ASTERISK-29891 - [patch] provide a display name for RLS
  subscriptions
  (Reported by Alexei Gradinari)
 * ASTERISK-29866 - cli: add core dump information to core show
  settings
  (Reported by N A)
 * ASTERISK-29898 - documentation: Add default attributes to
  documentation
  (Reported by N A)
 * ASTERISK-29900 - app_mp3: Document and warn about https
  incompatibility
  (Reported by N A)
 * ASTERISK-29877 - app_mf: Allow reading a maximum number of
  digits
  (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-18.11.0

Thank you for your continued support of Asterisk!
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] Asterisk 16.25.0 Now Available

2022-03-24 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
16.25.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.25.0 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-29945 - pjproject: Security fixes for things
 
  (Reported by Kevin Harwell)

New Features made in this release:
---
 * ASTERISK-29853 - ami: Allow events to be globally disabled
  
  (Reported by N A)
 * ASTERISK-29840 - func_channel: Add LASTCONTEXT and LASTEXTEN
  fields
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-29924 - res_config_pgsql: omit "unsupported column
  type 'text'" error
  (Reported by Boris P. Korzun)
 * ASTERISK-29923 - docs, LICENSE: pbx.digium.com no longer
  exists
  (Reported by N A)
 * ASTERISK-29904 - RLS: Batched Notifications stop working

  (Reported by Alexei Gradinari)
 * ASTERISK-29365 - taskprocessor: Can cause assert at shutdown

  (Reported by Joshua C. Colp)
 * ASTERISK-29873 - [patch] Queue Realtime load
  (Reported
  by Alexei Gradinari)
 * ASTERISK-18416 - [patch] Realtime queue agents unavailable
  via AMI before a call event.
  (Reported by kwk)
 * ASTERISK-27597 - AMI Queuestatus not working (with realtime
  queue)
  (Reported by cagdas kopuz)
 * ASTERISK-29886 - Asterisk AMI sends not-valid XML
 
  (Reported by Napadailo Yaroslav)

Improvements made in this release:
---
 * ASTERISK-29906 - [patch] update RLS to reflect the changes to
  the lists
  (Reported by Alexei Gradinari)
 * ASTERISK-29909 - app_queue: Add support for withdrawing a
  call
  (Reported by Kfir Itzhak)
 * ASTERISK-29353 - Qualify jansson 2.14 for asterisk
 
  (Reported by George Joseph)
 * ASTERISK-29897 - channels: Increase core debug levels for
  chatty debugs
  (Reported by N A)
 * ASTERISK-29896 - xmldocs: Add since tag
  (Reported by N
  A)
 * ASTERISK-29861 - asterisk.h: add macro for curl user agent
  
  (Reported by N A)
 * ASTERISK-29920 - app_voicemail: Warn if trying to manage
  nonexistent mailbox
  (Reported by N A)
 * ASTERISK-29925 - func_db: Warn about malformed key names

  (Reported by N A)
 * ASTERISK-29809 - curl, stir_shaken: refactor curl code
 
  (Reported by N A)
 * ASTERISK-29891 - [patch] provide a display name for RLS
  subscriptions
  (Reported by Alexei Gradinari)
 * ASTERISK-29866 - cli: add core dump information to core show
  settings
  (Reported by N A)
 * ASTERISK-29898 - documentation: Add default attributes to
  documentation
  (Reported by N A)
 * ASTERISK-29900 - app_mp3: Document and warn about https
  incompatibility
  (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-16.25.0

Thank you for your continued support of Asterisk!
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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[asterisk-users] Asterisk 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13 Now Available (Security)

2022-03-04 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 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.

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-2022-004: pjproject: integer underflow on STUN message
  The header length on incoming STUN messages that contain an ERROR-CODE
  attribute is not properly checked. This can result in an integer underflow.
  Note, this requires ICE or WebRTC support to be in use with a malicious remote
  party.

* AST-2022-005: pjproject: undefined behavior after freeing a dialog set
  When acting as a UAC, and when placing an outgoing call to a target that then
  forks Asterisk may experience undefined behavior (crashes, hangs, etcâ¦)
  after a dialog set is prematurely freed.

* AST-2022-006: pjproject: unconstrained malformed multipart SIP message
  If an incoming SIP message contains a malformed multi-part body an out of
  bounds read access may occur, which can result in undefined behavior. Note,
  itâs currently uncertain if there is any externally exploitable vector
  within Asterisk for this issue, but providing this as a security issue out of
  caution.

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

https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13

The security advisories are available at:

https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
https://downloads.asterisk.org/pub/security/AST-2022-006.pdf

Thank you for your continued support of Asterisk!-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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[asterisk-users] Asterisk 19.2.0 Now Available

2022-02-10 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
19.2.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 19.2.0 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-29808 - cdr: allow disabling CDR by default
 
  (Reported by N A)
 * ASTERISK-29830 - ami: Add AMI event for Wink
  (Reported
  by N A)
 * ASTERISK-29802 - app_sf: Add full tech-agnostic SF support
  
  (Reported by N A)
 * ASTERISK-29759 - app_sendtext: Add ReceiveText application
  
  (Reported by N A)
 * ASTERISK-29706 - func_json: Add JSON parsing function
 
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-29888 - res_pjsip_outbound_authenticator_digest:
  ABRT attempting to clean up auth_sess
  (Reported by George
  Joseph)
 * ASTERISK-29854 - func_frame_drop: fix buffer usage typo
 
  (Reported by N A)
 * ASTERISK-29857 - res_tonedetect: fix logic errors in code
   
  (Reported by N A)
 * ASTERISK-29869 - rtp sequence number can skip after DTMF
  under certain bridges
  (Reported by Torrey Searle)
 * ASTERISK-29817 - gethostbyname_r is misdetected on NetBSD and
  causes a build failure
  (Reported by MichaÅ GÃ³rny)
 * ASTERISK-29698 - Segfault if sorcery object_lifetime_maximum
  and qualify_frequency the same value
  (Reported by Alexei
  Gradinari)
 * ASTERISK-29852 - make_version uses GNU-ism that break
  git-svn-id parsing on NetBSD
  (Reported by MichaÅ GÃ³rny)
 * ASTERISK-29850 - ast_get_tid() not implemented for NetBSD
   
  (Reported by MichaÅ GÃ³rny)
 * ASTERISK-29851 - rdtsc is not enabled (stubbed out) on
  NetBSD
  (Reported by MichaÅ GÃ³rny)
 * ASTERISK-29818 - Build failure on NetBSD due to hmac function
  collision
  (Reported by MichaÅ GÃ³rny)
 * ASTERISK-29856 - res_rtp_asterisk: Invalid comparison creates
  unreachable code
  (Reported by N A)
 * ASTERISK-29867 - configure fails if libsrtp dev files are not
  installed
  (Reported by Sean Bright)
 * ASTERISK-29813 - res_pjsip_session doesn't support multipart
  message bodies
  (Reported by George Joseph)
 * ASTERISK-29858 - Regression:  Using external pjproject not
  working after "hack" commit
  (Reported by George Joseph)
 * ASTERISK-29859 - VoiceMailMain() fails when encountering
  non-numeric CALLERID(num)
  (Reported by Mark Murawski)
 * ASTERISK-29847 - pbx_variables: ASTSBINDIR is missing
 
  (Reported by N A)
 * ASTERISK-29824 - It's hard to make changes to bundled
  pjproject
  (Reported by George Joseph)
 * ASTERISK-29695 - SAY.CONF wrong logic when converting 24hour
  time to say 12 hour am/pm
  (Reported by Vincent Dubois)
 * ASTERISK-29664 - PJSIP processing token with % incorrectly
  
  (Reported by Dan Cropp)
 * ASTERISK-29827 - Support for Nordic language syntax in
  Queues
  (Reported by Mark Petersen)
 * ASTERISK-29515 - app_queue: QueueSummary and QueueStatus
  events don't exist in documentation
  (Reported by Luke
  Escude)
 * ASTERISK-29746 - tcptls.c: TCP client connect fails due to
  interrupt
  (Reported by Kevin Harwell)
 * ASTERISK-29806 - app_queue: extension state incorrect
 
  (Reported by Steve Davies)
 * ASTERISK-29816 - SAY_DTMF_INTERRUPT channel variable is not
  honored
  (Reported by Sean Bright)
 * ASTERISK-28863 - The ast_rtp_codecs_payloads functions don't
  preserve order
  (Reported by George Joseph)
 * ASTERISK-29320 - res_pjsip_sdp_rtp: Codec preference order of
  remote is not correct on unhold
  (Reported by Ross Beer)
 * ASTERISK-29821 - Deadlock in bridge_channel_internal_join()
  on local channels.
  (Reported by Krzysztof Trempala)
 * ASTERISK-29722 - test_timezone_watch breaks during DST to ST
  transition
  (Reported by Josh Soref)
 * ASTERISK-29804 - bundled_pjproject: sip_inv is missing
  multipart support in some cases
  (Reported by George
  Joseph)
 * ASTERISK-29794 - ast_coredumper does not delete results when
  requested and a specific output dir is set
  (Reported by
  Frederic Van Espen)
 * ASTERISK-29803 - pbx_variables: cp4 variables is used
  uninitialized
  (Reported by N A)
 * ASTERISK-29766 - pbx_variables: MSet truncates sets after 24
  variables
  (Reported by N A)
 * ASTERISK-29772 - chan_sip: ${CHANNEL(ruri)} in Dial/Queue
  b(test,s,1) cause a coredump
  (Reported by Mark Petersen)
 * ASTERISK-29790 - xmldoc: Dump invalid to XML DTD: XSLT
 
  (Reported by Alexander Traud)
 * ASTERISK-29791 - xmldoc: Dump invalid to XM

[asterisk-users] Asterisk 18.10.0 Now Available

2022-02-10 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
18.10.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.10.0 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-29808 - cdr: allow disabling CDR by default
 
  (Reported by N A)
 * ASTERISK-29830 - ami: Add AMI event for Wink
  (Reported
  by N A)
 * ASTERISK-29802 - app_sf: Add full tech-agnostic SF support
  
  (Reported by N A)
 * ASTERISK-29759 - app_sendtext: Add ReceiveText application
  
  (Reported by N A)
 * ASTERISK-29706 - func_json: Add JSON parsing function
 
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-29888 - res_pjsip_outbound_authenticator_digest:
  ABRT attempting to clean up auth_sess
  (Reported by George
  Joseph)
 * ASTERISK-29857 - res_tonedetect: fix logic errors in code
   
  (Reported by N A)
 * ASTERISK-29854 - func_frame_drop: fix buffer usage typo
 
  (Reported by N A)
 * ASTERISK-29869 - rtp sequence number can skip after DTMF
  under certain bridges
  (Reported by Torrey Searle)
 * ASTERISK-29817 - gethostbyname_r is misdetected on NetBSD and
  causes a build failure
  (Reported by MichaÅ GÃ³rny)
 * ASTERISK-29698 - Segfault if sorcery object_lifetime_maximum
  and qualify_frequency the same value
  (Reported by Alexei
  Gradinari)
 * ASTERISK-29851 - rdtsc is not enabled (stubbed out) on
  NetBSD
  (Reported by MichaÅ GÃ³rny)
 * ASTERISK-29852 - make_version uses GNU-ism that break
  git-svn-id parsing on NetBSD
  (Reported by MichaÅ GÃ³rny)
 * ASTERISK-29850 - ast_get_tid() not implemented for NetBSD
   
  (Reported by MichaÅ GÃ³rny)
 * ASTERISK-29818 - Build failure on NetBSD due to hmac function
  collision
  (Reported by MichaÅ GÃ³rny)
 * ASTERISK-29856 - res_rtp_asterisk: Invalid comparison creates
  unreachable code
  (Reported by N A)
 * ASTERISK-29867 - configure fails if libsrtp dev files are not
  installed
  (Reported by Sean Bright)
 * ASTERISK-29813 - res_pjsip_session doesn't support multipart
  message bodies
  (Reported by George Joseph)
 * ASTERISK-29858 - Regression:  Using external pjproject not
  working after "hack" commit
  (Reported by George Joseph)
 * ASTERISK-29859 - VoiceMailMain() fails when encountering
  non-numeric CALLERID(num)
  (Reported by Mark Murawski)
 * ASTERISK-29847 - pbx_variables: ASTSBINDIR is missing
 
  (Reported by N A)
 * ASTERISK-29824 - It's hard to make changes to bundled
  pjproject
  (Reported by George Joseph)
 * ASTERISK-29695 - SAY.CONF wrong logic when converting 24hour
  time to say 12 hour am/pm
  (Reported by Vincent Dubois)
 * ASTERISK-29664 - PJSIP processing token with % incorrectly
  
  (Reported by Dan Cropp)
 * ASTERISK-29827 - Support for Nordic language syntax in
  Queues
  (Reported by Mark Petersen)
 * ASTERISK-29515 - app_queue: QueueSummary and QueueStatus
  events don't exist in documentation
  (Reported by Luke
  Escude)
 * ASTERISK-29746 - tcptls.c: TCP client connect fails due to
  interrupt
  (Reported by Kevin Harwell)
 * ASTERISK-29806 - app_queue: extension state incorrect
 
  (Reported by Steve Davies)
 * ASTERISK-29816 - SAY_DTMF_INTERRUPT channel variable is not
  honored
  (Reported by Sean Bright)
 * ASTERISK-28863 - The ast_rtp_codecs_payloads functions don't
  preserve order
  (Reported by George Joseph)
 * ASTERISK-29320 - res_pjsip_sdp_rtp: Codec preference order of
  remote is not correct on unhold
  (Reported by Ross Beer)
 * ASTERISK-29821 - Deadlock in bridge_channel_internal_join()
  on local channels.
  (Reported by Krzysztof Trempala)
 * ASTERISK-29722 - test_timezone_watch breaks during DST to ST
  transition
  (Reported by Josh Soref)
 * ASTERISK-29804 - bundled_pjproject: sip_inv is missing
  multipart support in some cases
  (Reported by George
  Joseph)
 * ASTERISK-29794 - ast_coredumper does not delete results when
  requested and a specific output dir is set
  (Reported by
  Frederic Van Espen)
 * ASTERISK-29803 - pbx_variables: cp4 variables is used
  uninitialized
  (Reported by N A)
 * ASTERISK-29766 - pbx_variables: MSet truncates sets after 24
  variables
  (Reported by N A)
 * ASTERISK-29772 - chan_sip: ${CHANNEL(ruri)} in Dial/Queue
  b(test,s,1) cause a coredump
  (Reported by Mark Petersen)
 * ASTERISK-29790 - xmldoc: Dump invalid to XML DTD: XSLT
 
  (Reported by Alexander Traud)
 * ASTERISK-29791 - xmldoc: Dump invalid to XM

[asterisk-users] Asterisk 16.24.0 Now Available

2022-02-10 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
16.24.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.24.0 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-29808 - cdr: allow disabling CDR by default
 
  (Reported by N A)
 * ASTERISK-29830 - ami: Add AMI event for Wink
  (Reported
  by N A)
 * ASTERISK-29802 - app_sf: Add full tech-agnostic SF support
  
  (Reported by N A)
 * ASTERISK-29759 - app_sendtext: Add ReceiveText application
  
  (Reported by N A)
 * ASTERISK-29706 - func_json: Add JSON parsing function
 
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-29888 - res_pjsip_outbound_authenticator_digest:
  ABRT attempting to clean up auth_sess
  (Reported by George
  Joseph)
 * ASTERISK-29854 - func_frame_drop: fix buffer usage typo
 
  (Reported by N A)
 * ASTERISK-29857 - res_tonedetect: fix logic errors in code
   
  (Reported by N A)
 * ASTERISK-29869 - rtp sequence number can skip after DTMF
  under certain bridges
  (Reported by Torrey Searle)
 * ASTERISK-29817 - gethostbyname_r is misdetected on NetBSD and
  causes a build failure
  (Reported by MichaÅ GÃ³rny)
 * ASTERISK-29698 - Segfault if sorcery object_lifetime_maximum
  and qualify_frequency the same value
  (Reported by Alexei
  Gradinari)
 * ASTERISK-29851 - rdtsc is not enabled (stubbed out) on
  NetBSD
  (Reported by MichaÅ GÃ³rny)
 * ASTERISK-29852 - make_version uses GNU-ism that break
  git-svn-id parsing on NetBSD
  (Reported by MichaÅ GÃ³rny)
 * ASTERISK-29850 - ast_get_tid() not implemented for NetBSD
   
  (Reported by MichaÅ GÃ³rny)
 * ASTERISK-29818 - Build failure on NetBSD due to hmac function
  collision
  (Reported by MichaÅ GÃ³rny)
 * ASTERISK-29867 - configure fails if libsrtp dev files are not
  installed
  (Reported by Sean Bright)
 * ASTERISK-29856 - res_rtp_asterisk: Invalid comparison creates
  unreachable code
  (Reported by N A)
 * ASTERISK-29813 - res_pjsip_session doesn't support multipart
  message bodies
  (Reported by George Joseph)
 * ASTERISK-29858 - Regression:  Using external pjproject not
  working after "hack" commit
  (Reported by George Joseph)
 * ASTERISK-29859 - VoiceMailMain() fails when encountering
  non-numeric CALLERID(num)
  (Reported by Mark Murawski)
 * ASTERISK-29847 - pbx_variables: ASTSBINDIR is missing
 
  (Reported by N A)
 * ASTERISK-29824 - It's hard to make changes to bundled
  pjproject
  (Reported by George Joseph)
 * ASTERISK-29695 - SAY.CONF wrong logic when converting 24hour
  time to say 12 hour am/pm
  (Reported by Vincent Dubois)
 * ASTERISK-29664 - PJSIP processing token with % incorrectly
  
  (Reported by Dan Cropp)
 * ASTERISK-29827 - Support for Nordic language syntax in
  Queues
  (Reported by Mark Petersen)
 * ASTERISK-29515 - app_queue: QueueSummary and QueueStatus
  events don't exist in documentation
  (Reported by Luke
  Escude)
 * ASTERISK-29746 - tcptls.c: TCP client connect fails due to
  interrupt
  (Reported by Kevin Harwell)
 * ASTERISK-29806 - app_queue: extension state incorrect
 
  (Reported by Steve Davies)
 * ASTERISK-29816 - SAY_DTMF_INTERRUPT channel variable is not
  honored
  (Reported by Sean Bright)
 * ASTERISK-29821 - Deadlock in bridge_channel_internal_join()
  on local channels.
  (Reported by Krzysztof Trempala)
 * ASTERISK-29722 - test_timezone_watch breaks during DST to ST
  transition
  (Reported by Josh Soref)
 * ASTERISK-29804 - bundled_pjproject: sip_inv is missing
  multipart support in some cases
  (Reported by George
  Joseph)
 * ASTERISK-29794 - ast_coredumper does not delete results when
  requested and a specific output dir is set
  (Reported by
  Frederic Van Espen)
 * ASTERISK-29803 - pbx_variables: cp4 variables is used
  uninitialized
  (Reported by N A)
 * ASTERISK-29766 - pbx_variables: MSet truncates sets after 24
  variables
  (Reported by N A)
 * ASTERISK-29772 - chan_sip: ${CHANNEL(ruri)} in Dial/Queue
  b(test,s,1) cause a coredump
  (Reported by Mark Petersen)
 * ASTERISK-29790 - xmldoc: Dump invalid to XML DTD: XSLT
 
  (Reported by Alexander Traud)
 * ASTERISK-29791 - xmldoc: Dump invalid to XML DTD: ACO
  Matchfield
  (Reported by Alexander Traud)
 * ASTERISK-26991 - documentation: Doxygen site is no longer
  being updated
  (Reported by Joshua C. Colp)
 * ASTERISK-20259 - [patch] Update Doxygen Configuration for
  mak

[asterisk-users] Asterisk Community Services are Down

2022-01-03 Thread Asterisk Development Team

This includes Jira, the Wiki and Gerrit due to loss of internet access.  If
you're currently signed in to the community forums, you're OK but new
logins won't be accepted. The IRC channels are unaffected.

Probably has something to do with this in Huntsville:
[image: image.png]


-- 
George Joseph
Asterisk Software Developer
Check us out at www.sangoma.com and www.asterisk.org
[image: image.png]
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] Asterisk 19.1.0 Now Available

2021-12-09 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
19.1.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 19.1.0 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-29720 - res_tonedetect: Add call progress tone
  detection
  (Reported by N A)
 * ASTERISK-18069 - [patch] app_queue Add Login Time and Last
  Paused Times to Queue Members
  (Reported by Jamuel Starkey)

Bugs fixed in this release:
---
 * ASTERISK-29779 - progdocs: Hidden code sections with syntax
  errors.
  (Reported by Alexander Traud)
 * ASTERISK-29732 - progdocs: Fix grouping for latest Doxygen
  
  (Reported by Alexander Traud)
 * ASTERISK-29771 - Crash occurs when 2 realtime sippeers mysql
  connections are configured and we have a schema warning
 
  (Reported by Mario Ban)
 * ASTERISK-29776 - stir/shaken: Requires GNU designator
 
  (Reported by Alexander Traud)
 * ASTERISK-29773 - progdocs: doxyref.h outdated
  (Reported
  by Alexander Traud)
 * ASTERISK-29765 - xmldoc: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29762 - channels: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29730 - Segfault in __ao2_ref if refdebug = yes

  (Reported by Alexei Gradinari)
 * ASTERISK-29748 - bridging: Infinite loop when both Local
  channel halves in same bridge
  (Reported by Joshua C. Colp)
 * ASTERISK-29754 - odbc: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29753 - parking: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29756 - res_ari: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29755 - frame: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29751 - channel: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29752 - app: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29750 - stasis: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29749 - res_xmpp: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29742 - addons: Fix for Doxygen.
  (Reported by
  Alexander Traud)
 * ASTERISK-29747 - res_pjsip: Fix for Doxygen
  (Reported
  by Alexander Traud)
 * ASTERISK-29737 - chan_iax2: Fix for Doxygen
  (Reported
  by Alexander Traud)
 * ASTERISK-29743 - bridges: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29740 - apps: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29741 - tests: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29736 - bridge_channel: Fix for Doxygen
 
  (Reported by Alexander Traud)
 * ASTERISK-29734 - progdocs: Use Doxygen \example correctly
   
  (Reported by Alexander Traud)
 * ASTERISK-29735 - progdocs: Avoid multiple use of section
  labels
  (Reported by Alexander Traud)
 * ASTERISK-29733 - progdocs: Avoid name with Doxygen \file

  (Reported by Alexander Traud)
 * ASTERISK-29744 - app_morsecode: Fix deadlock
  (Reported
  by N A)
 * ASTERISK-29705 - app_read: Fix custom terminator
  functionality regression
  (Reported by N A)
 * ASTERISK-29703 - res_pjsip_callerid: Fix OLI parsing
 
  (Reported by N A)
 * ASTERISK-29724 - BuildSystem: In POSIX sh, == in place of =
  is undefined.
  (Reported by Alexander Traud)
 * ASTERISK-29702 - sig_analog: Fix truncated buffer copy
 
  (Reported by N A)
 * ASTERISK-28040 - pbx: "dialplan reload" is removing minus
  symbol from dynamic hints
  (Reported by Daniel Zanutti)
 * ASTERISK-29391 - VoiceMail does not cancel recording on
  rerecord hangup
  (Reported by N A)
 * ASTERISK-29709 - res_snmp: Not build on recent Debian
  distributions.
  (Reported by Alexander Traud)
 * ASTERISK-29717 - res_config_sqlite: not removed in
  makeopts.in
  (Reported by Alexander Traud)
 * ASTERISK-29710 - stasis: Clang 13 warns about the unused but
  set variable dispatched.
  (Reported by Alexander Traud)
 * ASTERISK-29711 - aelparse: GCC 11.2 found two maybe
  uninitialized
  (Reported by Alexander Traud)
 * ASTERISK-29713 - GCC 11.2: two stringop-overread
 
  (Reported by Alexander Traud)
 * ASTERISK-29682 - Squash compiler issues generated by gcc 11
 
  (Reported by George Joseph)
 * ASTERISK-29693 - Using --with-crypto and --with-ssl fails on
  a recompile
  (Reported by George Joseph)
 * ASTERISK-27816 - func_talkdetect's logic is completely
  broken
  (Reported by Moritz Fain)
 * ASTERISK-26497 - make install downlo

[asterisk-users] Asterisk 18.9.0 Now Available

2021-12-09 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
18.9.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.9.0 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-29720 - res_tonedetect: Add call progress tone
  detection
  (Reported by N A)
 * ASTERISK-18069 - [patch] app_queue Add Login Time and Last
  Paused Times to Queue Members
  (Reported by Jamuel Starkey)

Bugs fixed in this release:
---
 * ASTERISK-29779 - progdocs: Hidden code sections with syntax
  errors.
  (Reported by Alexander Traud)
 * ASTERISK-29732 - progdocs: Fix grouping for latest Doxygen
  
  (Reported by Alexander Traud)
 * ASTERISK-29771 - Crash occurs when 2 realtime sippeers mysql
  connections are configured and we have a schema warning
 
  (Reported by Mario Ban)
 * ASTERISK-29776 - stir/shaken: Requires GNU designator
 
  (Reported by Alexander Traud)
 * ASTERISK-29764 - chan_misdn: Fix for Doxygen
  (Reported
  by Alexander Traud)
 * ASTERISK-29773 - progdocs: doxyref.h outdated
  (Reported
  by Alexander Traud)
 * ASTERISK-29765 - xmldoc: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29730 - Segfault in __ao2_ref if refdebug = yes

  (Reported by Alexei Gradinari)
 * ASTERISK-29762 - channels: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29748 - bridging: Infinite loop when both Local
  channel halves in same bridge
  (Reported by Joshua C. Colp)
 * ASTERISK-29754 - odbc: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29753 - parking: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29755 - frame: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29756 - res_ari: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29751 - channel: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29750 - stasis: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29752 - app: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29749 - res_xmpp: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29742 - addons: Fix for Doxygen.
  (Reported by
  Alexander Traud)
 * ASTERISK-29747 - res_pjsip: Fix for Doxygen
  (Reported
  by Alexander Traud)
 * ASTERISK-29737 - chan_iax2: Fix for Doxygen
  (Reported
  by Alexander Traud)
 * ASTERISK-29743 - bridges: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29741 - tests: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29740 - apps: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29733 - progdocs: Avoid name with Doxygen \file

  (Reported by Alexander Traud)
 * ASTERISK-29736 - bridge_channel: Fix for Doxygen
 
  (Reported by Alexander Traud)
 * ASTERISK-29735 - progdocs: Avoid multiple use of section
  labels
  (Reported by Alexander Traud)
 * ASTERISK-29734 - progdocs: Use Doxygen \example correctly
   
  (Reported by Alexander Traud)
 * ASTERISK-29744 - app_morsecode: Fix deadlock
  (Reported
  by N A)
 * ASTERISK-29703 - res_pjsip_callerid: Fix OLI parsing
 
  (Reported by N A)
 * ASTERISK-29705 - app_read: Fix custom terminator
  functionality regression
  (Reported by N A)
 * ASTERISK-29724 - BuildSystem: In POSIX sh, == in place of =
  is undefined.
  (Reported by Alexander Traud)
 * ASTERISK-29702 - sig_analog: Fix truncated buffer copy
 
  (Reported by N A)
 * ASTERISK-28040 - pbx: "dialplan reload" is removing minus
  symbol from dynamic hints
  (Reported by Daniel Zanutti)
 * ASTERISK-29391 - VoiceMail does not cancel recording on
  rerecord hangup
  (Reported by N A)
 * ASTERISK-29709 - res_snmp: Not build on recent Debian
  distributions.
  (Reported by Alexander Traud)
 * ASTERISK-29710 - stasis: Clang 13 warns about the unused but
  set variable dispatched.
  (Reported by Alexander Traud)
 * ASTERISK-29711 - aelparse: GCC 11.2 found two maybe
  uninitialized
  (Reported by Alexander Traud)
 * ASTERISK-29713 - GCC 11.2: two stringop-overread
 
  (Reported by Alexander Traud)
 * ASTERISK-29682 - Squash compiler issues generated by gcc 11
 
  (Reported by George Joseph)
 * ASTERISK-29693 - Using --with-crypto and --with-ssl fails on
  a recompile
  (Reported by George Joseph)
 * ASTERISK-27816 - func_talkdetect's logic is completely
  broken
  (Reported by Moritz Fain)
 * ASTERISK-29691 - stun: Not all users pro

[asterisk-users] Asterisk 16.23.0 Now Available

2021-12-09 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
16.23.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.23.0 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-29720 - res_tonedetect: Add call progress tone
  detection
  (Reported by N A)
 * ASTERISK-18069 - [patch] app_queue Add Login Time and Last
  Paused Times to Queue Members
  (Reported by Jamuel Starkey)

Bugs fixed in this release:
---
 * ASTERISK-29779 - progdocs: Hidden code sections with syntax
  errors.
  (Reported by Alexander Traud)
 * ASTERISK-29732 - progdocs: Fix grouping for latest Doxygen
  
  (Reported by Alexander Traud)
 * ASTERISK-29771 - Crash occurs when 2 realtime sippeers mysql
  connections are configured and we have a schema warning
 
  (Reported by Mario Ban)
 * ASTERISK-29776 - stir/shaken: Requires GNU designator
 
  (Reported by Alexander Traud)
 * ASTERISK-29773 - progdocs: doxyref.h outdated
  (Reported
  by Alexander Traud)
 * ASTERISK-29765 - xmldoc: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29764 - chan_misdn: Fix for Doxygen
  (Reported
  by Alexander Traud)
 * ASTERISK-29762 - channels: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29730 - Segfault in __ao2_ref if refdebug = yes

  (Reported by Alexei Gradinari)
 * ASTERISK-29748 - bridging: Infinite loop when both Local
  channel halves in same bridge
  (Reported by Joshua C. Colp)
 * ASTERISK-29753 - parking: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29754 - odbc: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29756 - res_ari: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29755 - frame: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29751 - channel: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29750 - stasis: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29752 - app: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29749 - res_xmpp: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29737 - chan_iax2: Fix for Doxygen
  (Reported
  by Alexander Traud)
 * ASTERISK-29747 - res_pjsip: Fix for Doxygen
  (Reported
  by Alexander Traud)
 * ASTERISK-29743 - bridges: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29742 - addons: Fix for Doxygen.
  (Reported by
  Alexander Traud)
 * ASTERISK-29741 - tests: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29740 - apps: Fix for Doxygen
  (Reported by
  Alexander Traud)
 * ASTERISK-29736 - bridge_channel: Fix for Doxygen
 
  (Reported by Alexander Traud)
 * ASTERISK-29733 - progdocs: Avoid name with Doxygen \file

  (Reported by Alexander Traud)
 * ASTERISK-29734 - progdocs: Use Doxygen \example correctly
   
  (Reported by Alexander Traud)
 * ASTERISK-29735 - progdocs: Avoid multiple use of section
  labels
  (Reported by Alexander Traud)
 * ASTERISK-29744 - app_morsecode: Fix deadlock
  (Reported
  by N A)
 * ASTERISK-29705 - app_read: Fix custom terminator
  functionality regression
  (Reported by N A)
 * ASTERISK-29703 - res_pjsip_callerid: Fix OLI parsing
 
  (Reported by N A)
 * ASTERISK-29702 - sig_analog: Fix truncated buffer copy
 
  (Reported by N A)
 * ASTERISK-29724 - BuildSystem: In POSIX sh, == in place of =
  is undefined.
  (Reported by Alexander Traud)
 * ASTERISK-28040 - pbx: "dialplan reload" is removing minus
  symbol from dynamic hints
  (Reported by Daniel Zanutti)
 * ASTERISK-29391 - VoiceMail does not cancel recording on
  rerecord hangup
  (Reported by N A)
 * ASTERISK-29709 - res_snmp: Not build on recent Debian
  distributions.
  (Reported by Alexander Traud)
 * ASTERISK-29710 - stasis: Clang 13 warns about the unused but
  set variable dispatched.
  (Reported by Alexander Traud)
 * ASTERISK-29711 - aelparse: GCC 11.2 found two maybe
  uninitialized
  (Reported by Alexander Traud)
 * ASTERISK-29713 - GCC 11.2: two stringop-overread
 
  (Reported by Alexander Traud)
 * ASTERISK-29682 - Squash compiler issues generated by gcc 11
 
  (Reported by George Joseph)
 * ASTERISK-29693 - Using --with-crypto and --with-ssl fails on
  a recompile
  (Reported by George Joseph)
 * ASTERISK-27816 - func_talkdetect's logic is completely
  broken
  (Reported by Moritz Fain)
 * ASTERISK-29691 - stun: Not all users pro

[asterisk-users] Asterisk 19.0.0 Now Available

2021-11-02 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
19.0.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 19.0.0 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-29381 - chan_pjsip: Remote denial of service by an
  authenticated user
  (Reported by Ivan Poddubny)
 * ASTERISK-29415 - Crash in PJSIP TLS transport 
 
  (Reported by Andrew Yager)
 * ASTERISK-29305 - ASTERISK-29203 / AST-2021-002 -- Another
  scenario is causing a crash
  (Reported by Gregory Massel)
 * ASTERISK-29260 - sRTP Replay Protection ignored; even tears
  down long calls
  (Reported by Alexander Traud)
 * ASTERISK-29227 - res_pjsip_diversion: sending multiple 181
  responses causes memory corruption and crash
  (Reported by
  Ivan Poddubny)
 * ASTERISK-29219 - res_pjsip_diversion: Crash if Tel URI
  contains History-Info
  (Reported by Torrey Searle)
 * ASTERISK-29057 - pjsip: Crash on call rejection during high
  load
  (Reported by Sandro Gauci)

New Features made in this release:
---
 * ASTERISK-29656 - Add CHANNEL_EXISTS function
  (Reported
  by N A)
 * ASTERISK-29496 - Add SendMF application
  (Reported by N
  A)
 * ASTERISK-29627 - Add STRBETWEEN function
  (Reported by N
  A)
 * ASTERISK-29628 - Add file and directory functions
 
  (Reported by N A)
 * ASTERISK-29531 - Add SAYFILES function
  (Reported by N
  A)
 * ASTERISK-29546 - Add tone detection module
  (Reported by
  N A)
 * ASTERISK-18454 - Option for Read to be able to accept #
 
  (Reported by Sta Retji)
 * ASTERISK-29542 - Add audio scrambler
  (Reported by N A)
 * ASTERISK-29478 - Function to drop frames in the TX or RX

[asterisk-users] Asterisk 18.8.0 Now Available

2021-11-02 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
18.8.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.8.0 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-29656 - Add CHANNEL_EXISTS function
  (Reported
  by N A)

Bugs fixed in this release:
---
 * ASTERISK-20219 - [patch] - IAX2 Call Encryption Fails with
  RSA authentication
  (Reported by Michael Munger)
 * ASTERISK-29402 - res_pjsip_t38: Socket is bound to IPv4/IPv6
  but platform does not support it
  (Reported by Matthew
  Kern)
 * ASTERISK-29673 - app_read: Fix null pointer crash regression

  (Reported by N A)
 * ASTERISK-29671 - res_rtp_asterisk: memory leak
 
  (Reported by Jean Aunis - Prescom)
 * ASTERISK-29668 - ari: Listing bridges fails when dialing
  bridge exists
  (Reported by Joshua C. Colp)
 * ASTERISK-29663 - messaging: AMI MessageSend does not support
  same parameters as dialplan application
  (Reported by Brian
  J. Murrell)
 * ASTERISK-29578 - app_queue: Custom device state using
  included hints do not update
  (Reported by N A)
 * ASTERISK-29660 - Build failure when disabling PJSIP support
 
  (Reported by Guido Falsi)

Improvements made in this release:
---
 * ASTERISK-29637 - Add support for future dates in Say.c
 
  (Reported by Shloime Rosenblum)
 * ASTERISK-29525 - PJSIP remove_existing unavailable contacts
 
  (Reported by Joseph Nadiv)
 * ASTERISK-29661 - func_vmcount: Add support for multiple
  mailboxes
  (Reported by N A)
 * ASTERISK-29275 - Support of MIME-type for wav16
 
  (Reported by Boris P. Korzun)
 * ASTERISK-29529 - Add custom logging level
  (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-18.8.0

Thank you for your continued support of Asterisk!
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Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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[asterisk-users] Asterisk 16.22.0 Now Available

2021-11-02 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
16.22.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.22.0 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-29656 - Add CHANNEL_EXISTS function
  (Reported
  by N A)

Bugs fixed in this release:
---
 * ASTERISK-20219 - [patch] - IAX2 Call Encryption Fails with
  RSA authentication
  (Reported by Michael Munger)
 * ASTERISK-29402 - res_pjsip_t38: Socket is bound to IPv4/IPv6
  but platform does not support it
  (Reported by Matthew
  Kern)
 * ASTERISK-29673 - app_read: Fix null pointer crash regression

  (Reported by N A)
 * ASTERISK-29671 - res_rtp_asterisk: memory leak
 
  (Reported by Jean Aunis - Prescom)
 * ASTERISK-29663 - messaging: AMI MessageSend does not support
  same parameters as dialplan application
  (Reported by Brian
  J. Murrell)
 * ASTERISK-29578 - app_queue: Custom device state using
  included hints do not update
  (Reported by N A)
 * ASTERISK-29660 - Build failure when disabling PJSIP support
 
  (Reported by Guido Falsi)

Improvements made in this release:
---
 * ASTERISK-29637 - Add support for future dates in Say.c
 
  (Reported by Shloime Rosenblum)
 * ASTERISK-29525 - PJSIP remove_existing unavailable contacts
 
  (Reported by Joseph Nadiv)
 * ASTERISK-29661 - func_vmcount: Add support for multiple
  mailboxes
  (Reported by N A)
 * ASTERISK-29275 - Support of MIME-type for wav16
 
  (Reported by Boris P. Korzun)
 * ASTERISK-29529 - Add custom logging level
  (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-16.22.0

Thank you for your continued support of Asterisk!
-- 
_
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Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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[asterisk-users] Certified Asterisk 16.8-cert12 Now Available

2021-10-21 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Certified 
Asterisk 16.8-cert12.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/certified-asterisk

The release of Certified Asterisk 16.8-cert12 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-29693 - Using --with-crypto and --with-ssl fails on
  a recompile
  (Reported by George Joseph)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-16.8-cert12

Thank you for your continued support of Asterisk!
-- 
_
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Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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[asterisk-users] Asterisk 18.7.1 Now Available

2021-10-11 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
18.7.1.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.7.1 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-29685 - pbx_ael: Infinite loop on reload
 
  (Reported by Joshua C. Colp)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1

Thank you for your continued support of Asterisk!
-- 
_
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Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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[asterisk-users] Asterisk 16.21.1 Now Available

2021-10-11 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
16.21.1.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.21.1 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-29685 - pbx_ael: Infinite loop on reload
 
  (Reported by Joshua C. Colp)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1

Thank you for your continued support of Asterisk!
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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[asterisk-users] Asterisk 18.7.0 Now Available

2021-10-07 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
18.7.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.7.0 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:

Deprecations made in this release:
---
 * 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-29554 - cdr_mysql: Deprecated in 1.8, to be removed
  in 19
  (Reported by Joshua C. Colp)
 * ASTERISK-29555 - app_mysql: Deprecated in 1.8, to be removed
  in 19
  (Reported by Joshua C. Colp)
 * ASTERISK-29557 - app_ices: Deprecated in 16, to be removed in
  19
  (Reported by Joshua C. Colp)
 * ASTERISK-29558 - app_macro: Deprecated in 16, to be removed
  in 21
  (Reported by Joshua C. Colp)
 * ASTERISK-29559 - app_fax: Deprecated in 16, to be removed in
  19
  (Reported by Joshua C. Colp)
 * ASTERISK-29560 - app_url: Deprecated in 16, to be removed in
  19
  (Reported by Joshua C. Colp)
 * ASTERISK-29561 - app_image: Deprecated in 16, to be removed
  in 19
  (Reported by Joshua C. Colp)
 * ASTERISK-29562 - app_nbscat: Deprecated in 16, to be removed
  in 19
  (Reported by Joshua C. Colp)
 * ASTERISK-29563 - app_dahdiras: Deprecated in 16, to be
  removed in 19
  (Reported by Joshua C. Colp)
 * ASTERISK-29564 - cdr_syslog: Deprecated in 16, to be removed
  in 19
  (Reported by Joshua C. Colp)
 * ASTERISK-29565 - chan_oss: Deprecated in 16, to be removed in
  19
  (Reported by Joshua C. Colp)
 * ASTERISK-29566 - chan_phone: Deprecated in 16, to be removed
  in 19
  (Reported by Joshua C. Colp)
 * ASTERISK-29567 - chan_sip: Deprecated in 17, to be removed in
  21
  (Reported by Joshua C. Colp)
 * ASTERISK-29568 - chan_nbs: Deprecated in 16, to be removed in
  19
  (Reported by Joshua C. Colp)
 * ASTERISK-29569 - chan_misdn: Deprecated in 16, to be removed
  in 19
  (Reported by Joshua C. Colp)
 * ASTERISK-29570 - chan_vpb: Deprecated in 16, to be removed in
  19
  (Reported by Joshua C. Colp)
 * ASTERISK-29571 - res_config_sqlite: Deprecated in 16, to be
  removed in 19
  (Reported by Joshua C. Colp)
 * ASTERISK-29572 - res_monitor: Deprecated in 16, to be removed
  in 21
  (Reported by Joshua C. Colp)
 * ASTERISK-29573 - conf2ael: Deprecated in 16, to be removed in
  19
  (Reported by Joshua C. Colp)
 * ASTERISK-29574 - muted: Deprecated in 16, to be removed in
  19
  (Reported by Joshua C. Colp)

New Features made in this release:
---
 * ASTERISK-29496 - Add SendMF application
  (Reported by N
  A)
 * ASTERISK-29627 - Add STRBETWEEN function
  (Reported by N
  A)
 * ASTERISK-29628 - Add file and directory functions
 
  (Reported by N A)
 * ASTERISK-29531 - Add SAYFILES function
  (Reported by N
  A)
 * ASTERISK-29546 - Add tone detection module
  (Reported by
  N A)
 * ASTERISK-18454 - Option for Read to be able to accept #
 
  (Reported by Sta Retji)
 * ASTERISK-29542 - Add audio scrambler
  (Reported by N A)
 * ASTERISK-29478 - Function to drop frames in the TX or RX
  directions
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-29673 - app_read: Fix null pointer crash regression

  (Reported by N A)
 * ASTERISK-29660 - Build failure when disabling PJSIP support
 
  (Reported by Guido Falsi)
 * ASTERISK-29635 - MP3Player don' t work with actual mpg123
  versions
  (Reported by Carlos Oliva)
 * ASTERISK-29654 - pjproject includes trailing whitespace in
  sdp format attributes
  (Reported by George Joseph)
 * ASTERISK-29629 - ARI external media channel creation doesn't
  set option data
  (Reported by sungtae kim)
 * ASTERISK-27176 - test_abstract_jb: frames leak
 
  (Reported by Corey Farrell)
 * ASTERISK-29634 - res_snmp:  gcc 11 needs -fPIC to compile
  correctly
  (Reported by George Joseph)
 * ASTERISK-29630 - Asterisk is unable to read extended number
  format terminfo files
  (Reported by Sean Bright)
 * ASTERISK-28004

[asterisk-users] Asterisk 16.21.0 Now Available

2021-10-07 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
16.21.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.21.0 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-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-29554 - cdr_mysql: Deprecated in 1.8, to be removed
  in 19
  (Reported by Joshua C. Colp)
 * ASTERISK-29555 - app_mysql: Deprecated in 1.8, to be removed
  in 19
  (Reported by Joshua C. Colp)
 * ASTERISK-29557 - app_ices: Deprecated in 16, to be removed in
  19
  (Reported by Joshua C. Colp)
 * ASTERISK-29558 - app_macro: Deprecated in 16, to be removed
  in 21
  (Reported by Joshua C. Colp)
 * ASTERISK-29559 - app_fax: Deprecated in 16, to be removed in
  19
  (Reported by Joshua C. Colp)
 * ASTERISK-29560 - app_url: Deprecated in 16, to be removed in
  19
  (Reported by Joshua C. Colp)
 * ASTERISK-29561 - app_image: Deprecated in 16, to be removed
  in 19
  (Reported by Joshua C. Colp)
 * ASTERISK-29562 - app_nbscat: Deprecated in 16, to be removed
  in 19
  (Reported by Joshua C. Colp)
 * ASTERISK-29563 - app_dahdiras: Deprecated in 16, to be
  removed in 19
  (Reported by Joshua C. Colp)
 * ASTERISK-29564 - cdr_syslog: Deprecated in 16, to be removed
  in 19
  (Reported by Joshua C. Colp)
 * ASTERISK-29565 - chan_oss: Deprecated in 16, to be removed in
  19
  (Reported by Joshua C. Colp)
 * ASTERISK-29566 - chan_phone: Deprecated in 16, to be removed
  in 19
  (Reported by Joshua C. Colp)
 * ASTERISK-29567 - chan_sip: Deprecated in 17, to be removed in
  21
  (Reported by Joshua C. Colp)
 * ASTERISK-29568 - chan_nbs: Deprecated in 16, to be removed in
  19
  (Reported by Joshua C. Colp)
 * ASTERISK-29569 - chan_misdn: Deprecated in 16, to be removed
  in 19
  (Reported by Joshua C. Colp)
 * ASTERISK-29570 - chan_vpb: Deprecated in 16, to be removed in
  19
  (Reported by Joshua C. Colp)
 * ASTERISK-29571 - res_config_sqlite: Deprecated in 16, to be
  removed in 19
  (Reported by Joshua C. Colp)
 * ASTERISK-29572 - res_monitor: Deprecated in 16, to be removed
  in 21
  (Reported by Joshua C. Colp)
 * ASTERISK-29573 - conf2ael: Deprecated in 16, to be removed in
  19
  (Reported by Joshua C. Colp)
 * ASTERISK-29574 - muted: Deprecated in 16, to be removed in
  19
  (Reported by Joshua C. Colp)

Improvements made in this release:
---
 * ASTERISK-29472 - res_pjsip: OLI/ANI2 support missing
 
  (Reported by N A)
 * ASTERISK-29626 - app_stack: Include calling location if
  attempting to branch to nonexistent location
  (Reported by
  N A)
 * ASTERISK-29632 - Add option to Application_VoiceMail to
  suppress instructions only when a custom greeting is present
   
  (Reported by Charlie Smurthwaite)
 * ASTERISK-29605 - chan_iax2: Add ANI2
  (Reported by N A)
 * ASTERISK-29508 - STUN server address refresh
  (Reported
  by SÃ©bastien Duthil)
 * ASTERISK-29612 - bridge_basic: Don't throw warning if
  attended transfer is cancelled
  (Reported by N A)
 * ASTERISK-29544 - Media Cache - Delayed remote sound file
  retrieve delays all playbacks
  (Reported by Andre Barbosa)
 * ASTERISK-29541 - app_morsecode: Add American Morse code
 
  (Reported by N A)
 * ASTERISK-29495 - Return integer instead of float if response
  is a whole number
  (Reported by N A)
 * ASTERISK-29543 - app_originate: Allow specifying codec(s) to
  use
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-29673 - app_read: Fix null pointer crash regression

  (Reported by N A)
 * ASTERISK-29660 - Build failure when disabling PJSIP support
 
  (Reported by Guido Falsi)
 * ASTERISK-29654 - pjproject includes trailing whitespace in
  sdp format attributes
  (Reported by George Joseph)
 * ASTERISK-29635 - MP3Player don' t work with actual mpg123
  versions
  (Reported by Carlos

[asterisk-users] Certified Asterisk 16.8-cert11 Now Available

2021-08-12 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Certified 
Asterisk 16.8-cert11.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/certified-asterisk

The release of Certified Asterisk 16.8-cert11 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-29235 - res_pjsip_nat: Contact is rewritten on
  REGISTER responses with external_signaling_address
 
  (Reported by Brian Paboojian)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-16.8-cert11

Thank you for your continued support of Asterisk!
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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[asterisk-users] Asterisk 18.6.0 Now Available

2021-08-12 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
18.6.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.6.0 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:

Security bugs fixed in this release:
---
 * 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-29389 - Add PJSIP_HEADERS() and ability to read
  header by pattern
  (Reported by Igor Goncharovsky)
 * ASTERISK-29477 - Function to asynchronously store digits
  dialed
  (Reported by N A)
 * ASTERISK-29454 - New application to reload modules
 
  (Reported by N A)
 * ASTERISK-29444 - Add application to wait for condition
 
  (Reported by N A)
 * ASTERISK-29442 - app_dial: Expand A option to allow
  announcement playback to caller
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-29494 - cdr_adaptive_odbc: Prevent throwing warnings
  if CDR filtering is used
  (Reported by N A)
 * ASTERISK-29513 - statsd: Remove non-standard metric type
  Meter
  (Reported by Rijnhard Hessel)
 * ASTERISK-29526 - G729 audio gets corrupted by Asterisk due to
  smoother
  (Reported by under)
 * ASTERISK-29392 - chan_iax2: Asterisk crashes when queueing
  video with format
  (Reported by Michael Welk)
 * ASTERISK-29507 - STUN timeout is silently delaying calls

  (Reported by SÃ©bastien Duthil)
 * ASTERISK-27871 - Remote URL in playback must end with file
  extension
  (Reported by Caesar)
 * ASTERISK-29514 - ari: Audiosocket segfault when no data
  specified
  (Reported by Igor Goncharovsky)
 * ASTERISK-29503 - Updated identify/match syntax not supported
  by config wizard
  (Reported by Sean Bright)
 * ASTERISK-29480 - fixedjitterbuffer contains an un-wrappered
  assert that triggers on a negative time slew
  (Reported by
  Dan Cropp)
 * ASTERISK-29485 - core: Inband generation of tones for Busy()
  and Congestion() may not occur
  (Reported by Joshua C.
  Colp)
 * ASTERISK-29479 - [patch] Channels are not put on hold for
  Session Progress with inactive audio
  (Reported by Bernd
  Zobl)

Improvements made in this release:
---
 * ASTERISK-29528 - Add support for multiple files for agent
  announcements
  (Reported by N A)
 * ASTERISK-29501 - ARI - Stasis Playback doesn't hangup call
  when processing a list of invalid files
  (Reported by Andre
  Barbosa)
 * ASTERISK-29464 - ARI - PlaybackFinish skip error events
 
  (Reported by Andre Barbosa)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0

Thank you for your continued support of Asterisk!
-- 
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[asterisk-users] Asterisk 16.20.0 Now Available

2021-08-12 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
16.20.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.20.0 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-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-29389 - Add PJSIP_HEADERS() and ability to read
  header by pattern
  (Reported by Igor Goncharovsky)
 * ASTERISK-29477 - Function to asynchronously store digits
  dialed
  (Reported by N A)
 * ASTERISK-29454 - New application to reload modules
 
  (Reported by N A)
 * ASTERISK-29444 - Add application to wait for condition
 
  (Reported by N A)
 * ASTERISK-29442 - app_dial: Expand A option to allow
  announcement playback to caller
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-29494 - cdr_adaptive_odbc: Prevent throwing warnings
  if CDR filtering is used
  (Reported by N A)
 * ASTERISK-29513 - statsd: Remove non-standard metric type
  Meter
  (Reported by Rijnhard Hessel)
 * ASTERISK-29526 - G729 audio gets corrupted by Asterisk due to
  smoother
  (Reported by under)
 * ASTERISK-29392 - chan_iax2: Asterisk crashes when queueing
  video with format
  (Reported by Michael Welk)
 * ASTERISK-29507 - STUN timeout is silently delaying calls

  (Reported by SÃ©bastien Duthil)
 * ASTERISK-27871 - Remote URL in playback must end with file
  extension
  (Reported by Caesar)
 * ASTERISK-29503 - Updated identify/match syntax not supported
  by config wizard
  (Reported by Sean Bright)
 * ASTERISK-29480 - fixedjitterbuffer contains an un-wrappered
  assert that triggers on a negative time slew
  (Reported by
  Dan Cropp)
 * ASTERISK-29485 - core: Inband generation of tones for Busy()
  and Congestion() may not occur
  (Reported by Joshua C.
  Colp)
 * ASTERISK-29479 - [patch] Channels are not put on hold for
  Session Progress with inactive audio
  (Reported by Bernd
  Zobl)

Improvements made in this release:
---
 * ASTERISK-29528 - Add support for multiple files for agent
  announcements
  (Reported by N A)
 * ASTERISK-29501 - ARI - Stasis Playback doesn't hangup call
  when processing a list of invalid files
  (Reported by Andre
  Barbosa)
 * ASTERISK-29464 - ARI - PlaybackFinish skip error events
 
  (Reported by Andre Barbosa)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0

Thank you for your continued support of Asterisk!
-- 
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New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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[asterisk-users] Asterisk 13.38.3, 16.19.1, 17.9.4, 18.5.1 and 16.8-cert10 Now Available (Security)

2021-07-22 Thread Asterisk Development Team

The Asterisk Development Team would like to announce security releases for
Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and 16.8-cert10.

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-2021-007: Remote Crash Vulnerability in PJSIP channel driver
  When Asterisk receives a re-INVITE without SDP after having sent a BYE request
  a crash will occur. This occurs due to the Asterisk channel no longer being
  present while code assumes it is.

* AST-2021-008: Remote crash when using IAX2 channel driver
  If the IAX2 channel driver receives a packet that contains an

* AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
handshake
  Depending on the timing, itâs possible for Asterisk to crash when using a
  TLS connection if the underlying socket parent/listener gets destroyed during
  the handshake.

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

https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10

The security advisories are available at:

https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
https://downloads.asterisk.org/pub/security/AST-2021-009.pdf

Thank you for your continued support of Asterisk!-- 
_
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Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
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[asterisk-users] Asterisk 18.5.0 Now Available

2021-06-24 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
18.5.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.5.0 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:

New Features made in this release:
---
 * ASTERISK-29446 - app_confbridge: New ConfKick application
   
  (Reported by N A)
 * ASTERISK-29440 - app_confbridge: Allow ConfBridge answer to
  be suppressed
  (Reported by N A)
 * ASTERISK-29431 - Minimum and maximum dialplan functions
 
  (Reported by N A)
 * ASTERISK-29439 - func_volume: Volume function can't be read
 
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-29475 - SayNumber triggers WARNING if caller hangs
  up during application execution
  (Reported by N A)
 * ASTERISK-29404 - Consolidate res_pjsip_messaging fixes for
  domain name
  (Reported by George Joseph)
 * ASTERISK-29441 - Core reload making TCP endpoints go offline

  (Reported by Luke Escude)
 * ASTERISK-28237 - "FRACK!, Failed assertion bad magic number"
  happens when unsubscribe an application from an event source
   
  (Reported by Lucas Tardioli Silveira)
 * ASTERISK-28393 - Multidomain support issue
  (Reported by
  Andrea Sannucci)
 * ASTERISK-29433 - res_rtp_asterisk: Server reflexive
  candidates use incorrect raddr for RTCP
  (Reported by
  Chris)
 * ASTERISK-29397 - pjsip: Asterisk isn't tolerant of RFC8760
  UASs
  (Reported by George Joseph)
 * ASTERISK-24601 - [patch]Missing RFC4235 tags and attributes
  in PJSIP NOTIFY event: dialog  XML body
  (Reported by Marco
  Paland)
 * ASTERISK-29370 - chan_sip does not recognize
  application/hook-flash
  (Reported by N A)
 * ASTERISK-29377 - cpool_release_pool "double free or
  corruption (out)"
  (Reported by Robert Sutton)
 * ASTERISK-29372 - file.c switch does not account for flash
  events
  (Reported by N A)
 * ASTERISK-29358 - chan_pjsip: Trace message for progress is
  output even if frame is not queued
  (Reported by Michael
  Maier)
 * ASTERISK-29407 - chan_local: Filtering audio formats should
  not occur on removed streams
  (Reported by Joshua C. Colp)
 * ASTERISK-29030 - res_rtp_asterisk: Additional RTP-frame (with
  wrong SSRC) gets inserted when switching from progress to
  established
  (Reported by Matthias Hensler)

Improvements made in this release:
---
 * ASTERISK-29450 - Allow setting channel variables using
  Originate application
  (Reported by N A)
 * ASTERISK-29459 - Missing configuration from PJSIP to SIP
  conversion script
  (Reported by N A)
 * ASTERISK-29460 - Recognize application/hook-flash in PJSIP
  
  (Reported by N A)
 * ASTERISK-29434 - Asterisk reveals pjproject version in STUN
  packets
  (Reported by Jeremy LainÃ©)
 * ASTERISK-29349 - Silent voicemail option is not completely
  silent
  (Reported by N A)
 * ASTERISK-29380 - Add Flash AMI event to handle flash events
 
  (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-18.5.0

Thank you for your continued support of Asterisk!
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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[asterisk-users] Asterisk 16.19.0 Now Available

2021-06-24 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
16.19.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.19.0 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-29446 - app_confbridge: New ConfKick application
   
  (Reported by N A)
 * ASTERISK-29440 - app_confbridge: Allow ConfBridge answer to
  be suppressed
  (Reported by N A)
 * ASTERISK-29431 - Minimum and maximum dialplan functions
 
  (Reported by N A)
 * ASTERISK-29439 - func_volume: Volume function can't be read
 
  (Reported by N A)

Bugs fixed in this release:
---
 * ASTERISK-29475 - SayNumber triggers WARNING if caller hangs
  up during application execution
  (Reported by N A)
 * ASTERISK-29404 - Consolidate res_pjsip_messaging fixes for
  domain name
  (Reported by George Joseph)
 * ASTERISK-29441 - Core reload making TCP endpoints go offline

  (Reported by Luke Escude)
 * ASTERISK-29433 - res_rtp_asterisk: Server reflexive
  candidates use incorrect raddr for RTCP
  (Reported by
  Chris)
 * ASTERISK-28237 - "FRACK!, Failed assertion bad magic number"
  happens when unsubscribe an application from an event source
   
  (Reported by Lucas Tardioli Silveira)
 * ASTERISK-28393 - Multidomain support issue
  (Reported by
  Andrea Sannucci)
 * ASTERISK-29397 - pjsip: Asterisk isn't tolerant of RFC8760
  UASs
  (Reported by George Joseph)
 * ASTERISK-24601 - [patch]Missing RFC4235 tags and attributes
  in PJSIP NOTIFY event: dialog  XML body
  (Reported by Marco
  Paland)
 * ASTERISK-29372 - file.c switch does not account for flash
  events
  (Reported by N A)
 * ASTERISK-29377 - cpool_release_pool "double free or
  corruption (out)"
  (Reported by Robert Sutton)
 * ASTERISK-29370 - chan_sip does not recognize
  application/hook-flash
  (Reported by N A)
 * ASTERISK-29358 - chan_pjsip: Trace message for progress is
  output even if frame is not queued
  (Reported by Michael
  Maier)
 * ASTERISK-29030 - res_rtp_asterisk: Additional RTP-frame (with
  wrong SSRC) gets inserted when switching from progress to
  established
  (Reported by Matthias Hensler)
 * ASTERISK-29407 - chan_local: Filtering audio formats should
  not occur on removed streams
  (Reported by Joshua C. Colp)

Improvements made in this release:
---
 * ASTERISK-29450 - Allow setting channel variables using
  Originate application
  (Reported by N A)
 * ASTERISK-29460 - Recognize application/hook-flash in PJSIP
  
  (Reported by N A)
 * ASTERISK-29459 - Missing configuration from PJSIP to SIP
  conversion script
  (Reported by N A)
 * ASTERISK-29434 - Asterisk reveals pjproject version in STUN
  packets
  (Reported by Jeremy LainÃ©)
 * ASTERISK-29349 - Silent voicemail option is not completely
  silent
  (Reported by N A)
 * ASTERISK-29380 - Add Flash AMI event to handle flash events
 
  (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-16.19.0

Thank you for your continued support of Asterisk!
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
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[asterisk-users] Asterisk 18.4.0 Now Available

2021-05-10 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of
Asterisk 18.4.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.4.0 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:

*Bugs fixed in this release:*
---

   - [ASTERISK-29328
   <https://issues.asterisk.org/jira/browse/ASTERISK-29328>] -

translate.c: possible buffer overflow when upsampling
(Reported by Jean Aunis - Prescom)

   - [ASTERISK-29379
   <https://issues.asterisk.org/jira/browse/ASTERISK-29379>] -

Segfault - ast_channel_is_multistream (chan=0x0) at
channel_internal_api.c:1590
(Reported by Ross Beer)

   - [ASTERISK-29130
   <https://issues.asterisk.org/jira/browse/ASTERISK-29130>] -

prometheus: Crash when scraping bridge
(Reported by Francisco Correia)

   - [ASTERISK-29364
   <https://issues.asterisk.org/jira/browse/ASTERISK-29364>] -

res_rtp_asterisk: standard deviation miscalculation
(Reported by Kevin Harwell)

   - [ASTERISK-29373
   <https://issues.asterisk.org/jira/browse/ASTERISK-29373>] -

res_rtp_asterisk: Flash events are duplicated
(Reported by N A)

   - [ASTERISK-28356
   <https://issues.asterisk.org/jira/browse/ASTERISK-28356>] -

app_queue: CLI set ringinuse for realtime member not working
(Reported by Michael)

   - [ASTERISK-24434
   <https://issues.asterisk.org/jira/browse/ASTERISK-24434>] -

Fix differing usage of assignment operators in modules.conf
(Reported by Rusty Newton)

   - [ASTERISK-24631
   <https://issues.asterisk.org/jira/browse/ASTERISK-24631>] -

Incorrect description of option "context" in queues.conf.sample
(Reported by Etienne Lessard)

   - [ASTERISK-26614
   <https://issues.asterisk.org/jira/browse/ASTERISK-26614>] -

app_queue: updatecdr option in queues.conf does effectively nothing
(Reported by Alexander Gonchiy)

   - [ASTERISK-25358
   <https://issues.asterisk.org/jira/browse/ASTERISK-25358>] -

dateformat not read from logger.conf by remote console
(Reported by Igor Liferenko)

   - [ASTERISK-27542
   <https://issues.asterisk.org/jira/browse/ASTERISK-27542>] -

app_queue: When "queue show" CLI command is executed a crash occurs
(Reported by Miguel Sanz)

   - [ASTERISK-29215
   <https://issues.asterisk.org/jira/browse/ASTERISK-29215>] -

res_pjsip_session: NULL active_media_state topology caused asterisk crash
(Reported by sungtae kim)

   - [ASTERISK-29355
   <https://issues.asterisk.org/jira/browse/ASTERISK-29355>] -

app_queue: Queue member status message sent even if status doesn't change
(Reported by Roman Pertsev)

   - [ASTERISK-29035
   <https://issues.asterisk.org/jira/browse/ASTERISK-29035>] -

chan_local: Multistream support breaks T.38 faxing
(Reported by Matthias Hensler)

   - [ASTERISK-29354
   <https://issues.asterisk.org/jira/browse/ASTERISK-29354>] -

res_pjsip: Allow partial reloading of transports
(Reported by Joshua C. Colp)

   - [ASTERISK-29348
   <https://issues.asterisk.org/jira/browse/ASTERISK-29348>] -

menuselect doesn't return errors in many cases
(Reported by George Joseph)

   - [ASTERISK-29352
   <https://issues.asterisk.org/jira/browse/ASTERISK-29352>] -

res_rtp_asterisk: Fix frame delivery time when SSRC changes
(Reported by Joshua C. Colp)

*Improvements made in this release:*
---

   - [ASTERISK-29339
   <https://issues.asterisk.org/jira/browse/ASTERISK-29339>] -

loader: Let's output warnings for deprecated modules!
(Reported by Joshua C. Colp)

   - [ASTERISK-29337
   <https://issues.asterisk.org/jira/browse/ASTERISK-29337>] -

menuselect: Add ability to set deprecated in and removed in versions for
modules
(Reported by Joshua C. Colp)

   - [ASTERISK-29336
   <https://issues.asterisk.org/jira/browse/ASTERISK-29336>] -

documentation: Fix inconsistent support levels
(Reported by Joshua C. Colp)

   - [ASTERISK-29335
   <https://issues.asterisk.org/jira/browse/ASTERISK-29335>] -

xml: Embed module information into core XML documentation.
(Reported by Joshua C. Colp)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0

*Thank you for your continued support of Asterisk!*
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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[asterisk-users] Asterisk 16.18.0 Now Available

2021-05-10 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of
Asterisk 16.18.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.18.0 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:

*Bugs fixed in this release:*
---

   - [ASTERISK-29328
   <https://issues.asterisk.org/jira/browse/ASTERISK-29328>] -

translate.c: possible buffer overflow when upsampling
(Reported by Jean Aunis - Prescom)

   - [ASTERISK-29379
   <https://issues.asterisk.org/jira/browse/ASTERISK-29379>] -

Segfault - ast_channel_is_multistream (chan=0x0) at
channel_internal_api.c:1590
(Reported by Ross Beer)

   - [ASTERISK-29364
   <https://issues.asterisk.org/jira/browse/ASTERISK-29364>] -

res_rtp_asterisk: standard deviation miscalculation
(Reported by Kevin Harwell)

   - [ASTERISK-29373
   <https://issues.asterisk.org/jira/browse/ASTERISK-29373>] -

res_rtp_asterisk: Flash events are duplicated
(Reported by N A)

   - [ASTERISK-28356
   <https://issues.asterisk.org/jira/browse/ASTERISK-28356>] -

app_queue: CLI set ringinuse for realtime member not working
(Reported by Michael)

   - [ASTERISK-24631
   <https://issues.asterisk.org/jira/browse/ASTERISK-24631>] -

Incorrect description of option "context" in queues.conf.sample
(Reported by Etienne Lessard)

   - [ASTERISK-26614
   <https://issues.asterisk.org/jira/browse/ASTERISK-26614>] -

app_queue: updatecdr option in queues.conf does effectively nothing
(Reported by Alexander Gonchiy)

   - [ASTERISK-25358
   <https://issues.asterisk.org/jira/browse/ASTERISK-25358>] -

dateformat not read from logger.conf by remote console
(Reported by Igor Liferenko)

   - [ASTERISK-27542
   <https://issues.asterisk.org/jira/browse/ASTERISK-27542>] -

app_queue: When "queue show" CLI command is executed a crash occurs
(Reported by Miguel Sanz)

   - [ASTERISK-29215
   <https://issues.asterisk.org/jira/browse/ASTERISK-29215>] -

res_pjsip_session: NULL active_media_state topology caused asterisk crash
(Reported by sungtae kim)

   - [ASTERISK-29355
   <https://issues.asterisk.org/jira/browse/ASTERISK-29355>] -

app_queue: Queue member status message sent even if status doesn't change
(Reported by Roman Pertsev)

   - [ASTERISK-29035
   <https://issues.asterisk.org/jira/browse/ASTERISK-29035>] -

chan_local: Multistream support breaks T.38 faxing
(Reported by Matthias Hensler)

   - [ASTERISK-29354
   <https://issues.asterisk.org/jira/browse/ASTERISK-29354>] -

res_pjsip: Allow partial reloading of transports
(Reported by Joshua C. Colp)

   - [ASTERISK-29348
   <https://issues.asterisk.org/jira/browse/ASTERISK-29348>] -

menuselect doesn't return errors in many cases
(Reported by George Joseph)

   - [ASTERISK-29352
   <https://issues.asterisk.org/jira/browse/ASTERISK-29352>] -

res_rtp_asterisk: Fix frame delivery time when SSRC changes
(Reported by Joshua C. Colp)

*Improvements made in this release:*
---

   - [ASTERISK-29339
   <https://issues.asterisk.org/jira/browse/ASTERISK-29339>] -

loader: Let's output warnings for deprecated modules!
(Reported by Joshua C. Colp)

   - [ASTERISK-29337
   <https://issues.asterisk.org/jira/browse/ASTERISK-29337>] -

menuselect: Add ability to set deprecated in and removed in versions for
modules
(Reported by Joshua C. Colp)

   - [ASTERISK-29335
   <https://issues.asterisk.org/jira/browse/ASTERISK-29335>] -

xml: Embed module information into core XML documentation.
(Reported by Joshua C. Colp)

   - [ASTERISK-29336
   <https://issues.asterisk.org/jira/browse/ASTERISK-29336>] -

documentation: Fix inconsistent support levels
(Reported by Joshua C. Colp)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0

*Thank you for your continued support of Asterisk!*
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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To UNSUBSCRIBE or update options visit:
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[asterisk-users] Asterisk 18.3.0 Now Available

2021-03-25 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
18.3.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.3.0 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-29305 - ASTERISK-29203 / AST-2021-002 -- Another
  scenario is causing a crash
  (Reported by Gregory Massel)
 * ASTERISK-29260 - sRTP Replay Protection ignored; even tears
  down long calls
  (Reported by Alexander Traud)
 * ASTERISK-29227 - res_pjsip_diversion: sending multiple 181
  responses causes memory corruption and crash
  (Reported by
  Ivan Poddubny)

Bugs fixed in this release:
---
 * ASTERISK-29215 - res_pjsip_session: NULL active_media_state
  topology caused asterisk crash
  (Reported by sungtae kim)
 * ASTERISK-29035 - chan_local: Multistream support breaks T.38
  faxing
  (Reported by Matthias Hensler)
 * ASTERISK-29071 - app_confbridge: Memory rises when
  jitterbuffer enabled and muting over AMI occurs
  (Reported
  by Stefan Ruf)
 * ASTERISK-29329 - app_dial: DTMF to 'D' option gets duplicated
  if there are multiple progress events
  (Reported by N A)
 * ASTERISK-24434 - Fix differing usage of assignment operators
  in modules.conf
  (Reported by Rusty Newton)
 * ASTERISK-29306 - strings: Incorrect use of
  __attribute__((pure)) in ast_str_to_lower definition
 
  (Reported by Vitezslav Novy)
 * ASTERISK-29300 - res_rtp_asterisk: When native local bridging
  the remote SSRC becomes permanent
  (Reported by Sebastian
  Damm)
 * ASTERISK-29235 - res_pjsip_nat: Contact is rewritten on
  REGISTER responses with external_signaling_address
 
  (Reported by Brian Paboojian)
 * ASTERISK-29266 - ICE Role conflict with an unauthorized
  session
  (Reported by Salah Ahmed)
 * ASTERISK-29105 - chan_pjsip: 180 Ringing with SDP not changed
  into progress
  (Reported by Sebastian Damm)
 * ASTERISK-29297 - say: Y2021 problem â Asterisk cannot say
  year 2021 in Dutch
  (Reported by Jacek Konieczny)
 * ASTERISK-29315 - res_pjsip: re-registration gets stuck if
  setting initial auth credentials fails
  (Reported by Nick
  French)
 * ASTERISK-29312 - res_fax: asterisk fails to publish the
  Stasis and ReceiveFax status messages if the remote Station ID
  contains invalid UTF-8 characters
  (Reported by Alexei
  Gradinari)
 * ASTERISK-16799 - Callee declined when 'beep' audio file does
  not exist
  (Reported by IAMJames_)
 * ASTERISK-29313 - res_pjsip_refer:  Segfault in progress
  notify
  (Reported by George Joseph)
 * ASTERISK-29293 - res_config_pgsql: Limit realtime_pgsql() to
  return one (no more) record
  (Reported by Boris P. Korzun)
 * ASTERISK-29303 - pjsip: Re-invite occurs when it shouldn't
  
  (Reported by Benjamin Keith Ford)
 * ASTERISK-29311 - res_odbc_transaction sets forcecommit
  default value based on isolation level instead of forcecommit
  
  (Reported by Jaco Kroon)
 * ASTERISK-28452 - pjsip:  of SDP is not
  incremented though SDP may be changed on reinvite without SDP
  offer
  (Reported by Michael Maier)
 * ASTERISK-29287 - app.h: C++ compatibility broken
 
  (Reported by Jean Aunis - Prescom)
 * ASTERISK-28369 - app_queue: Member device state "invalid"
  when second call is ringing and hint is used
  (Reported by
  Boolah )
 * ASTERISK-29203 - res_pjsip_t38: Crash when changing state
   
  (Reported by Gregory Massel)
 * ASTERISK-29205 - res_rtp_asterisk: Asterisk crashes when
  making hold/unhold from webrtc client
  (Reported by Edvin
  Vidmar)
 * ASTERISK-29196 - res_pjsip: Segmentation fault
 
  (Reported by Mauri de Souza Meneguzzo (3CPlus))
 * ASTERISK-29280 - chan_sip: Allow peers without audio
  (text+video).
  (Reported by Alexander Traud)
 * ASTERISK-29265 - chan_sip: Allow text+video media streams,
  again.
  (Reported by Alexander Traud)
 * ASTERISK-29261 - res_pjsip: user=phone validation fail for
  isup numbers containing *#
  (Reported by Mark Petersen)
 * ASTERISK-29259 - channel: Allow text+video media streams,
  again.
  (Reported by Alexander Traud)
 * ASTERISK-29258 - chan_sip: Audio stream rejected, Other
  stream present: Invalid SDP.
  (Reported by Alexander Traud)
 * ASTERISK-29220 - After T38 reinvite response of 488 a
  subsequent G711 reinvite is not processed correctly. Instead the
  previous T38 session media is used
  (Reported by Robert
  Cripps)
 * ASTERISK-29248 - res_pjsip_session: res sometimes
  uninitialize

[asterisk-users] Asterisk 16.17.0 Now Available

2021-03-25 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
16.17.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.17.0 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:

Security bugs fixed in this release:
---
 * ASTERISK-29305 - ASTERISK-29203 / AST-2021-002 -- Another
  scenario is causing a crash
  (Reported by Gregory Massel)
 * ASTERISK-29260 - sRTP Replay Protection ignored; even tears
  down long calls
  (Reported by Alexander Traud)
 * ASTERISK-29227 - res_pjsip_diversion: sending multiple 181
  responses causes memory corruption and crash
  (Reported by
  Ivan Poddubny)

Bugs fixed in this release:
---
 * ASTERISK-29215 - res_pjsip_session: NULL active_media_state
  topology caused asterisk crash
  (Reported by sungtae kim)
 * ASTERISK-29035 - chan_local: Multistream support breaks T.38
  faxing
  (Reported by Matthias Hensler)
 * ASTERISK-29071 - app_confbridge: Memory rises when
  jitterbuffer enabled and muting over AMI occurs
  (Reported
  by Stefan Ruf)
 * ASTERISK-29329 - app_dial: DTMF to 'D' option gets duplicated
  if there are multiple progress events
  (Reported by N A)
 * ASTERISK-24434 - Fix differing usage of assignment operators
  in modules.conf
  (Reported by Rusty Newton)
 * ASTERISK-29306 - strings: Incorrect use of
  __attribute__((pure)) in ast_str_to_lower definition
 
  (Reported by Vitezslav Novy)
 * ASTERISK-29300 - res_rtp_asterisk: When native local bridging
  the remote SSRC becomes permanent
  (Reported by Sebastian
  Damm)
 * ASTERISK-29235 - res_pjsip_nat: Contact is rewritten on
  REGISTER responses with external_signaling_address
 
  (Reported by Brian Paboojian)
 * ASTERISK-29266 - ICE Role conflict with an unauthorized
  session
  (Reported by Salah Ahmed)
 * ASTERISK-29105 - chan_pjsip: 180 Ringing with SDP not changed
  into progress
  (Reported by Sebastian Damm)
 * ASTERISK-29297 - say: Y2021 problem â Asterisk cannot say
  year 2021 in Dutch
  (Reported by Jacek Konieczny)
 * ASTERISK-29312 - res_fax: asterisk fails to publish the
  Stasis and ReceiveFax status messages if the remote Station ID
  contains invalid UTF-8 characters
  (Reported by Alexei
  Gradinari)
 * ASTERISK-16799 - Callee declined when 'beep' audio file does
  not exist
  (Reported by IAMJames_)
 * ASTERISK-29313 - res_pjsip_refer:  Segfault in progress
  notify
  (Reported by George Joseph)
 * ASTERISK-28452 - pjsip:  of SDP is not
  incremented though SDP may be changed on reinvite without SDP
  offer
  (Reported by Michael Maier)
 * ASTERISK-29303 - pjsip: Re-invite occurs when it shouldn't
  
  (Reported by Benjamin Keith Ford)
 * ASTERISK-29293 - res_config_pgsql: Limit realtime_pgsql() to
  return one (no more) record
  (Reported by Boris P. Korzun)
 * ASTERISK-29311 - res_odbc_transaction sets forcecommit
  default value based on isolation level instead of forcecommit
  
  (Reported by Jaco Kroon)
 * ASTERISK-28369 - app_queue: Member device state "invalid"
  when second call is ringing and hint is used
  (Reported by
  Boolah )
 * ASTERISK-29287 - app.h: C++ compatibility broken
 
  (Reported by Jean Aunis - Prescom)
 * ASTERISK-29203 - res_pjsip_t38: Crash when changing state
   
  (Reported by Gregory Massel)
 * ASTERISK-29205 - res_rtp_asterisk: Asterisk crashes when
  making hold/unhold from webrtc client
  (Reported by Edvin
  Vidmar)
 * ASTERISK-29196 - res_pjsip: Segmentation fault
 
  (Reported by Mauri de Souza Meneguzzo (3CPlus))
 * ASTERISK-29280 - chan_sip: Allow peers without audio
  (text+video).
  (Reported by Alexander Traud)
 * ASTERISK-29265 - chan_sip: Allow text+video media streams,
  again.
  (Reported by Alexander Traud)
 * ASTERISK-29259 - channel: Allow text+video media streams,
  again.
  (Reported by Alexander Traud)
 * ASTERISK-29261 - res_pjsip: user=phone validation fail for
  isup numbers containing *#
  (Reported by Mark Petersen)
 * ASTERISK-29258 - chan_sip: Audio stream rejected, Other
  stream present: Invalid SDP.
  (Reported by Alexander Traud)
 * ASTERISK-29220 - After T38 reinvite response of 488 a
  subsequent G711 reinvite is not processed correctly. Instead the
  previous T38 session media is used
  (Reported by Robert
  Cripps)
 * ASTERISK-29248 - res_pjsip_session: res sometimes
  uninitialized reported by compiler Clang.
  (Reported by
  Alexander Traud)

Improvements made in this release:
---

[asterisk-users] Asterisk 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7 Now Available (Security)

2021-03-04 Thread Asterisk Development Team

The Asterisk Development Team would like to announce security releases for
Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.

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-2021-006: Crash when negotiating T.38 with a zero port
  When Asterisk sends a re-invite initiating T.38 faxing and the endpoint
  responds with a m=image line and zero port, a crash will occur in Asterisk.
  This is a reoccurrence of AST-2019-004.

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

https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7

The security advisory is available at:

https://downloads.asterisk.org/pub/security/AST-2021-006.pdf

Thank you for your continued support of Asterisk!-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] Asterisk 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6 Now Available (Security)

2021-02-18 Thread Asterisk Development Team

The Asterisk Development Team would like to announce security releases for
Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.

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-2021-001: Remote crash in res_pjsip_diversion
  If a registered user is tricked into dialing a

* AST-2021-002: Remote crash possible when negotiating T.38
  When

* AST-2021-003: Remote attacker could prematurely tear down SRTP calls
  An unauthenticated remote attacker could replay SRTP packets which could cause
  an Asterisk instance configured without strict RTP validation to tear down
  calls prematurely.

* AST-2021-004: An unsuspecting user could crash Asterisk with multiple
hold/unhold requests
  Due to a signedness comparison mismatch, an authenticated WebRTC client could
  cause a stack overflow and Asterisk crash by sending multiple hold/unhold
  requests in quick succession.

* AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver
  Given a scenario where an outgoing call is placed from Asterisk to a remote
  SIP server it is possible for a crash to occur.

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

https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6

The security advisories are available at:

https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
https://downloads.asterisk.org/pub/security/AST-2021-005.pdf

Thank you for your continued support of Asterisk!-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] Asterisk 18.2.0 Now Available

2021-01-21 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
18.2.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.2.0 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:

Security bugs fixed in this release:
---
 * ASTERISK-29219 - res_pjsip_diversion: Crash if Tel URI
  contains History-Info
  (Reported by Torrey Searle)

Bugs fixed in this release:
---
 * ASTERISK-29229 - Stasis/messaging: text messages not
  dispatched to all subscribers when using generic subscription
  
  (Reported by Jean Aunis - Prescom)
 * ASTERISK-29240 - chan_pjsip: Incoming PJSIP calls set global
  SIPDOMAIN instead of a channel variable
  (Reported by Ivan
  Poddubny)
 * ASTERISK-29238 - chan_sip: SDP: Offers without any enabled
  stream are accepted.
  (Reported by Alexander Traud)
 * ASTERISK-29237 - chan_sip: SDP: m=video is parsed even when
  disabled.
  (Reported by Alexander Traud)
 * ASTERISK-29222 - chan_sip: Hold/Resume an sRTP call on a
  video enabled user-agent.
  (Reported by Alexander Traud)
 * ASTERISK-27902 - chan_pjsip isn't updating hangupcause on 4XX
  responses
  (Reported by George Joseph)
 * ASTERISK-28016 - PJSIP sends duplicate 183 Progress
  responses
  (Reported by Alex Hermann)
 * ASTERISK-28185 - chan_pjsip: Subsequent same responses are
  not stopped
  (Reported by Julien)
 * ASTERISK-29230 - pjsip: Asterisk goes crazy and massively
  spams logfile if registration can't be send
  (Reported by
  Michael Maier)
 * ASTERISK-29231 - pjsip: SIGSEGV in CLI if no trunk is
  registered
  (Reported by Michael Maier)
 * ASTERISK-29217 - LOCK() can grant the same lock to multiple
  channels spuriously
  (Reported by Jaco Kroon)
 * ASTERISK-29201 - Crash occurs when Transfer and execute
  Hangup before the Transfer result 
  (Reported by Dan Cropp)
 * ASTERISK-28947 - Segmentation fault in mixmonitor_ds_destroy

  (Reported by Robert Sutton)
 * ASTERISK-29168 - Asterisk crashes during call transfer
 
  (Reported by Dalius Mockevicius)
 * ASTERISK-29210 - res_pjsip: Crash when examining transport
  
  (Reported by N GM )
 * ASTERISK-29191 - tel: URI in Diversion header causes crash
  
  (Reported by Mikhail Ivanov)
 * ASTERISK-28883 - Spyee information ist missing in ChanSpyStop
  AMI Event
  (Reported by Hendrik Wedhorn)
 * ASTERISK-29188 - null media causing the Asterisk crash
 
  (Reported by sungtae kim)
 * ASTERISK-29024 - pjsip: Route Header in Cancel request
  incorrectly set
  (Reported by Flole Systems)
 * ASTERISK-29209 - Debug messages printed by scope trace might
  be missing newlines
  (Reported by Alexander Traud)
 * ASTERISK-29211 - res_musiconhold: Segfault on realtime music
  on hold without entries
  (Reported by Nathan Bruning)
 * ASTERISK-29022 - Crash when manipulating PJSIP invite dlg ref
  counts
  (Reported by Sean Bright)
 * ASTERISK-29173 - Media cache URL requests allow infinite
  redirects
  (Reported by Sean Bright)
 * ASTERISK-29175 - res_pjsip_stir_shaken: Fix module
  description
  (Reported by Stanislav Abramenkov)
 * ASTERISK-29148 - AST_MODULE_INFO no, MODULEINFO depend
 
  (Reported by Alexander Traud)
 * ASTERISK-29165 - res_pjsip: malformed header Accept-Encoding
  in OPTIONS response
  (Reported by Alexander Greiner-Baer)
 * ASTERISK-28798 - [patch] chan_sip: TCP/TLS client without
  server.
  (Reported by Alexander Traud)
 * ASTERISK-29161 - Incorrect setup of recall channels
 
  (Reported by Boris P. Korzun)
 * ASTERISK-29155 - app_queue: Deadlock between queues container
  and individual queues
  (Reported by George Joseph)

Improvements made in this release:
---
 * ASTERISK-28549 - Two repeated 183
  (Reported by Gant
  Liu)
 * ASTERISK-29216 - contrib: systemd asterisk service for
  centos8 or other newer linux versions
  (Reported by Mark
  Petersen)
 * ASTERISK-29143 - res_http_media_cache: HTTP media cache
  stored hardcoded in /tmp
  (Reported by laszlovl)
 * ASTERISK-29118 - VoiceMail() should have an option to play
  greetings as Early Media
  (Reported by Juan Carlos Castro y
  Castro)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0

Thank you for your continued support of Asterisk!
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https

[asterisk-users] Asterisk 16.16.0 Now Available

2021-01-21 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
16.16.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.16.0 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-29219 - res_pjsip_diversion: Crash if Tel URI
  contains History-Info
  (Reported by Torrey Searle)

Bugs fixed in this release:
---
 * ASTERISK-29229 - Stasis/messaging: text messages not
  dispatched to all subscribers when using generic subscription
  
  (Reported by Jean Aunis - Prescom)
 * ASTERISK-29238 - chan_sip: SDP: Offers without any enabled
  stream are accepted.
  (Reported by Alexander Traud)
 * ASTERISK-29237 - chan_sip: SDP: m=video is parsed even when
  disabled.
  (Reported by Alexander Traud)
 * ASTERISK-29222 - chan_sip: Hold/Resume an sRTP call on a
  video enabled user-agent.
  (Reported by Alexander Traud)
 * ASTERISK-29240 - chan_pjsip: Incoming PJSIP calls set global
  SIPDOMAIN instead of a channel variable
  (Reported by Ivan
  Poddubny)
 * ASTERISK-27902 - chan_pjsip isn't updating hangupcause on 4XX
  responses
  (Reported by George Joseph)
 * ASTERISK-28016 - PJSIP sends duplicate 183 Progress
  responses
  (Reported by Alex Hermann)
 * ASTERISK-28185 - chan_pjsip: Subsequent same responses are
  not stopped
  (Reported by Julien)
 * ASTERISK-29230 - pjsip: Asterisk goes crazy and massively
  spams logfile if registration can't be send
  (Reported by
  Michael Maier)
 * ASTERISK-29231 - pjsip: SIGSEGV in CLI if no trunk is
  registered
  (Reported by Michael Maier)
 * ASTERISK-29217 - LOCK() can grant the same lock to multiple
  channels spuriously
  (Reported by Jaco Kroon)
 * ASTERISK-29201 - Crash occurs when Transfer and execute
  Hangup before the Transfer result 
  (Reported by Dan Cropp)
 * ASTERISK-28947 - Segmentation fault in mixmonitor_ds_destroy

  (Reported by Robert Sutton)
 * ASTERISK-29191 - tel: URI in Diversion header causes crash
  
  (Reported by Mikhail Ivanov)
 * ASTERISK-28883 - Spyee information ist missing in ChanSpyStop
  AMI Event
  (Reported by Hendrik Wedhorn)
 * ASTERISK-29188 - null media causing the Asterisk crash
 
  (Reported by sungtae kim)
 * ASTERISK-29209 - Debug messages printed by scope trace might
  be missing newlines
  (Reported by Alexander Traud)
 * ASTERISK-29024 - pjsip: Route Header in Cancel request
  incorrectly set
  (Reported by Flole Systems)
 * ASTERISK-29211 - res_musiconhold: Segfault on realtime music
  on hold without entries
  (Reported by Nathan Bruning)
 * ASTERISK-29022 - Crash when manipulating PJSIP invite dlg ref
  counts
  (Reported by Sean Bright)
 * ASTERISK-29173 - Media cache URL requests allow infinite
  redirects
  (Reported by Sean Bright)
 * ASTERISK-29175 - res_pjsip_stir_shaken: Fix module
  description
  (Reported by Stanislav Abramenkov)
 * ASTERISK-29148 - AST_MODULE_INFO no, MODULEINFO depend
 
  (Reported by Alexander Traud)
 * ASTERISK-28798 - [patch] chan_sip: TCP/TLS client without
  server.
  (Reported by Alexander Traud)
 * ASTERISK-29165 - res_pjsip: malformed header Accept-Encoding
  in OPTIONS response
  (Reported by Alexander Greiner-Baer)
 * ASTERISK-29161 - Incorrect setup of recall channels
 
  (Reported by Boris P. Korzun)
 * ASTERISK-29155 - app_queue: Deadlock between queues container
  and individual queues
  (Reported by George Joseph)

Improvements made in this release:
---
 * ASTERISK-28549 - Two repeated 183
  (Reported by Gant
  Liu)
 * ASTERISK-29216 - contrib: systemd asterisk service for
  centos8 or other newer linux versions
  (Reported by Mark
  Petersen)
 * ASTERISK-29143 - res_http_media_cache: HTTP media cache
  stored hardcoded in /tmp
  (Reported by laszlovl)
 * ASTERISK-29118 - VoiceMail() should have an option to play
  greetings as Early Media
  (Reported by Juan Carlos Castro y
  Castro)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0

Thank you for your continued support of Asterisk!
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit

[asterisk-users] Asterisk 13.38.1, 16.15.1, 17.9.1 and 18.1.1 Now Available (Security)

2020-12-22 Thread Asterisk Development Team

The Asterisk Development Team would like to announce security releases for
Asterisk 13, 16, 17 and 18. The available releases are released as versions
13.38.1, 16.15.1, 17.9.1 and 18.1.1.

These releases are available for immediate download at

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

The following security vulnerabilities were resolved in these versions:

* AST-2020-003: Remote crash in res_pjsip_diversion
  A crash can occur in Asterisk when a SIP message is received that has a
  History-Info header, which contains a tel-uri.

* AST-2020-004: Remote crash in res_pjsip_diversion
  A crash can occur in Asterisk when a SIP 181 response is received that has a
  Diversion header, which contains a tel-uri.

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

https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1

The security advisories are available at:

https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
https://downloads.asterisk.org/pub/security/AST-2020-004.pdf

Thank you for your continued support of Asterisk!-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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[asterisk-users] Asterisk 18.1.0 Now Available

2020-11-19 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
18.1.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.1.0 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-29057 - pjsip: Crash on call rejection during high
  load
  (Reported by Sandro Gauci)

New Features made in this release:
---
 * ASTERISK-29027 - Implement support for History-Info
 
  (Reported by Torrey Searle)

Bugs fixed in this release:
---
 * ASTERISK-28933 - res_pjsip.so fails to load when bundled
  pjproject is compiled without libssl
  (Reported by Walter
  Doekes)
 * ASTERISK-28825 - Any curl response checks out as valid even
  if 404 is returned.
  (Reported by dovid)
 * ASTERISK-29013 - res_pjsip: Asterisk doesn't stop sending
  invites (with auth) on 407 replies
  (Reported by Sebastian
  Damm)
 * ASTERISK-29142 - sip_to_pjsip.py: doesn't read globbed
  includes
  (Reported by Michael Newton)
 * ASTERISK-29144 - GCC Warnings with OPTIMIZE=-Og make
 
  (Reported by Alexander Traud)
 * ASTERISK-29145 - GCC Warnings with OPTIMIZE=-Os make
 
  (Reported by Alexander Traud)
 * ASTERISK-29146 - GCC Warnings: â%sâ directive argument is
  null.
  (Reported by Alexander Traud)
 * ASTERISK-29124 - res_pjsip: flow transport broken for
  outbound requests
  (Reported by Nick French)
 * ASTERISK-29136 - config: Sample features.conf incorrectly
  includes " around sound files
  (Reported by Benjamin M.)
 * ASTERISK-29123 - logger.conf.sample missing comment mark on
  line 115
  (Reported by Andrew Siplas)
 * ASTERISK-29109 - res_pjsip_session: Asterisk 18 does not
  progress calls due to codec negotiation after upgrading from
  Asterisk 16
  (Reported by Ross Beer)
 * ASTERISK-28430 - res_rtp_asterisk.c: FRACK!, Failed assertion
  errno != EBADF
  (Reported by under)
 * ASTERISK-29108 - resource_endpoints.c : Memory leak if
  endpoint not found
  (Reported by Jean Aunis - Prescom)
 * ASTERISK-26424 - app_voicemail: Undocumented behavior from
  VMSayName
  (Reported by Eric Smith)
 * ASTERISK-29097 - res_pjsip_config_wizard: Crash when freeing
  string when failing to add extension
  (Reported by Vieri)
 * ASTERISK-29091 - Crash when ast_translator_build_path fails
 
  (Reported by Jasper van der Neut)
 * ASTERISK-29051 - res_pjsip_sdp_rtp: Does not set correct
  values on RTP instance when "auto" DTMF is used
  (Reported
  by Sebastian Damm)
 * ASTERISK-29099 - res_musiconhold: Realtime MOH only loads a
  single entry
  (Reported by laszlovl)
 * ASTERISK-28311 - dsp: ast_dsp_silence_noise_with_energy wrong
  judgment of frame format
  (Reported by å¨å®¶å»º)
 * ASTERISK-24329 - Music On Hold announcement cuts intro of
  music the first time it is played
  (Reported by Thomas
  Frederiksen)
 * ASTERISK-29085 - func_curl: Segmentation fault when using
  CURL after setting httpheader CURLOPT
  (Reported by PÃ©ter
  JuhÃ¡sz)
 * ASTERISK-29089 - RTP Ports not cleared after hangup
 
  (Reported by Ross Beer)
 * ASTERISK-29081 - res_stasis: Add compare function for bridges
  moh container
  (Reported by Hajek Michal)
 * ASTERISK-28416 - Unable to get rtp codec payload code for
  slin
  (Reported by Brian J. Murrell)
 * ASTERISK-29014 - res_pjsip_session: Re-INVITE collisions
  aren't handled correctly
  (Reported by George Joseph)

Improvements made in this release:
---
 * ASTERISK-29054 - Logger: Add debug logging categories
 
  (Reported by Kevin Harwell)
 * ASTERISK-29056 - Increase reg_server column size for
  ps_contacts table realtime
  (Reported by sungtae kim)
 * ASTERISK-29055 - Create a Bridge with video_single mode
 
  (Reported by sungtae kim)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0

Thank you for your continued support of Asterisk!
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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To UNSUBSCRIBE or update options visit:
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[asterisk-users] Asterisk 17.9.0 Now Available

2020-11-19 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
17.9.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 17.9.0 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-29057 - pjsip: Crash on call rejection during high
  load
  (Reported by Sandro Gauci)

Improvements made in this release:
---
 * ASTERISK-29055 - Create a Bridge with video_single mode
 
  (Reported by sungtae kim)
 * ASTERISK-29056 - Increase reg_server column size for
  ps_contacts table realtime
  (Reported by sungtae kim)

Bugs fixed in this release:
---
 * ASTERISK-29013 - res_pjsip: Asterisk doesn't stop sending
  invites (with auth) on 407 replies
  (Reported by Sebastian
  Damm)
 * ASTERISK-29124 - res_pjsip: flow transport broken for
  outbound requests
  (Reported by Nick French)
 * ASTERISK-29123 - logger.conf.sample missing comment mark on
  line 115
  (Reported by Andrew Siplas)
 * ASTERISK-28430 - res_rtp_asterisk.c: FRACK!, Failed assertion
  errno != EBADF
  (Reported by under)
 * ASTERISK-29108 - resource_endpoints.c : Memory leak if
  endpoint not found
  (Reported by Jean Aunis - Prescom)
 * ASTERISK-29097 - res_pjsip_config_wizard: Crash when freeing
  string when failing to add extension
  (Reported by Vieri)
 * ASTERISK-26424 - app_voicemail: Undocumented behavior from
  VMSayName
  (Reported by Eric Smith)
 * ASTERISK-29091 - Crash when ast_translator_build_path fails
 
  (Reported by Jasper van der Neut)
 * ASTERISK-29099 - res_musiconhold: Realtime MOH only loads a
  single entry
  (Reported by laszlovl)
 * ASTERISK-29051 - res_pjsip_sdp_rtp: Does not set correct
  values on RTP instance when "auto" DTMF is used
  (Reported
  by Sebastian Damm)
 * ASTERISK-28311 - dsp: ast_dsp_silence_noise_with_energy wrong
  judgment of frame format
  (Reported by å¨å®¶å»º)
 * ASTERISK-29085 - func_curl: Segmentation fault when using
  CURL after setting httpheader CURLOPT
  (Reported by PÃ©ter
  JuhÃ¡sz)
 * ASTERISK-24329 - Music On Hold announcement cuts intro of
  music the first time it is played
  (Reported by Thomas
  Frederiksen)
 * ASTERISK-29089 - RTP Ports not cleared after hangup
 
  (Reported by Ross Beer)
 * ASTERISK-29081 - res_stasis: Add compare function for bridges
  moh container
  (Reported by Hajek Michal)
 * ASTERISK-28416 - Unable to get rtp codec payload code for
  slin
  (Reported by Brian J. Murrell)
 * ASTERISK-29014 - res_pjsip_session: Re-INVITE collisions
  aren't handled correctly
  (Reported by George Joseph)

New Features made in this release:
---
 * ASTERISK-29027 - Implement support for History-Info
 
  (Reported by Torrey Searle)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0

Thank you for your continued support of Asterisk!
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] Asterisk 16.15.0 Now Available

2020-11-19 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
16.15.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.15.0 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:

Security bugs fixed in this release:
---
 * ASTERISK-29057 - pjsip: Crash on call rejection during high
  load
  (Reported by Sandro Gauci)

New Features made in this release:
---
 * ASTERISK-29027 - Implement support for History-Info
 
  (Reported by Torrey Searle)

Bugs fixed in this release:
---
 * ASTERISK-28933 - res_pjsip.so fails to load when bundled
  pjproject is compiled without libssl
  (Reported by Walter
  Doekes)
 * ASTERISK-28825 - Any curl response checks out as valid even
  if 404 is returned.
  (Reported by dovid)
 * ASTERISK-29013 - res_pjsip: Asterisk doesn't stop sending
  invites (with auth) on 407 replies
  (Reported by Sebastian
  Damm)
 * ASTERISK-29142 - sip_to_pjsip.py: doesn't read globbed
  includes
  (Reported by Michael Newton)
 * ASTERISK-29144 - GCC Warnings with OPTIMIZE=-Og make
 
  (Reported by Alexander Traud)
 * ASTERISK-29146 - GCC Warnings: â%sâ directive argument is
  null.
  (Reported by Alexander Traud)
 * ASTERISK-29145 - GCC Warnings with OPTIMIZE=-Os make
 
  (Reported by Alexander Traud)
 * ASTERISK-29136 - config: Sample features.conf incorrectly
  includes " around sound files
  (Reported by Benjamin M.)
 * ASTERISK-29123 - logger.conf.sample missing comment mark on
  line 115
  (Reported by Andrew Siplas)
 * ASTERISK-29108 - resource_endpoints.c : Memory leak if
  endpoint not found
  (Reported by Jean Aunis - Prescom)
 * ASTERISK-28430 - res_rtp_asterisk.c: FRACK!, Failed assertion
  errno != EBADF
  (Reported by under)
 * ASTERISK-29097 - res_pjsip_config_wizard: Crash when freeing
  string when failing to add extension
  (Reported by Vieri)
 * ASTERISK-26424 - app_voicemail: Undocumented behavior from
  VMSayName
  (Reported by Eric Smith)
 * ASTERISK-29099 - res_musiconhold: Realtime MOH only loads a
  single entry
  (Reported by laszlovl)
 * ASTERISK-29051 - res_pjsip_sdp_rtp: Does not set correct
  values on RTP instance when "auto" DTMF is used
  (Reported
  by Sebastian Damm)
 * ASTERISK-29091 - Crash when ast_translator_build_path fails
 
  (Reported by Jasper van der Neut)
 * ASTERISK-28311 - dsp: ast_dsp_silence_noise_with_energy wrong
  judgment of frame format
  (Reported by å¨å®¶å»º)
 * ASTERISK-29085 - func_curl: Segmentation fault when using
  CURL after setting httpheader CURLOPT
  (Reported by PÃ©ter
  JuhÃ¡sz)
 * ASTERISK-24329 - Music On Hold announcement cuts intro of
  music the first time it is played
  (Reported by Thomas
  Frederiksen)
 * ASTERISK-29089 - RTP Ports not cleared after hangup
 
  (Reported by Ross Beer)
 * ASTERISK-29081 - res_stasis: Add compare function for bridges
  moh container
  (Reported by Hajek Michal)
 * ASTERISK-28416 - Unable to get rtp codec payload code for
  slin
  (Reported by Brian J. Murrell)
 * ASTERISK-29014 - res_pjsip_session: Re-INVITE collisions
  aren't handled correctly
  (Reported by George Joseph)

Improvements made in this release:
---
 * ASTERISK-29054 - Logger: Add debug logging categories
 
  (Reported by Kevin Harwell)
 * ASTERISK-29055 - Create a Bridge with video_single mode
 
  (Reported by sungtae kim)
 * ASTERISK-29056 - Increase reg_server column size for
  ps_contacts table realtime
  (Reported by sungtae kim)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0

Thank you for your continued support of Asterisk!
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] Asterisk 13.38.0 Now Available

2020-11-19 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
13.38.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.38.0 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-29057 - pjsip: Crash on call rejection during high
  load
  (Reported by Sandro Gauci)

Improvements made in this release:
---
 * ASTERISK-29056 - Increase reg_server column size for
  ps_contacts table realtime
  (Reported by sungtae kim)
 * ASTERISK-29055 - Create a Bridge with video_single mode
 
  (Reported by sungtae kim)

Bugs fixed in this release:
---
 * ASTERISK-29013 - res_pjsip: Asterisk doesn't stop sending
  invites (with auth) on 407 replies
  (Reported by Sebastian
  Damm)
 * ASTERISK-29108 - resource_endpoints.c : Memory leak if
  endpoint not found
  (Reported by Jean Aunis - Prescom)
 * ASTERISK-29097 - res_pjsip_config_wizard: Crash when freeing
  string when failing to add extension
  (Reported by Vieri)
 * ASTERISK-26424 - app_voicemail: Undocumented behavior from
  VMSayName
  (Reported by Eric Smith)
 * ASTERISK-29051 - res_pjsip_sdp_rtp: Does not set correct
  values on RTP instance when "auto" DTMF is used
  (Reported
  by Sebastian Damm)
 * ASTERISK-28311 - dsp: ast_dsp_silence_noise_with_energy wrong
  judgment of frame format
  (Reported by å¨å®¶å»º)
 * ASTERISK-24329 - Music On Hold announcement cuts intro of
  music the first time it is played
  (Reported by Thomas
  Frederiksen)
 * ASTERISK-29081 - res_stasis: Add compare function for bridges
  moh container
  (Reported by Hajek Michal)
 * ASTERISK-29085 - func_curl: Segmentation fault when using
  CURL after setting httpheader CURLOPT
  (Reported by PÃ©ter
  JuhÃ¡sz)
 * ASTERISK-28416 - Unable to get rtp codec payload code for
  slin
  (Reported by Brian J. Murrell)

New Features made in this release:
---
 * ASTERISK-29027 - Implement support for History-Info
 
  (Reported by Torrey Searle)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.38.0

Thank you for your continued support of Asterisk!
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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[asterisk-users] Asterisk 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5 Now Available (Security)

2020-11-05 Thread Asterisk Development Team

The Asterisk Development Team would like to announce security releases for
Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.

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-2020-001: Remote crash in res_pjsip_session
  Upon receiving a new SIP Invite, Asterisk did not return the created dialog
  locked or referenced.

* AST-2020-002: Outbound INVITE loop on challenge with different nonce.
  If Asterisk is challenged on an outbound INVITE and the nonce is changed in
  each response, Asterisk will continually send INVITEs in a loop. This causes
  Asterisk to consume more and more memory since the transaction will never
  terminate (even if the call is hung up), ultimately leading to a restart or
  shutdown of Asterisk. Outbound authentication must be configured on the
  endpoint for this to occur.

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

https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5

The security advisories are available at:

https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
https://downloads.asterisk.org/pub/security/AST-2020-002.pdf

Thank you for your continued support of Asterisk!-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] Asterisk 18.0.0 Now Available

2020-10-20 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
18.0.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.0.0 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:

Security bugs fixed in this release:
---
 * ASTERISK-28589 - chan_sip: Depending on configuration an
  INVITE can alter Addr of a peer
  (Reported by Andrey  V.
  T.)
 * ASTERISK-28580 - Bypass SYSTEM write permission in manager
  action allows system commands execution
  (Reported by Eliel
  SardaÃ±ons)
 * ASTERISK-28495 - res_pjsip_t38: 200 OK with SDP answer with
  declined stream causes crash
  (Reported by Alexei
  Gradinari)

New Features made in this release:
---
 * ASTERISK-6863 - [patch] allow Asterisk to set high ToS bits
  as non-root on Linux
  (Reported by Matt Addison)
 * ASTERISK-17491 - CURLOPT() needs a "followlocation" parameter
  / "maxredirs" doesn't do anything
  (Reported by candrews)
 * ASTERISK-28639 - res_pjsip_endpoint_identifier_ip: Add
  ability to match on source port
  (Reported by Sean Bright)
 * ASTERISK-28614 - app_senddtmf: Allow "receiving" DTMF with
  PlayDTMF instead of only "sending"
  (Reported by lvl)
 * ASTERISK-28613 - func_curl: CURLOPT cannot set Content-Type
  header
  (Reported by Martin Tomec)
 * ASTERISK-28533 - func_jitterbuffer: Add support for video
  synchronization
  (Reported by Joshua C. Colp)
 * ASTERISK-17808 - [patch] Unregister a realtime moh class

  (Reported by Byron Clark)
 * ASTERISK-28489 - Channel variable SIPFROMDOMAIN for
  chan_pjsip to setup From header URI domain
  (Reported by
  Stas Kobzar)

Bugs fixed in this release:
---
 * ASTERISK-29109 - res_pjsip_session: Asterisk 18 does not
  progress calls due to codec negotiation after upgrading from
  Asterisk 16
  (Reported by Ross Beer)
 * ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY
  events
  (Reported by Ove Aursand)
 * ASTERISK-29043 - app_queue: Leave empty sometimes not
  recorded as abandoned
  (Reported by Kfir Itzhak)
 * ASTERISK-29042 - res_parking: Parker UUID is no longer
  copied
  (Reported by Misha Vodsedalek)
 * ASTERISK-28878 - chan_pjsip: PJSIP_MEDIA_OFFER Broken
  asterisk 16
  (Reported by Joseph Ades)
 * ASTERISK-29046 - pbx: Deadlock when doing a reload, while
  simultaneously doing an ExtensionState on a pattern match hint
  that ends up adding an extension
  (Reported by Ramarajan)
 * ASTERISK-29040 - res_speech: Assertion on format
 
  (Reported by Nickolay V. Shmyrev)
 * ASTERISK-29001 - chan_pjsip does not process or forward 181
  responses
  (Reported by Torrey Searle)
 * ASTERISK-29034 - Lastpause of realtime members is reseting
  
  (Reported by Evandro CÃ©sar Arruda)
 * ASTERISK-27273 - app_voicemail: When a voicemail is marked as
  "Urgent", it is not sent by email/processed by the mailcmd
  command
  (Reported by Leandro Dardini)
 * ASTERISK-29033 - res_pjsip_session: Aggressively terminates
  session on failed re-INVITE
  (Reported by Joshua C. Colp)
 * ASTERISK-28974 - res_rtp_asterisk: T.140 messages have
  appended RTP string to each message block.
  (Reported by
  Thomas Johnson)
 * ASTERISK-29011 - chan_sip: ToHost property not cleared on
  reload
  (Reported by Dennis)
 * ASTERISK-29021 - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on
  certified versions
  (Reported by cmaj)
 * ASTERISK-28927 - Asterisk crash in music on hold
 
  (Reported by David Cunningham)
 * ASTERISK-28973 - Malformed IP address in SDP of 2nd SIP timer
  triggered INVITE when NAT is active (UDP transport with
  external_media_address)
  (Reported by Michael Neuhauser)
 * ASTERISK-28995 - res_pjsip_registrar: Expires on statically
  configured contacts is not correct
  (Reported by tootai)
 * ASTERISK-28987 - BridgeCreated ARI event shows wrong
  video_mode info
  (Reported by sungtae kim)
 * ASTERISK-28978 - acl: named_acl rule misconfiguration results
  in segfault on reading rule from realtime
  (Reported by
  Andrew Yager)
 * ASTERISK-28975 - res_http_websocket: Text payload data
  doesn't necessary include trailing zero
  (Reported by
  Nickolay V. Shmyrev)
 * ASTERISK-28951 - Inconsistent behaviour queues.conf when
  there is (not) a [general] section
  (Reported by Walter
  Doekes)
 * ASTERISK-28965 - res_pjsip: Apply outbound proxy to static
  contacts on AOR
  (Reported by Joshua C. Colp)
 * ASTERI

[asterisk-users] Asterisk 17.8.0 Now Available

2020-10-20 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
17.8.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 17.8.0 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:

Bugs fixed in this release:
---
 * ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY
  events
  (Reported by Ove Aursand)
 * ASTERISK-29043 - app_queue: Leave empty sometimes not
  recorded as abandoned
  (Reported by Kfir Itzhak)
 * ASTERISK-29042 - res_parking: Parker UUID is no longer
  copied
  (Reported by Misha Vodsedalek)
 * ASTERISK-28878 - chan_pjsip: PJSIP_MEDIA_OFFER Broken
  asterisk 16
  (Reported by Joseph Ades)
 * ASTERISK-29046 - pbx: Deadlock when doing a reload, while
  simultaneously doing an ExtensionState on a pattern match hint
  that ends up adding an extension
  (Reported by Ramarajan)
 * ASTERISK-29040 - res_speech: Assertion on format
 
  (Reported by Nickolay V. Shmyrev)
 * ASTERISK-29001 - chan_pjsip does not process or forward 181
  responses
  (Reported by Torrey Searle)
 * ASTERISK-29034 - Lastpause of realtime members is reseting
  
  (Reported by Evandro CÃ©sar Arruda)
 * ASTERISK-27273 - app_voicemail: When a voicemail is marked as
  "Urgent", it is not sent by email/processed by the mailcmd
  command
  (Reported by Leandro Dardini)
 * ASTERISK-29033 - res_pjsip_session: Aggressively terminates
  session on failed re-INVITE
  (Reported by Joshua C. Colp)
 * ASTERISK-28974 - res_rtp_asterisk: T.140 messages have
  appended RTP string to each message block.
  (Reported by
  Thomas Johnson)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0

Thank you for your continued support of Asterisk!
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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[asterisk-users] Asterisk 16.14.0 Now Available

2020-10-20 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
16.14.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.14.0 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:

Bugs fixed in this release:
---
 * ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY
  events
  (Reported by Ove Aursand)
 * ASTERISK-29043 - app_queue: Leave empty sometimes not
  recorded as abandoned
  (Reported by Kfir Itzhak)
 * ASTERISK-29042 - res_parking: Parker UUID is no longer
  copied
  (Reported by Misha Vodsedalek)
 * ASTERISK-29029 - Voicemail "pollmailboxes"-option not
  working, bug in function handle_subscribe
  (Reported by
  Karsten Wemheuer)
 * ASTERISK-28878 - chan_pjsip: PJSIP_MEDIA_OFFER Broken
  asterisk 16
  (Reported by Joseph Ades)
 * ASTERISK-29046 - pbx: Deadlock when doing a reload, while
  simultaneously doing an ExtensionState on a pattern match hint
  that ends up adding an extension
  (Reported by Ramarajan)
 * ASTERISK-29040 - res_speech: Assertion on format
 
  (Reported by Nickolay V. Shmyrev)
 * ASTERISK-29001 - chan_pjsip does not process or forward 181
  responses
  (Reported by Torrey Searle)
 * ASTERISK-29034 - Lastpause of realtime members is reseting
  
  (Reported by Evandro CÃ©sar Arruda)
 * ASTERISK-27273 - app_voicemail: When a voicemail is marked as
  "Urgent", it is not sent by email/processed by the mailcmd
  command
  (Reported by Leandro Dardini)
 * ASTERISK-29033 - res_pjsip_session: Aggressively terminates
  session on failed re-INVITE
  (Reported by Joshua C. Colp)
 * ASTERISK-28974 - res_rtp_asterisk: T.140 messages have
  appended RTP string to each message block.
  (Reported by
  Thomas Johnson)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0

Thank you for your continued support of Asterisk!
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] Asterisk 13.37.0 Now Available

2020-10-20 Thread Asterisk Development Team

The Asterisk Development Team would like to announce the release of Asterisk 
13.37.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.37.0 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:

Bugs fixed in this release:
---
 * ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY
  events
  (Reported by Ove Aursand)
 * ASTERISK-29043 - app_queue: Leave empty sometimes not
  recorded as abandoned
  (Reported by Kfir Itzhak)
 * ASTERISK-29042 - res_parking: Parker UUID is no longer
  copied
  (Reported by Misha Vodsedalek)
 * ASTERISK-29029 - Voicemail "pollmailboxes"-option not
  working, bug in function handle_subscribe
  (Reported by
  Karsten Wemheuer)
 * ASTERISK-28878 - chan_pjsip: PJSIP_MEDIA_OFFER Broken
  asterisk 16
  (Reported by Joseph Ades)
 * ASTERISK-29046 - pbx: Deadlock when doing a reload, while
  simultaneously doing an ExtensionState on a pattern match hint
  that ends up adding an extension
  (Reported by Ramarajan)
 * ASTERISK-29040 - res_speech: Assertion on format
 
  (Reported by Nickolay V. Shmyrev)
 * ASTERISK-29001 - chan_pjsip does not process or forward 181
  responses
  (Reported by Torrey Searle)
 * ASTERISK-27273 - app_voicemail: When a voicemail is marked as
  "Urgent", it is not sent by email/processed by the mailcmd
  command
  (Reported by Leandro Dardini)
 * ASTERISK-29033 - res_pjsip_session: Aggressively terminates
  session on failed re-INVITE
  (Reported by Joshua C. Colp)
 * ASTERISK-28974 - res_rtp_asterisk: T.140 messages have
  appended RTP string to each message block.
  (Reported by
  Thomas Johnson)

Improvements made in this release:
---
 * ASTERISK-29010 - Allow disabling of FollowMe prompt
 
  (Reported by Dennis)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.37.0

Thank you for your continued support of Asterisk!
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

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