subject:"\[asterisk\-users\] asterisk"

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

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

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

Thank You!


Change Log for Release asterisk-21.0.2


Links:


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

Summary:


- res_rtp_asterisk: Fix regression issues with DTLS client check

User Notes:



Upgrade Notes:



Closed Issues:


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

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[asterisk-users] asterisk release 20.5.2

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

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

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

Thank You!


Change Log for Release asterisk-20.5.2


Links:


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

Summary:


- res_rtp_asterisk: Fix regression issues with DTLS client check

User Notes:



Upgrade Notes:



Closed Issues:


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

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[asterisk-users] asterisk release 18.20.2

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

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

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

Thank You!


Change Log for Release asterisk-18.20.2


Links:


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

Summary:


- res_rtp_asterisk: Fix regression issues with DTLS client check

User Notes:



Upgrade Notes:



Closed Issues:


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

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

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

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

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

Thank You!


Change Log for Release asterisk-certified-18.9-cert7


Links:


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

Summary:


- res_rtp_asterisk: Fix regression issues with DTLS client check

User Notes:



Upgrade Notes:



Closed Issues:


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

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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

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

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[asterisk-users] asterisk release 21.0.1

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

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

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

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


Change Log for Release asterisk-20.5.1


Links:


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

Summary:


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

User Notes:



Upgrade Notes:



Closed Issues:


None

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[asterisk-users] asterisk release 18.20.1

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

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

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


Change Log for Release asterisk-18.20.1


Links:


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

Summary:


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

User Notes:



Upgrade Notes:



Closed Issues:


None

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Re: [asterisk-users] Asterisk 13 / chan_sip / registration after reject

2023-12-04 Thread Benoit Panizzon

Hello

> > How do I achieve the same with chan_sip?  
> We run a cron script each 10min who will check the registration state 
> and send a register if not registered.

Well it's a simple CPE which needs to be autoprovisioned via either a
tftp config file or TR69.

So that cronjob somehow would also need to be put on the device via one
of those mechanism. We check if there is a way.

Mit freundlichen Grüssen

-Benoît Panizzon-
-- 
I m p r o W a r e   A G-Leiter Commerce Kunden
__

Zurlindenstrasse 29 Tel  +41 61 826 93 00
CH-4133 PrattelnFax  +41 61 826 93 01
Schweiz Web  http://www.imp.ch
__

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Re: [asterisk-users] Asterisk 13 / chan_sip / registration after reject

2023-12-04 Thread List Support


Hello

Le 04/12/2023 à 10:56, Benoit Panizzon (by way of Benoit Panizzon 
) a écrit :

Hi List

We have some CPE which run an embedded asterisk 13 with chan_sip.

Unfortunately, when a registration is rejected, those stop trying.

I am familiar with pjsip which allows to configure:

auth_rejection_permanent=no

How do I achieve the same with chan_sip?
We run a cron script each 10min who will check the registration state 
and send a register if not registered.


--
Daniel
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[asterisk-users] Asterisk 13 / chan_sip / registration after reject

2023-12-04 Thread Benoit Panizzon

Hi List

We have some CPE which run an embedded asterisk 13 with chan_sip.

Unfortunately, when a registration is rejected, those stop trying.

I am familiar with pjsip which allows to configure:

auth_rejection_permanent=no

How do I achieve the same with chan_sip?

Mit freundlichen Grüssen

-Benoît Panizzon-
-- 
I m p r o W a r e   A G-Leiter Commerce Kunden
__

Zurlindenstrasse 29 Tel  +41 61 826 93 00
CH-4133 PrattelnFax  +41 61 826 93 01
Schweiz Web  http://www.imp.ch
__

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Re: [asterisk-users] Asterisk and Teams integration?

2023-10-26 Thread Antony Stone

On Thursday 26 October 2023 at 19:11:45, Carlos Chavez wrote:

>  Does anyone know of a good solution to integrate Asterisk and MS
> Teams?  Something where you can use the MS Teams client as a regular
> extension?

Kamailio is the usual intermediary I have seen for doing this.

Antony.

-- 
If at first you don't succeed, destroy all the evidence that you tried.

   Please reply to the list;
 please *don't* CC me.

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[asterisk-users] Asterisk and Teams integration?

2023-10-26 Thread Carlos Chavez

    Does anyone know of a good solution to integrate Asterisk and MS 
Teams?  Something where you can use the MS Teams client as a regular 
extension?


--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)8116-9161


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[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.
This also

[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 priority change

[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 priority change

Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

2023-09-14 Thread Marek Greško

Hello Jerry,

when you run asterisk using su, ownership of audio device files does not get 
updated. When you login, you get the permissions. You can verify by ls -l and 
getfacl on the device file.

Marek

--- Original Message ---
On Thursday, September 14th, 2023 at 14:33, Jerry Geis  
wrote:

> On Wed, Sep 13, 2023 at 5:20 PM Jerry Geis  wrote:
>
>>>An issue[1] was already created by asterisk at phreaknet.org and they also 
>>>put
>>>a fix up for review and inclusion[2].
>>
>>>[1] https://github.com/asterisk/asterisk/issues/308
>>>[2] https://github.com/asterisk/asterisk/pull/309
>>
>> The change "seems" to be working.
>> Will test more tomorrow - had to leave.
>> THANKS!
>> Jerry
>
> Yes - this fix is working for me.
>
> Only issue I have now is, I used to run asterisk like this:
> su silentm -c "/usr/sbin/asterisk -fn"
> I also tried
> su silentm -l -c "/usr/sbin/asterisk -fn"
>
> these do not work for the chan_console. I have to actually login as silentm 
> and then run asterisks - to HEAR the audio.
> doing su above I do not hear the audio - but the CLI looks the same - no 
> errors.
>
> Thoughts?
>
> Jerry-- 
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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

2023-09-14 Thread Jerry Geis

On Wed, Sep 13, 2023 at 5:20 PM Jerry Geis  wrote:

> >An issue[1] was already created by asterisk at phreaknet.org and they
> also put
> >a fix up for review and inclusion[2].
>
> >[1] https://github.com/asterisk/asterisk/issues/308
> >[2] https://github.com/asterisk/asterisk/pull/309
>
>
> The change "seems" to be working.
> Will test more tomorrow - had to leave.
> THANKS!
>
> Jerry
>

Yes - this fix is working for me.

Only issue I have now is, I used to run asterisk like this:
su silentm -c "/usr/sbin/asterisk -fn"
I also tried
su silentm -l -c "/usr/sbin/asterisk -fn"

these do not work for the chan_console.  I have to actually login as
silentm and then run asterisks - to HEAR the audio.
doing su above I do not hear the audio - but the CLI looks the same - no
errors.

Thoughts?

Jerry
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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

>An issue[1] was already created by asterisk at phreaknet.org and they also
put
>a fix up for review and inclusion[2].

>[1] https://github.com/asterisk/asterisk/issues/308
>[2] https://github.com/asterisk/asterisk/pull/309


The change "seems" to be working.
Will test more tomorrow - had to leave.
THANKS!

Jerry
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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

2023-09-13 Thread Joshua C. Colp

An issue[1] was already created by aster...@phreaknet.org and they also put
a fix up for review and inclusion[2].

[1] https://github.com/asterisk/asterisk/issues/308
[2] https://github.com/asterisk/asterisk/pull/309

On Wed, Sep 13, 2023 at 4:27 PM Jerry Geis  wrote:

>
> I have found that I can add the restart of asterisk (killall -9 asterisk)
> to the h extension- BOY is that UGLY.
>
> chan_console is not a testing device - how can we get this nasty bug fixed
> ?
>
> Jerry
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> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
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-- 
Joshua C. Colp
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Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

I have found that I can add the restart of asterisk (killall -9 asterisk)
to the h extension- BOY is that UGLY.

chan_console is not a testing device - how can we get this nasty bug fixed ?

Jerry
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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

> After a hung call, can you run core restart now from the asterisk console?

Doing a "killall -9 asterisk" is the only thing that works
I tried killall asterisk - does not free up the channel
the asterisk "core restart now" takes like a good 20 seconds to return but
does work.

The issue is I cannot run it after teh Dial() as the
Dial(Console/default,20,g) never returns to the dial plan.

jerry
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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

2023-09-13 Thread Doug Lytle

It worked with my test.  I'm on Asterisk 18.19.0

-- Executing [517xxx@voipms:4] System("IAX2/voipms-15815", "asterisk 
-rx 'core restart now'") in new stack
-- Remote UNIX connection
Asterisk uncleanly ending (0).
Executing last minute cleanups
  == Destroying musiconhold processes
  == Manager unregistered action DBGet
  == Manager unregistered action DBGetTree
  == Manager unregistered action DBPut
  == Manager unregistered action DBDel
  == Manager unregistered action DBDelTree
Preparing for Asterisk restart...
Asterisk is now restarting...
asterisk*CLI> 
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups

After a hung call, can you run core restart now from the asterisk console?

Doug

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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

>Using system() you could issue a asterisk -rx 'core restart now'

So I tried this


exten => s,1,Playback(beep)
exten => s,n,Dial(Console/default,20,g)
exten => s,n,Hangup
exten => s,n,System(asterisk -rx 'core restart now')

But it does not continue. Last thing I see is "Exited non zero"
so its not doing the hangup or the system.

jerry
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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

I have noticed that once my message speaks - the server thinks its done and
HUNGUP,
the endpoint STILL thinks the channel is active - the last message says
"Rx: BYE" on sip show channels
I tried ADDING to Dial() ,20,g and then had a Hangup after teh dial.
Its NOT getting there to hangup.

Jerry
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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

2023-09-13 Thread Doug Lytle

>> Is there a dial plan call that can "exit asterisk" or completely reload 
>> everything - killall active calls and start over ?

Using system() you could issue a asterisk -rx 'core restart now'

Doug

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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

Is there a dial plan call that can "exit asterisk" or completely reload
everything - killall active calls and start over ?

seems the console/dummy (chan_console) is holding some resource. How do I
just "exit" and start over after call came in ?

Thanks

Jerry
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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

2023-09-10 Thread olivas


I don't know if this will help you, but looking back through an old config I 
have for an older version of Asterisk, I had used chan_console with the old and 
now defunct app_rpt app to listen to audio on various nodes via the console for 
testing.

Here is what I did:

In console.conf, I defined this:
[default]
input_device = default
output_device = default
autoanswer = no
context = 
extension = 
callerid = 
language = en
overridecontext = no
mohintrepret = default
active = yes

In modules.conf I loaded the audio module (in this case it was chan_alsa.so, 
but I also could use chan_oss.so).  I made sure noload was commented out for 
chan_alsa.so

In alsa.conf, I defined some of the same things as in console.conf:
[general]
autoanswer=no
context=
extension=
inputdevice=plughw:0,1
otuputdevice=plughw:0,0
mute=true 



You'll need to check your ALSA device to see what the input and output devices 
are.

That last line is important, since on the console you may not have a mic that 
works to talk, you just want to listen,


In extensions.conf, I defined a dialplan that instead of trying to dial out, it 
just answered the call and then threw me into the app.

Then to dial from the console, I woudl use:
console dial 

And it woudl use the context I defined and launch the Rpt app.

What you could do is define somehting like this ,but have the extension use DISA so that 
you can then get dumped into your normal dialplan logic where you could use "console 
dial xxx".


No guarantees that this will work with a newer version of Asterisk, but this 
did work with a 1.8 setup I used to have (that I have the configs saved for).

-Stacy

On 9/8/23 10:28 AM, Jerry Geis  wrote:


So I have done through chan_console.c and searched for 
console_pct_lock() - every one - has a matching console_pvt_unlock()


How is the deadlock occurring ?

jerry





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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

2023-09-08 Thread Jerry Geis

So I have done through chan_console.c and searched for console_pct_lock() -
every one - has a matching console_pvt_unlock()

How is the deadlock occurring ?

jerry
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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

2023-09-08 Thread Doug Lytle

>>> How do we get this working

For the time being, go back to 18.14.0

Doug

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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

2023-09-08 Thread Jerry Geis

>
>
> Not sure if this is the same thing you're seeing, but chan_console
> currently has a known deadlock issue that has not been resolved:
> https://issues-archive.asterisk.org/ASTERISK-30481
> It's quite easy to reproduce, so it may be the case that the module is
> currently unusable.
>

Well this is a bummer

 [Sep  8 08:45:28] WARNING[313684][C-0001]: chan_console.c:651
console_indicate: Don't know how to display condition 26 on Console/default
  --- <("<) --- Hangup on Console --- (>")> ---
[Sep  8 08:45:28] ERROR[313683]: utils.c:1027 lock_info_destroy: Thread
'stream_monitor   started at [  390] chan_console.c start_stream()'
still has a lock! - 'pvt' (0x55e647a245e0) from 'stream_monitor' in
chan_console.c:281!

How do we get this working

Jerry
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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

2023-09-08 Thread asterisk


On 9/8/2023 8:18 AM, Jerry Geis wrote:

But when I do a second test. Asterisk HANGS on ChanIsAvail()

Then I thought lets SKIP that - and just let it do the Dial() - I 
stopped everything - got it running again. - and then the Dial() hangs 
on the second call.


So both ChanIsAvail() or Dial() both hang on the second call in.

So only 1 call in will work.
Below is the CLI report of the call that works.

This is my context
[smvoice-mediacontroller-public-address]
exten => s,1,ChanIsAvail(Console/default)
exten => s,n,GotoIf($["${AVAILCHAN}" = ""]?smvoice-busy,s,1)
exten => s,n,Playback(beep)
exten => s,n,Dial(Console/default)
exten => s,n,Hangup


Not sure if this is the same thing you're seeing, but chan_console 
currently has a known deadlock issue that has not been resolved: 
https://issues-archive.asterisk.org/ASTERISK-30481
It's quite easy to reproduce, so it may be the case that the module is 
currently unusable.


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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

2023-09-08 Thread Jerry Geis

Some progress to report:

I had to run asterisk as the user logged in -  actually not even that. I
could not "su user -c " to that user - I had to run it as that user.
Then I did a test and got audio! Great...
But when I do a second test. Asterisk HANGS on ChanIsAvail()

Then I thought lets SKIP that - and just let it do the Dial() - I stopped
everything - got it running again. - and then the Dial() hangs on the
second call.

So both ChanIsAvail() or Dial() both hang on the second call in.

So only 1 call in will work.
Below is the CLI report of the call that works.

This is my context
[smvoice-mediacontroller-public-address]
exten => s,1,ChanIsAvail(Console/default)
exten => s,n,GotoIf($["${AVAILCHAN}" = ""]?smvoice-busy,s,1)
exten => s,n,Playback(beep)
exten => s,n,Dial(Console/default)
exten => s,n,Hangup

Now what ???

Jerry


onnected to Asterisk 18.18.0 currently running on nuc11cdev2 (pid = 282195)
  == Using SIP RTP CoS mark 5
   > 0x7feeec0086b0 -- Strict RTP learning after remote address set to:
192.168.1.8:17526
-- Executing [public_address@smvoice-mediacontroller:1]
SoftHangup("SIP/devgeis_to_nuc11cdev2-", "ALSA/dummy") in new stack
-- Executing [public_address@smvoice-mediacontroller:2]
Goto("SIP/devgeis_to_nuc11cdev2-",
"smvoice-mediacontroller-public-address,s,1") in new stack
-- Goto (smvoice-mediacontroller-public-address,s,1)
-- Executing [s@smvoice-mediacontroller-public-address:1]
NoOp("SIP/devgeis_to_nuc11cdev2-", "JERRY") in new stack
-- Executing [s@smvoice-mediacontroller-public-address:2]
Playback("SIP/devgeis_to_nuc11cdev2-", "beep") in new stack
   > 0x7feeec0086b0 -- Strict RTP switching to RTP target address
192.168.1.8:17526 as source
--  Playing 'beep.gsm' (language
'en')
-- Executing [s@smvoice-mediacontroller-public-address:3]
Dial("SIP/devgeis_to_nuc11cdev2-", "Console/default") in new stack
  --- <("<) --- Call to device 'default' on console from 'MyName Here'
<2564286000> --- (>")> ---
  --- <("<) --- Auto-answered --- (>")> ---
-- Called Console/default
-- Console/default answered SIP/devgeis_to_nuc11cdev2-
-- Channel Console/default joined 'simple_bridge' basic-bridge

[Sep  8 08:07:10] WARNING[282457][C-0001]: chan_console.c:651
console_indicate: Don't know how to display condition 26 on Console/default
-- Channel SIP/devgeis_to_nuc11cdev2- joined 'simple_bridge'
basic-bridge 
   > 0x7feeec0086b0 -- Strict RTP learning complete - Locking on source
address 192.168.1.8:17526
-- Channel SIP/devgeis_to_nuc11cdev2- left 'simple_bridge'
basic-bridge 
-- Channel Console/default left 'simple_bridge' basic-bridge

[Sep  8 08:07:17] WARNING[282457][C-0001]: chan_console.c:651
console_indicate: Don't know how to display condition 26 on Console/default
  --- <("<) --- Hangup on Console --- (>")> ---
  == Spawn extension (smvoice-mediacontroller-public-address, s, 3) exited
non-zero on 'SIP/devgeis_to_nuc11cdev2-'
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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

2023-09-07 Thread Doug Lytle

In the old days when I was using console/dsp, I would have to use 
alsamix from the console to verify that the output wasn't muted.  Maybe 
it's still the same.


Doug

On 9/7/23 03:43 PM, Jerry Geis wrote:

ok switching to "Console/default" does show the text
 --- <("<) --- Call to device 'default' on console from 'default' 
<2564286000> --- (>")> ---

  --- <("<) --- Auto-answered --- (>")> ---

However I don't hear any audio.



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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

ok switching to "Console/default" does show the text
 --- <("<) --- Call to device 'default' on console from 'default'
<2564286000> --- (>")> ---
  --- <("<) --- Auto-answered --- (>")> ---

However I don't hear any audio.

Thanks

Jerry
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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

On Thu, Sep 7, 2023 at 3:27 PM Jerry Geis  wrote:

>
> I found "console list available"
>
> ===
> === -
> === Device Name: default
> === ---> Default Input Device
> === ---> Default Output Device
> === -
> ===
> === -
> === Device Name: dmix
> === ---> Output Device
> === -
> ===
> =
>
> dmix is there and default is there
> I tried both - and get the same error
> Console device "dmix" not found . etc.
>

Yes, because that lists the available devices. You have to configure it in
console.conf in order to be able to dial it. If you haven't configured a
thing named "dmix" in console.conf, then it's not going to work.

"console list available" show available devices that you can use in the
configuration
"console list devices" show what is actually configured

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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

I found "console list available"

===
=== -
=== Device Name: default
=== ---> Default Input Device
=== ---> Default Output Device
=== -
===
=== -
=== Device Name: dmix
=== ---> Output Device
=== -
===
=

dmix is there and default is there
I tried both - and get the same error
Console device "dmix" not found . etc.


Jerry
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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

On Thu, Sep 7, 2023 at 3:20 PM Joshua C. Colp  wrote:

> On Thu, Sep 7, 2023 at 3:15 PM Jerry Geis  wrote:
>
>> Joshua
>>
>> Asterisk 18.14.0 with chan_alsa and Console/dsp works.
>> This does not work in 18.18.0 with chan_console enabled.
>> I am on Ubuntu 20.04 LTS.
>>
>> Is there a howto for the new chan_console ?
>>
>
> I'm not aware of one. The module itself has existed since at least
> Asterisk 1.8
>
>
>> how can I get this working again ?
>> I am trying to just play audio on pulse audio.
>>
>
> I don't have anything additional to add beyond what I've said and the
> config file I've provided.
>

I can say that with the default configuration file it would be
Console/default though, and would use the system default input and output
devices according to PortAudio.

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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

On Thu, Sep 7, 2023 at 3:15 PM Jerry Geis  wrote:

> Joshua
>
> Asterisk 18.14.0 with chan_alsa and Console/dsp works.
> This does not work in 18.18.0 with chan_console enabled.
> I am on Ubuntu 20.04 LTS.
>
> Is there a howto for the new chan_console ?
>

I'm not aware of one. The module itself has existed since at least Asterisk
1.8

> how can I get this working again ?
> I am trying to just play audio on pulse audio.
>

I don't have anything additional to add beyond what I've said and the
config file I've provided.

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[asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa

Joshua

Asterisk 18.14.0 with chan_alsa and Console/dsp works.
This does not work in 18.18.0 with chan_console enabled.
I am on Ubuntu 20.04 LTS.

Is there a howto for the new chan_console ?
how can I get this working again ?
I am trying to just play audio on pulse audio.

Thanks,

Jerry
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Re: [asterisk-users] asterisk 18.18.0 and chan_console

On Thu, Sep 7, 2023 at 3:07 PM Jerry Geis  wrote:

>
> I am trying to get audio to play on Pulse - so just the monitor basically.
>
> I have tried Console/dsp, Console/dmix, Console/pulse, and a couple others.
>
> The error is always the same "console_request: Console device 'dmix' not
> found.
>
> What is the correct "Console/" to play on pulse for UBuntu 20.04 LTS ?
> I can "aplay /usr/share/sounds/alsa/Front_Center.wav" no problem.
>
> Thoughts?
>

It has a configuration file[1] that defines the various devices and their
referenced name. If default is in use then I'd expect Console/default

[1]
https://github.com/asterisk/asterisk/blob/master/configs/samples/console.conf.sample

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[asterisk-users] Asterisk 16.23.0 strange issue where Answer request succeeds and able to perform actions but Asterisk never sent 200 OK to answer call

2023-09-07 Thread Dan Cropp

Some background...
We use AMI and AsyncAGI to be able to receive events about calls (and other 
Asterisk details) and control it from our application.
Works great and have about 100 sites (some newer, some older) without issues.


I was notified this morning about a customer who had something strange happen 
and I can't explain it.

Asterisk 16.23.0 and PJSIP.

Call comes into Asterisk.
Asterisk sends the Trying.
Via AMI, notified of the call and dial plan has it go to AsyncAGI for our 
application to be able to control the call.
Via AMI, we tell Asterisk to Answer.
Asterisk processes it and indicates it was answered.
The Asterisk AMI/AGI indicates call was answered successfully, call state is 
Up, etc.
Everything appears to be normal.
We perform various actions on the call, example play a file, music, tones, etc.

However, Asterisk never sent the 200 OK to answer the call.
Seems as though Asterisk is in a bizarre state where it thinks it is handling 
the call, but it really isn't.


Reports are this happened to several calls.

Eventually, they restarted the entire VM and everything started working well.


We think this may be caused by something their switch is doing.
Through the grapevine, heard they had some network issue but don't know the 
details of their switches and architecture for calls coming into Asterisk.

We noticed we are seeing two INVITEs happen with the same Call-ID, but 2 
additional Record-Route header/value pairs and 3 additional Via header/value 
pairs.  At least in first glance, the rest seems to be the same.  I see 
Asterisk created two different PJSIP calls for each despite same Call-ID, but I 
am guessing that's because of the additional Via or Record-Route pairs.

Is it possible multiple of these double INVITEs could cause Asterisk or PJSIP 
on this older software to get into a bad state to cause the issues with AMI and 
Asterisk state?

Dan
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Re: [asterisk-users] asterisk 18.18.0 and chan_console

I am trying to get audio to play on Pulse - so just the monitor basically.

I have tried Console/dsp, Console/dmix, Console/pulse, and a couple others.

The error is always the same "console_request: Console device 'dmix' not
found.

What is the correct "Console/" to play on pulse for UBuntu 20.04 LTS ?
I can "aplay /usr/share/sounds/alsa/Front_Center.wav" no problem.

Thoughts?

Jerry
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Re: [asterisk-users] asterisk 18.18.0 and chan_console

2023-09-07 Thread Doug Lytle


On 9/6/23 03:23 PM, Jerry Geis wrote:

I am trying to just play on PulseAudio actually.
This used to work - I have just recently updated to 18.18.0, so I'm 
puzzled.


All of my Asterisk installs are running in virtual machines, so I have 
no way to test.


Doug
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Re: [asterisk-users] asterisk 18.18.0 and chan_console

> What is the device that you're connecting to?

I am trying to just play on PulseAudio actually.
This used to work - I have just recently updated to 18.18.0, so I'm puzzled.

Jerry
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Re: [asterisk-users] asterisk 18.18.0 and chan_console

What is the device that you're connecting to?

Doug

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Re: [asterisk-users] asterisk 18.18.0 and chan_console

This is hte error I get for Console/dsp or console/dsp

ERROR[230711][C-0001]: chan_console.c:477 console_request: Console
device 'dsp' not found

Jerry
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Re: [asterisk-users] asterisk 18.18.0 and chan_console

In a past work life, I did use console/dsp to connect to a sound card that 
hooked up to a bogan paging amp.  I still have access to the programming and 
everything I have show as using a lower case c for console

Doug

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Re: [asterisk-users] asterisk 18.18.0 and chan_console

> I don't use it; just figured I'd try to help

Thanks Doug...

So then for the list - I have chan_console working now
But I am trying Console/dsp and Console/ALSA and both give an error about
not found.

What have I missed ?
Thanks

Jerry
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Re: [asterisk-users] asterisk 18.18.0 and chan_console

>>> hi Doug - so what device do you use?  I am getting and error for Console/dsp

I don't use it; just figured I'd try to help.

Doug

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Re: [asterisk-users] asterisk 18.18.0 and chan_console

>
>
>
> Oh that is a good one - I thought I did - but apparently not. menuconfig
> now shows "*"
>
> So is chan_alsa going away ? What is it being replaced with?
>
> thank you!
>
> Jerry
>

hi Doug - so what device do you use?  I am getting and error for Console/dsp

 exten => s,1,ChanIsAvail(Console/dsp)
exten => s,n,GotoIf($["${AVAILCHAN}" = ""]?smvoice-busy,s,1)
exten => s,n,Playback(beep)
exten => s,n,Dial(Console/dsp)
exten => s,n,Hangup

I also tried Console/ALSA and both gave errors.

Thanks

Jerry
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Re: [asterisk-users] asterisk 18.18.0 and chan_console

2023-09-06 Thread Joshua C. Colp

On Wed, Sep 6, 2023 at 12:01 PM Jerry Geis  wrote:

>
>> Just to verify that you did rerun configure after installing the
>> libraries?
>>
>> Doug
>>
>
> Oh that is a good one - I thought I did - but apparently not. menuconfig
> now shows "*"
>
> So is chan_alsa going away ? What is it being replaced with?
>

The chan_alsa module has been removed in Asterisk 21[1]. The recommended
replacement is chan_console.

[1] https://docs.asterisk.org/Development/Asterisk-Module-Deprecations/

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Re: [asterisk-users] asterisk 18.18.0 and chan_console

>
>
> Just to verify that you did rerun configure after installing the libraries?
>
> Doug
>

Oh that is a good one - I thought I did - but apparently not. menuconfig
now shows "*"

So is chan_alsa going away ? What is it being replaced with?

thank you!

Jerry
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Re: [asterisk-users] asterisk 18.18.0 and chan_console


>>> Thanks doug - I did that - still showing XXX for chan_console 

Just to verify that you did rerun configure after installing the libraries?

Doug


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Re: [asterisk-users] asterisk 18.18.0 and chan_console

> portaudio19-dev

Thanks doug - I did that - still showing XXX for chan_console


libportaudio2/focal,now 19.6.0-1build1 amd64 [installed]
libportaudiocpp0/focal,now 19.6.0-1build1 amd64 [installed,automatic]
portaudio19-dev/focal,now 19.6.0-1build1 amd64 [installed]


Jerry
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Re: [asterisk-users] asterisk 18.18.0 and chan_console

On my debian 11 install I needed to install

portaudio19-dev

Doug



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[asterisk-users] asterisk 18.18.0 and chan_console

I am still using chan_console.
I compiled 18.18.0 and chan_console is not built.
I am using ubuntu 20.04.6 LTS

make menuselect says XXX chan_consoel and it needs "portaudio"

What do I do next ?

Also menuconfig is saying XXX on Also - what alsa library is needed ?

Thanks

jerry
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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 the

[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

Re: [asterisk-users] asterisk sees private IP address of a device behind NAT

2023-07-11 Thread Antony Stone

On Tuesday 11 July 2023 at 10:00:22, Fourhundred Thecat wrote:

> Hello,
> 
> my asterisk is working fine, I am just confused why, on the server I see
> private IP address of an endpoint

SIP is rather like FTP in that it embeds IP addresses (layer 3 of the OSI 
network model) in the protocol itself (layer 7).  I have always found it 
amazing that after we'd been struggling with this design flaw in FTP for years, 
it was decided to repeat it in RFC 2543.

It is one of the reasons why SIP through NAT is more challenging than, say, 
HTTP is, and one of its side effects is that you often see "real" (private) IP 
addresses of endpoints inside the communications when the packet source and 
destination addresses are the routed (public) versions.

Unless you experience actual problems in call setup or take-down, or things 
like one-way audio, just ignore it and accept that SIP was perhaps not as well 
designed as it could have been.

Antony.

-- 
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 please *don't* CC me.

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[asterisk-users] asterisk sees private IP address of a device behind NAT

2023-07-11 Thread Fourhundred Thecat


Hello,

my asterisk is working fine, I am just confused why, on the server I see
private IP address of an endpoint:

  WARNING: Retransmission timeout reached on transmission
0_252301488@10.1.3.8 for seqno 2 (Critical Response)

the IP 10.1.3.8 is a phone behind NAT.
Does it mean something is misconfigured?
Shouldn't asterisk only see the public IP of the router ?

thanks,

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Re: [asterisk-users] Asterisk Release 20.3.1

2023-07-08 Thread Michael Maier


At the moment, I can't see any differences here. sha512sum is identical.

Regards
Michael

On 08.07.23 at 01:50 Jean-Denis Girard wrote:

Le 07/07/2023 à 12:49, Joshua C. Colp a écrit :
On Fri, Jul 7, 2023 at 6:40 PM Jean-Denis Girard > wrote:


    There seems to be a problem with the tar.gz archive on github. It's
    correct on downloads.asterisk.org .

Can you be more specific? They are identical and the same tarball. I just 
downloaded both from each place and confirmed that, and confirmed they both 
extract fine.


Downloading from github (I tried 5 times), I get:
10353870  7 juil. 13:44 'asterisk-20.3.1(1).tar.gz'
10353870  7 juil. 13:46 'asterisk-20.3.1(2).tar.gz'
sha256sum is:
aec7271fda5eb1e185bb94f3f52977c636783bd456e9c361dd853cd0eba10203
Extracting is fine.

Downloading from asterisk.org, I get:
28176262  7 juil. 11:34  asterisk-20.3.1.tar.gz
5d7dea82b11ce97eec294ba0234c3a68fe2f05065c04a4279baa4a4442f4f628


Bien cordialement,



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Re: [asterisk-users] Asterisk Release 20.3.1

2023-07-07 Thread Jean-Denis Girard


Le 07/07/2023 à 12:49, Joshua C. Colp a écrit :
On Fri, Jul 7, 2023 at 6:40 PM Jean-Denis Girard > wrote:


There seems to be a problem with the tar.gz archive on github. It's
correct on downloads.asterisk.org . 



Can you be more specific? They are identical and the same tarball. I 
just downloaded both from each place and confirmed that, and confirmed 
they both extract fine.


Downloading from github (I tried 5 times), I get:
10353870  7 juil. 13:44 'asterisk-20.3.1(1).tar.gz'
10353870  7 juil. 13:46 'asterisk-20.3.1(2).tar.gz'
sha256sum is:
aec7271fda5eb1e185bb94f3f52977c636783bd456e9c361dd853cd0eba10203
Extracting is fine.

Downloading from asterisk.org, I get:
28176262  7 juil. 11:34  asterisk-20.3.1.tar.gz
5d7dea82b11ce97eec294ba0234c3a68fe2f05065c04a4279baa4a4442f4f628


Bien cordialement,
--
Jean-Denis Girard

SysNux   Systèmes   Linux   en   Polynésie  française
https://www.sysnux.pf/   Tél: +689 40.50.10.40 / GSM: +689 87.797.527

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Re: [asterisk-users] Asterisk Release 20.3.1

2023-07-07 Thread Joshua C. Colp

On Fri, Jul 7, 2023 at 6:40 PM Jean-Denis Girard 
wrote:

> There seems to be a problem with the tar.gz archive on github. It's
> correct on downloads.asterisk.org.

Can you be more specific? They are identical and the same tarball. I just
downloaded both from each place and confirmed that, and confirmed they both
extract fine.

-- 
Joshua C. Colp
Asterisk Project Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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Re: [asterisk-users] Asterisk Release 20.3.1

2023-07-07 Thread Jean-Denis Girard

There seems to be a problem with the tar.gz archive on github. It's
correct on downloads.asterisk.org.

Thanks,
--
Jean-Denis Girard

SysNux Systèmes Linux en Polynésie française
https://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.797.527

Le 07/07/2023 à 10:10, Asterisk Development Team a écrit :

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

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

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

Change Log for Release 20.3.1

Links:

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

Summary:

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

User Notes:

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

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

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

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

Upgrade Notes:

Closed Issues:

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

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[asterisk-users] Asterisk Release 20.3.1

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:

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

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

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

Upgrade Notes:

Closed Issues:

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

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

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

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

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

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


Change Log for Release 19.8.1


Links:


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

Summary:


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

User Notes:



Upgrade Notes:



Closed Issues:


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

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[asterisk-users] Asterisk Release 18.18.1

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

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

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

Change Log for Release 18.18.1

Links:

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

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

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

Summary:

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

User Notes:

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

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

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

Upgrade Notes:

Closed Issues:

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

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[asterisk-users] Asterisk Release 16.30.1

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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Re: [asterisk-users] Asterisk not replacing private FROM ip with public IP in INVITE

2023-06-21 Thread Joshua C. Colp

On Wed, Jun 21, 2023 at 4:07 PM TTT  wrote:

> Something perhaps noteworth, since this is a multihomed system I bound the
> transport to 172.31.253.4:5060
>
> I don't *think* that would cause Asterisk to use that IP in the FROM...at
> least it shouldn't.
>
>
Copy/paste from FreePBX forum:

It doesn’t touch the From header because it doesn’t matter for normal use.
There is a “from_domain” option which can be used to explicitly set the
domain portion of the From header. It’s unlikely to be your problem, unless
Twilio requires a specific From domain.

-- 
Joshua C. Colp
Asterisk Project Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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Re: [asterisk-users] Asterisk not replacing private FROM ip with public IP in INVITE

Something perhaps noteworth, since this is a multihomed system I bound the 
transport to 172.31.253.4:5060

I don't *think* that would cause Asterisk to use that IP in the FROM...at least 
it shouldn't. 

-Original Message-
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of TTT
Sent: Wednesday, June 21, 2023 2:58 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 

Subject: Re: [asterisk-users] Asterisk not replacing private FROM ip with 
public IP in INVITE

I tried that (only needed to add rewrite_contact=yes) but it didn't help.

BTW, the CONTACT: line holds the correct ip!  Only the FROM: line holds the 
wrong (private) IP.

I'm still learning SIP...but I assume the FROM should also hold the rewritten 
public IP.  Just don't know how to force Asterisk to do that.

-Original Message-
From: Eric Wieling [mailto:ewiel...@nyigc.com] 
Sent: Wednesday, June 21, 2023 2:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
; TTT 
Subject: Re: [asterisk-users] Asterisk not replacing private FROM ip with 
public IP in INVITE

type=endpoint
rewrite_contact=yes
force_rport=yes
rtp_symmetric=yes


On 6/21/23 14:36, TTT wrote:
> I've split this thread off from another (PJSIP authentication) because I 
> think the root cause is something different.I think the problem is the 
> following FROM line in my SIP INVITE transaction:
> 
> From: "MYNAME" 
> ;tag=773a3e6a-a677-4fb1-95fc-54b379b650a4
> 
> The IP address above is an internal/non-routable IP, so Twilio is rejecting 
> it.  For some reason Asterisk is not replacing the private IP with my public 
> IP address.  My pjsip.transport.conf contains a stanza for this transport 
> with:
> 
> local_net=172.31.0.0/16
> 
> Is that all that's needed for Asterisk to replace the from IP with the 
> external IP?  I'm not clear on why Asterisk is not substituting the private 
> FROM ip with a public one...
> 
> 
> 

-- 
http://help.nyigc.net/


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Re: [asterisk-users] Asterisk not replacing private FROM ip with public IP in INVITE

I tried that (only needed to add rewrite_contact=yes) but it didn't help.

BTW, the CONTACT: line holds the correct ip!  Only the FROM: line holds the 
wrong (private) IP.

I'm still learning SIP...but I assume the FROM should also hold the rewritten 
public IP.  Just don't know how to force Asterisk to do that.

-Original Message-
From: Eric Wieling [mailto:ewiel...@nyigc.com] 
Sent: Wednesday, June 21, 2023 2:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
; TTT 
Subject: Re: [asterisk-users] Asterisk not replacing private FROM ip with 
public IP in INVITE

type=endpoint
rewrite_contact=yes
force_rport=yes
rtp_symmetric=yes


On 6/21/23 14:36, TTT wrote:
> I've split this thread off from another (PJSIP authentication) because I 
> think the root cause is something different.I think the problem is the 
> following FROM line in my SIP INVITE transaction:
> 
> From: "MYNAME" 
> ;tag=773a3e6a-a677-4fb1-95fc-54b379b650a4
> 
> The IP address above is an internal/non-routable IP, so Twilio is rejecting 
> it.  For some reason Asterisk is not replacing the private IP with my public 
> IP address.  My pjsip.transport.conf contains a stanza for this transport 
> with:
> 
> local_net=172.31.0.0/16
> 
> Is that all that's needed for Asterisk to replace the from IP with the 
> external IP?  I'm not clear on why Asterisk is not substituting the private 
> FROM ip with a public one...
> 
> 
> 

-- 
http://help.nyigc.net/


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Re: [asterisk-users] Asterisk not replacing private FROM ip with public IP in INVITE

I've already done that.  However, I used the FQDN instead of an IP address 
which I think should be ok.

-Original Message-
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of Carlos Chavez
Sent: Wednesday, June 21, 2023 2:53 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk not replacing private FROM ip with 
public IP in INVITE

You need to put your external IP in the transport configuration:

external_media_address=X.X.X.X
external_signaling_address=X.X.X.X
external_signaling_port=5060


On 21/06/23 12:36, TTT wrote:
> I've split this thread off from another (PJSIP authentication) because I 
> think the root cause is something different.I think the problem is the 
> following FROM line in my SIP INVITE transaction:
>
> From: "MYNAME" 
> ;tag=773a3e6a-a677-4fb1-95fc-54b379b650a4
>
> The IP address above is an internal/non-routable IP, so Twilio is rejecting 
> it.  For some reason Asterisk is not replacing the private IP with my public 
> IP address.  My pjsip.transport.conf contains a stanza for this transport 
> with:
>
> local_net=172.31.0.0/16
>
> Is that all that's needed for Asterisk to replace the from IP with the 
> external IP?  I'm not clear on why Asterisk is not substituting the private 
> FROM ip with a public one...
>
>
>
-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)8116-9161


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Re: [asterisk-users] Asterisk not replacing private FROM ip with public IP in INVITE

2023-06-21 Thread Carlos Chavez


You need to put your external IP in the transport configuration:

external_media_address=X.X.X.X
external_signaling_address=X.X.X.X
external_signaling_port=5060


On 21/06/23 12:36, TTT wrote:

I've split this thread off from another (PJSIP authentication) because I think 
the root cause is something different.I think the problem is the following 
FROM line in my SIP INVITE transaction:

From: "MYNAME" 
;tag=773a3e6a-a677-4fb1-95fc-54b379b650a4

The IP address above is an internal/non-routable IP, so Twilio is rejecting it. 
 For some reason Asterisk is not replacing the private IP with my public IP 
address.  My pjsip.transport.conf contains a stanza for this transport with:

local_net=172.31.0.0/16

Is that all that's needed for Asterisk to replace the from IP with the external 
IP?  I'm not clear on why Asterisk is not substituting the private FROM ip with 
a public one...




--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)8116-9161


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Re: [asterisk-users] Asterisk not replacing private FROM ip with public IP in INVITE

2023-06-21 Thread Eric Wieling


type=endpoint
rewrite_contact=yes
force_rport=yes
rtp_symmetric=yes


On 6/21/23 14:36, TTT wrote:

I've split this thread off from another (PJSIP authentication) because I think 
the root cause is something different.I think the problem is the following 
FROM line in my SIP INVITE transaction:

From: "MYNAME" 
;tag=773a3e6a-a677-4fb1-95fc-54b379b650a4

The IP address above is an internal/non-routable IP, so Twilio is rejecting it. 
 For some reason Asterisk is not replacing the private IP with my public IP 
address.  My pjsip.transport.conf contains a stanza for this transport with:

local_net=172.31.0.0/16

Is that all that's needed for Asterisk to replace the from IP with the external 
IP?  I'm not clear on why Asterisk is not substituting the private FROM ip with 
a public one...





--
http://help.nyigc.net/

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[asterisk-users] Asterisk not replacing private FROM ip with public IP in INVITE