from:"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-dev] 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-dev] 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-dev] 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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[asterisk-dev] CORRECTED asterisk release 21.0.1

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

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

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

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


Change Log for Release asterisk-21.0.1


Links:


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

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

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

Summary:


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

User Notes:


Upgrade Notes:


Closed Issues:


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

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

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

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

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


Change Log for Release asterisk-certified-18.9-cert6


Links:


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

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

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

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

Summary:


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

User Notes:


Upgrade Notes:


Closed Issues:


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

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-dev] 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-dev] 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-dev] 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-dev] Mailing List Future

2023-12-13 Thread asterisk

On 12/13/2023 7:55 AM, Joshua C. Colp wrote:
On Wed, Dec 13, 2023 at 8:45 AM Jonathan Simpson
mailto:jsimp...@jdsnetwork.com>> wrote:

The mixed content is useful.

Learning about stir shaken updates, useful. Would that have been
in a github notification? Would the subject line be parsable?

My inquiry was strictly regarding release notifications and security
advisories. If discussions were done in GitHub then it would have been
a GitHub notification and parseable if you opted to receive them.

I'll point out another issue with this as well. This assumes we're just
talking about the "asterisk" repo here, and friends, but the
asterisk-dev list has become the catch-all list for most discussion of
anything development related in the entire Asterisk family of software,
particularly as most of the other lists died a long time ago.

For example, in what repo should discussion of wanpipe take place? Some
of us might want to discuss issues with or trade patches[1], but there
isn't a wanpipe repo since it's not an "open source project". Or general
discussions that might cross over into multiple repos at once, like
something that affects both Asterisk and DAHDI Linux, or both DAHDI
Linux and DAHDI Tools? Should everyone now watch the asterisk-test-suite
repo too? There are a lot of edge cases this doesn't handle well.

I think it's also worth pointing out that, while I'm not one of these
individuals, there are a number of people that don't have a GitHub
account (and perhaps might not want one) that would be excluded if all
discussion was happening there. This very point came out when the
project moved away from Atlassian and there were comments to that effect
*on this list*. These people would have been completely unheard if
discussion had also moved to GitHub prior to that. Do you want to
intentionally exclude them now?

Some people I've noticed also subscribe to the digest version of this
list. I could be wrong but I doubt GitHub discussion has a "digest"
mechanism... because it isn't a real mailing list with all the options
of a real mailing list.

Sometimes people see something on the mailing list and reply privately
to the OP to continue a specific point of discussion off-list. On GitHub
discussions, where everyone is identified by their GitHub usernames and
not real names or email addresses, getting in touch with someone could
be considerably more difficult, particularly for people who might just
be looking at the discussion online.

And frankly, I think expecting 2100 people to reply to this thread is
downright unrealistic. On no mailing list ever does everybody
participate. The majority of mailing lists are dominated by the
discussion of a few while the rest sit back and listen (which is
perfectly fine), maybe 5% of posters generating 95% of the posts. Some
people don't want to contribute, but they do want to read. Nobody has
come out and said he or she wants the mailing list to go away or give
way to another format, and lack of a response is *not* tacit approval of
doing so. All the stakeholders that have spoken out are against the
decision.

I will say though that I have been receiving release announcements both
via the mailing list and via GitHub. For release announcements
specifically, they both work fine. In fact, since the recent 3.3.0 GA
DAHDI Linux release only went to GitHub and not the mailing lists,
that's how I noticed it. I think GitHub is probably just fine for this,
but less so for everything else.

I've already given my opinion before, but I'll reiterate that mailing
lists are accessible to everyone in a way that GitHub never has been and
never will be. I can fire up a terminal email client like mutt or alpine
and make a new post to the list[2][3]. Their website is notorious for
making random changes that break certain browsers and they don't give a
hoot. It's a proprietary platform that we're all at the complete mercy
of. There are already certain things that it's bad at, and there's no
reason to expect it will be better at other things in the future.

[1] This has been happening, but largely on another private mailing
list, not on the asterisk-dev list, though the latter is arguably a more
suitable location for this
[2] And given the audience of this list, I think it's reasonable to
expect that a number of subscribers do this or may want to, at least
occasionally
[3] I'm aware you can respond to a GitHub discussion from email, but you
can't start a discussion via email - see
https://webapps.stackexchange.com/questions/76055/can-i-create-an-issue-in-a-github-repository-by-sending-an-email
This alone is a major access barrier, considering that GitHub no longer
works in any of my preferred browsers, because they have no obligation
to comply with standards. Even though I have a GitHub account, I hate
using the Git

Re: [asterisk-dev] Mailing List Future

2023-12-04 Thread asterisk

I strongly object to not having an asterisk-dev list. Mailing lists are 
essential for FOSS developer discussion. The majority of non-ephemeral 
development discussion happens either on IRC or here on the asterisk-dev 
list - just check the archives to see that it's still active. Most of us 
are not on the community forums and/or couldn't be bothered to use them. 
You can go and see now that "Development" on the community forums is 
basically dead, because nobody wants to use it, so trying to push that 
on everyone is a terrible idea.


Even for users, I think the loss of asterisk-users will be a major loss. 
Far more *discussion* is happening on the Discourse forum, but far more 
*quality* discussion still happens on asterisk-users. Being on a mailing 
list seems to be a natural weedout for junk questions. More serious 
questions still seem to come through on the mailing list. The community 
forums is far fuller of useless postings from people who can't tell a 
hard drive from a memory stick. Nobody wants to wade through a bunch of 
low-quality posts to find the few that might have some use. Thus, 
getting rid of asterisk-users would see a significant drop in the 
average quality of user engagement. But at least, even if the -users 
list is dropped, the -dev list should stick around in some form.


I know the forums can have emails enabled that you can receive, and no, 
that's not a proper replacement for a mailing list.


GitHub Discussions aren't a proper mailing list, either, so ultimately I 
think that will run into the same issue. GitHub has a lot of bells and 
whistles but most of them aren't as built out as using the proper tool 
they try to emulate.


I think #3 is the right choice. It's using the right tool for the right 
job. If you don't want to maintain the lists, have somebody else do it. 
I do a combination of hosted and self-hosted for my own lists. Contrary 
to the opinions of some, people, especially technical people, have not 
"moved on" from mailing lists; they are widely used, and I get hundreds 
of emails a day from them that I have a good workflow for.


Most lists I'm on that used to be elsewhere (e.g. Yahoo Groups, Google 
Groups, mailman, LISTSERV, other custom or independent platforms) have 
now migrated to groups.io and are generally highly satisfied with it 
compared to other platforms. It used to be completely free; it's now 
free for lists under 100 members, or ones that are grandfathered in. As 
the maintainer of several lists there and a member of many more, I've 
been pretty happy with it.


I'd suggest creating a list there and letting people on this list 
manually opt into it, since there are probably a lot of people on 
mailman that aren't active anymore. If it's under 100 members, it's 
completely free anyways. If more than 100 people join, that means people 
here *really* like mailing lists and find value in them, and I'm sure 
Sangoma can afford $20 a month for it, if it really doesn't want to run 
mailman lists anymore that badly, and $20 is a small price to keep 
developers happy.


NA

On 12/4/2023 7:28 AM, Jaco Kroon wrote:


Hi,

My 5c.  Killing the dev list is a bad idea.

Most developers could not care about having to poll forums.  It also 
means that stuff that would previously get an audience will now get none.


github discussions are better than forums at least.

May I inquire as to the problem you're having with the ML? Perhaps I 
might be able to assist ...


Kind regards,
Jaco

On 2023/12/04 14:00, Joshua C. Colp wrote:


Greetings all,

Over the past few years, the use of the Asterisk mailing lists has 
diminished, with far more conversation happening on the Asterisk 
community forums[1]. The state of email, to ensure reliable delivery, 
has also gotten more complicated - emails get caught by spam filters, 
etc.. To continue the mailing lists would require a huge time and 
resource investment, for minimal use.


To that end, we’ve decided to discontinue the mailing lists effective 
February 1st, 2024.


This means the following:

1. Sending and receiving mailing list emails will no longer be possible.
2. The list archives, however, will remain available.

We need to decide the future of the asterisk-dev mailing list; 
specifically, where to hold discussions in the future. There are a 
few options:


1. A “Development” category exists on https://community.asterisk.org/ 
already that can be used.
2. We can use GitHub discussions, which keeps things with the GitHub 
project.

3. We can use a hosted mailing list elsewhere.

We suggest option #2, since it keeps things with the GitHub project, 
which is where everything development-related happens now regardless. 
This has been set up and enabled already.


If you have any input, now is the time to state it.

Cheers,

--
Joshua C. Colp
Asterisk Project Lead
Sangoma Technologies
Check us out at www.sangoma.com <http://www.sangoma.com> and 
www.asterisk.or

[asterisk-dev] DAHDI downloads

2023-10-24 Thread asterisk

I noticed last week that 3.3.0-rc1 for DAHDI Linux and DAHDI Tools are 
available on GitHub, but I don't see them on the downloads server. 
current is still symlinked to 3.2.0, which is more than a year old.


Is DAHDI not using the downloads server anymore? If not, is there any 
kind of permalink available for the latest current tarball?



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[asterisk-dev] asterisk release 21.0.0

2023-10-18 Thread Asterisk Development Team

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

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

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

Thank You!

Change Log for Release asterisk-21.0.0

Links:

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

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

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

Summary:

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

User Notes:

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

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

Upgrade Notes:

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

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

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

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

[asterisk-dev] asterisk release 20.5.0

2023-10-18 Thread Asterisk Development Team

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

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

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

Thank You!

Change Log for Release asterisk-20.5.0

Links:

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

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

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

Summary:

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

User Notes:

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

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

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

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

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

- ### app_queue: Add support for applying caller pr

[asterisk-dev] asterisk release 18.20.0

2023-10-18 Thread Asterisk Development Team

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

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

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

Thank You!

Change Log for Release asterisk-18.20.0

Links:

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

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

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

Summary:

User Notes:

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

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

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

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

- ### app_queue: Add support for applying caller pr

Re: [asterisk-dev] Permalinks including latest RC releases

2023-10-09 Thread asterisk

On 10/9/2023 5:11 AM, Joshua C. Colp wrote:
On Wed, Sep 27, 2023 at 6:31 PM <mailto:aster...@phreaknet.org>> wrote:

On 9/27/2023 5:26 PM, Andrew Latham wrote:
> I would have to look deeper again but my kneejerk was this
sounds like
> "nightly" to me. Just chiming in quickly

Yeah, it has the right connotation, though it might imply that these
builds are put out more frequently than they really are... "monthly"
would be more accurate at that.

I would prefer "testing" as the name. Generally we don't refer to 
things as "stable" or "unstable", and involving dates in any way such 
as "monthly" is inviting people to ask "why hasn't this been updated? 
it's been a month". The release process is in Github and the repo, so 
a PR can be made to add such a thing by anyone. Once done we could 
update the website. I would not advise changing things such as sending 
it to Github for download, the bandwidth from the downloads server 
isn't a problem.

Sounds good, I thought this might have involved more on the backend. I 
submitted a PR to the CI repo in the only place I found any reference to 
the -current suffix, so hopefully that does the trick.
When you say "update the website", are you referring to the downloads 
server or the documentation (which is now also in Git)?

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Re: [asterisk-dev] Permalinks including latest RC releases

2023-09-27 Thread asterisk


On 9/27/2023 5:26 PM, Andrew Latham wrote:
I would have to look deeper again but my kneejerk was this sounds like 
"nightly" to me. Just chiming in quickly


Yeah, it has the right connotation, though it might imply that these 
builds are put out more frequently than they really are... "monthly" 
would be more accurate at that.


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[asterisk-dev] Permalinks including latest RC releases

2023-09-27 Thread asterisk

I brought this up at one point prior to the GitHub migration, and it was 
tabled at the time until post-migration. I was discussing again with 
George in the past couple weeks, but at some point the list got dropped 
out of that so I wanted to bring that back onto the list.


What I'd requested at the time was a link similar to 
asterisk-20-current, for example, on the downloads server, that is 
inclusive of release candidates. -current excludes them, for good 
reason, but a link that included these would make it easier for release 
candidates to be deployed and by extension tested. A while back, Josh 
put out a post encouraging people to try out release candidates[1]. It 
would be significantly easier for people to try out release candidates 
if there were static links that would pull the latest RC if one was 
currently out, and otherwise the latest release as -current currently does.


Recapping George's concern with this was that people might download this 
stable/RC combo not realizing that it might link to a release candidates 
at times. I feel that giving it a sufficiently descriptive name, like 
asterisk-20-unstable or something like should make it clear enough that 
it's probably not what people want unless they're deliberately looking 
for that.


George had suggested a permalink that only linked to RCs when they were 
available. I don't like this idea, because then it's not a universal 
permalink, which was the entire point of my original request. Currently, 
if somebody wants to pull down a release candidate in the script, the 
release candidate name needs to be hardcoded somewhere. Hardcoding that 
a release candidate currently exists, even if the link didn't change, is 
no less problematic and doesn't help very much. It would really need to 
be a single link that works at all times, I think, to be useful, so that 
automated tools don't need to do any thinking and can just pull 
something down.


One major problem with the other proposals is the links are transient, 
so as soon as the official release candidate comes out, the links to the 
release candidates disappear and anything relying on that breaks 
immediately. There are workarounds for this, but all of them end up 
wasting somebody's time on a regular basis.


TL;DR is it would great if there was a link like 
asterisk-20-latest-unstable or something that always pointed to latest 
20 tarball, regardless of whether it's a regular or release candidate 
release. I don't care what the name is personally, as long as it exists, 
so as a sufficiently scary name seems like it should dissuade casual 
browsers from downloading it accidentally, and that's the only concern 
I've seen raised thus far.


With Asterisk 21 rc already out, is this something that could be added 
in some kind of permalink form, to allow for broader testing? It would 
be a win win situation since Sangoma would benefit from more people 
trying release candidates, and this would facilitate that. There are 
many cases where I think if downloading a release candidate were as easy 
and predictable as downloading the -current tarball, it would be done, 
and actually streamline other things in the process.


Another thought: since this would be a new link that didn't exist 
before, rather than symlinking, it could also 302 redirect to the latest 
GitHub tarball (if that were easy to do... I know those links aren't 
predictable). That would probably save a little bandwidth.


Thanks!

[1] https://www.asterisk.org/take-a-look-at-release-candidates/


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[asterisk-dev] asterisk release 21.0.0-rc1

2023-09-06 Thread Asterisk Development Team

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

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

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

Thank You!

Change Log for Release asterisk-21.0.0-rc1

Links:

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

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

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

Summary:

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

User Notes:

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

Upgrade Notes:

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

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

- ### res_monit

[asterisk-dev] asterisk release 20.5.0-rc1

2023-09-06 Thread Asterisk Development Team

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

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

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

Thank You!

Change Log for Release asterisk-20.5.0-rc1

Links:

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

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

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

Summary:

User Notes:

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

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

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

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

- ### app_queue: Add support

[asterisk-dev] asterisk release 18.20.0-rc1

2023-09-06 Thread Asterisk Development Team

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

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

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

Thank You!

Change Log for Release asterisk-18.20.0-rc1

Links:

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

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

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

Summary:

User Notes:

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

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

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

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

- ### app_queue

Re: [asterisk-dev] Pickup peculiarities

2023-09-03 Thread asterisk


On 9/3/2023 5:23 AM, Joshua C. Colp wrote:
A Local channel should work, because there are two of them and the one 
you pick up (;1) should be the one that is not executing dialplan. The 
dialplan one (;2) just has to do a Wait(360) or something.


Yes, that works, and as you described... not sure why it didn't the 
first time I tried that, but I must've done something differently. 
Perfectly suitable and simple workaround, and it even works with 
optimization.


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[asterisk-dev] Pickup peculiarities

2023-09-02 Thread asterisk

Some architectural questions about the current incarnation of the 
builtin call pickup code... at some point I decided it was horribly 
deficient and wrote my own module that I typically use, but I wanted to 
see if I could get something to work for an environment just using 
simple building blocks.


A slightly atypical use case I was toying with was picking up incoming 
calls arriving on FXO ports (chan_dahdi), but that aren't ringing any 
FXS stations. For example, the call could come in on FXO line 1 and ring 
FXS line 1, and the call ringing FXS line 1 could be picked up, but I 
was trying to see if the original call to FXO 1 could be picked up 
without ringing any channels, i.e. pickup without ringing (which is 
maybe a bit contradictory). In the case of an FXO channel, it's already 
executing dialplan (so has a PBX), although it may still be in the 
"ring" state since it hasn't answered yet.


In pickup.c, a channel is only eligible for pickup if there is no PBX 
running on it[1], so this seems to preclude the case above. As such, I 
have a couple questions, just to confirm I'm understanding this right:


 * Semantically, should the above scenario work with the builtin pickup
   functionality, or is it by design that this case is excluded, e.g.
   channels with a PBX but not yet answered (I'm guessing no, since how
   would one distinguish between valid cases such as these, and most
   any other? After all, to the core, it's a channel that's executing
   dialplan)
 * What would be the prototypical "Asterisk way" of doing the above
   scenario? Something like ChannelRedirect() should work, but I mean
   more within the bounds of the pickup construct (and maybe there
   isn't any, just want to confirm I haven't missed something). Put
   another way, how would you do the above, in the simplest way
   possible? (high level, no code necessary)
 * I'm thinking that one way to accomplish this given the way that
   pickup is would be to have some kind of dummy "sink" channel driver,
   e.g. something that can be called, but will never actually answer,
   and can't do anything useful. This should make the above scenario
   function without creating any further additional channels or ringing
   any "real" endpoints. Local channels would not suffice, because they
   begin executing dialplan immediately. The dummy channel driver
   wouldn't do that, or really do anything, it would just be a valid
   target for Dial() that would satisfy the properties expected by
   pickup.c, to allow a channel currently ringing the dummy channel
   driver to be picked up. A toy example:

[from-fxo-port] ; Allow the incoming call from the FXO port to be picked 
up by any station in the same call group for up to 30 seconds, go to 
voicemail otherwise.
exten => s,1,NoOp(Incoming call from ${CALLERID(all)}) ; after 1 ring, 
chan_dahdi spawns PBX execution to handle the FXO port
   same => n,Dial(WaitPickup/group1,30) ; not shown for simplicity, but 
would probably need to use a pre-dial subroutine to execute 
Set(CHANNEL(pickupgroup)=1) on the called channel, or the dummy driver 
would need to accept this and call ast_channel_callgroup_set in its _new 
callback.

   same => n,NoOp(channel was not picked up within 30 seconds)
   same => n,VoiceMail(1234)

Let's ignore exactly how an end user is alerted to the fact there is an 
incoming call on the FXO port; that's not relevant to the situation here 
- for example, suppose there's an external ringer in parallel on the line.


Any thoughts on doing something like this? I'm assuming there isn't such 
a channel driver already (since why would there be?), one would need to 
be written although it'd be fairly simple. Might there be a more elegant 
way of doing this that comes to mind?


[1] https://github.com/asterisk/asterisk/blob/master/main/pickup.c#L79


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Re: [asterisk-dev] issues-archive.asterisk.org is now available for preview

2023-08-27 Thread asterisk

On 8/4/2023 9:48 AM, George Joseph wrote:

> We've done a dump of all the ASTERISK-* issues and their
attachments
> from the issues.asterisk.org <http://issues.asterisk.org>
<http://issues.asterisk.org> Jira
> instance and made them available at
> https://issues-archive.asterisk.org. It'll be a few days before
Google
> gets around to indexing the entire site so the search bar isn't
> working yet but you can browse the issues right now. When the
search
> is fully working we'll announce it on the asterisk-users list as
well.

Something I noticed in the past few days...
issues.asterisk.org used to redirect to issues-archive.asterisk.org, but 
now redirects to GitHub, so issues.asterisk.org links no longer work 
properly. I'm assuming this wasn't intentional - just wanted to make you 
aware of it! Didn't seem like there was a good GitHub repo for this 
metaissue...

I'm thinking maybe it was intended that the root domain 
issues.asterisk.org itself redirect to GitHub, and anything with a path 
redirect to issues-archive, but that's just speculation on my part.

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Re: [asterisk-dev] Gerrit offline?

2023-08-23 Thread asterisk


On 8/23/2023 8:30 AM, George Joseph wrote:

On Tue, Aug 22, 2023 at 6:03 PM  wrote:

On 8/19/2023 11:19 AM, George Joseph wrote:

Here's a gist that does all the work. Create a directory to hold the
patch
files then run
the script from a gerrit asterisk clone directory providing the patch
directory.

Hi George,
 Sorry, just getting to this now. I'm assuming you meant to link a
GitHub gist, but I'm not seeing a link anywhere... didn't find any on
your profile either. Is this available somewhere?

Oops...
https://gist.github.com/gtjoseph/f98d5a583b0d2977686655a56e28ecff

Thanks George!
   Does this run successfully for you? I downloaded a fresh Gerrit repo 
and it seems git doesn't like the fetch/checkout combination used:


root@debian11:/usr/src/gerrit/asterisk# git fetch 
https://gerrit.asterisk.org/asterisk refs/changes/55/17655/25

From https://gerrit.asterisk.org/asterisk
 * branch  refs/changes/55/17655/25 -> FETCH_HEAD
root@debian11:/usr/src/gerrit/asterisk# git checkout -b change-17655 
FETCH HEAD
fatal: Cannot update paths and switch to branch 'change-17655' at the 
same time.


It seems to me like only one or the other is necessary, just want to 
make sure I'm not missing something important. If I do this they all 
have names like 
"patches/19467-pbx_dundi-Fix-PJSIP-endpoint-configuration-check.patch" 
but the actual Gerrit patch number is correct, the same name is used for 
all of them, e.g. below. Same if I swap the effective command. I'm not 
concerned about the name as much, but the patches are actually all 
identical (just the first one).


root@debian11:/usr/src/gerrit/asterisk# ls -la patches
total 140
drwxr-xr-x  2 root root 4096 Aug 23 10:57 .
drwxr-xr-x 33 root root 4096 Aug 23 10:56 ..
-rw-r--r--  1 root root 1047 Aug 23 10:56 
17655-pbx_dundi-Fix-PJSIP-endpoint-configuration-check.patch
-rw-r--r--  1 root root 1047 Aug 23 10:56 
17719-pbx_dundi-Fix-PJSIP-endpoint-configuration-check.patch
-rw-r--r--  1 root root 1047 Aug 23 10:57 
17948-pbx_dundi-Fix-PJSIP-endpoint-configuration-check.patch
-rw-r--r--  1 root root 1047 Aug 23 10:57 
18186-pbx_dundi-Fix-PJSIP-endpoint-configuration-check.patch
-rw-r--r--  1 root root 1047 Aug 23 10:57 
18369-pbx_dundi-Fix-PJSIP-endpoint-configuration-check.patch
-rw-r--r--  1 root root 1047 Aug 23 10:57 
18574-pbx_dundi-Fix-PJSIP-endpoint-configuration-check.patch
-rw-r--r--  1 root root 1047 Aug 23 10:56 
18577-pbx_dundi-Fix-PJSIP-endpoint-configuration-check.patch
-rw-r--r--  1 root root 1047 Aug 23 10:56 
18829-pbx_dundi-Fix-PJSIP-endpoint-configuration-check.patch
-rw-r--r--  1 root root 1047 Aug 23 10:57 
19211-pbx_dundi-Fix-PJSIP-endpoint-configuration-check.patch
-rw-r--r--  1 root root 1047 Aug 23 10:56 
19264-pbx_dundi-Fix-PJSIP-endpoint-configuration-check.patch
-rw-r--r--  1 root root 1047 Aug 23 10:57 
19447-pbx_dundi-Fix-PJSIP-endpoint-configuration-check.patch
-rw-r--r--  1 root root 1047 Aug 23 10:57 
19467-pbx_dundi-Fix-PJSIP-endpoint-configuration-check.patch
-rw-r--r--  1 root root 1047 Aug 23 10:57 
19534-pbx_dundi-Fix-PJSIP-endpoint-configuration-check.patch
-rw-r--r--  1 root root 1047 Aug 23 10:57 
19572-pbx_dundi-Fix-PJSIP-endpoint-configuration-check.patch
-rw-r--r--  1 root root 1047 Aug 23 10:57 
19655-pbx_dundi-Fix-PJSIP-endpoint-configuration-check.patch
-rw-r--r--  1 root root 1047 Aug 23 10:56 
19718-pbx_dundi-Fix-PJSIP-endpoint-configuration-check.patch
-rw-r--r--  1 root root 1047 Aug 23 10:56 
19740-pbx_dundi-Fix-PJSIP-endpoint-configuration-check.patch
-rw-r--r--  1 root root 1047 Aug 23 10:56 
19741-pbx_dundi-Fix-PJSIP-endpoint-configuration-check.patch
-rw-r--r--  1 root root 1047 Aug 23 10:56 
19748-pbx_dundi-Fix-PJSIP-endpoint-configuration-check.patch
-rw-r--r--  1 root root 1047 Aug 23 10:56 
19749-pbx_dundi-Fix-PJSIP-endpoint-configuration-check.patch
-rw-r--r--  1 root root 1047 Aug 23 10:56 
19793-pbx_dundi-Fix-PJSIP-endpoint-configuration-check.patch
-rw-r--r--  1 root root 1047 Aug 23 10:56 
19797-pbx_dundi-Fix-PJSIP-endpoint-configuration-check.patch
-rw-r--r--  1 root root 1047 Aug 23 10:56 
19921-pbx_dundi-Fix-PJSIP-endpoint-configuration-check.patch
-rw-r--r--  1 root root 1047 Aug 23 10:56 
19979-pbx_dundi-Fix-PJSIP-endpoint-configuration-check.patch
-rw-r--r--  1 root root 1047 Aug 23 10:56 
20033-pbx_dundi-Fix-PJSIP-endpoint-configuration-check.patch
-rw-r--r--  1 root root 1047 Aug 23 10:56 
20037-pbx_dundi-Fix-PJSIP-endpoint-configuration-check.patch
-rw-r--r--  1 root root 1047 Aug 23 10:56 
20038-pbx_dundi-Fix-PJSIP-endpoint-configuration-check.patch
-rw-r--r--  1 root root 1047 Aug 23 10:56 
20058-pbx_dundi-Fix-PJSIP-endpoint-configuration-check.patch
-rw-r--r--  1 root root 1047 Aug 23 10:56 
20059-pbx_dundi-Fix-PJSIP-endpoint-configuration-check.patch
-rw-r--r--  1 root root 1047 Aug 23 10:56 
20068-pbx_dundi-Fix-PJSIP-endpoint-configuration-check.patch
-rw-r--r--  1 root root 1047 Aug 23 10:56 
20069-pbx_dundi-Fix-PJSIP-endpoint-configuration-check.patch
-rw-r

Re: [asterisk-dev] Gerrit offline?

2023-08-22 Thread asterisk


On 8/19/2023 11:19 AM, George Joseph wrote:
Here's a gist that does all the work. Create a directory to hold the 
patch

files then run
the script from a gerrit asterisk clone directory providing the patch
directory.


Hi George,
   Sorry, just getting to this now. I'm assuming you meant to link a 
GitHub gist, but I'm not seeing a link anywhere... didn't find any on 
your profile either. Is this available somewhere?


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Re: [asterisk-dev] Gerrit offline?

2023-08-18 Thread asterisk


On 8/18/2023 9:15 AM, Sean Bright wrote:

On 8/17/2023 9:04 PM, aster...@phreaknet.org wrote:
Would it be at all possible to extend that possibly at least a couple 
days, perhaps through Wednesday at least?


Shouldn't be necessary, I opened two PRs in your repo that remove the 
references to gerrit so you should be good to go.


Thanks Sean, that helped a lot!

On 8/18/2023 9:51 AM, George Joseph wrote:

I  can leave it up until Wednesday 1900Z.


Thanks George, appreciate the flexibility!

Just what you have.   You can pull them down yourself easily with the 
following...


curl -s 'https://gerrit.asterisk.org/changes/ 
<https://gerrit.asterisk.org/changes/%5C>?q=is:open=CURRENT_REVISION=DOWNLOAD_COMMANDS' 
\

    | tail -n +2 | jq -r '.[] | .revisions[].fetch[].commands.Branch' \
> /tmp/get_reviews.sh  ; chmod a+x /tmp/get_reviews.sh ; 
/tmp/get_reviews.sh




I guess the idea here is you checkout each change, and then can export 
the diff into a file or something like that? I guess it should be 
straightforward to generate the patches from that, and then discard the 
repo, thanks for the code.
Sean's work simplified this a good bit, but there are a few I may go and 
manually track down for safe keeping over the weekend (mostly some of 
Mark's stuff that hasn't been migrated to GitHub yet... or may not be 
soon, not sure what the status is on those).


git fetch https://gerrit.asterisk.org/asterisk refs/changes/55/17655/25 
&& git checkout -b change-17655 FETCH_HEAD
git fetch https://gerrit.asterisk.org/asterisk refs/changes/79/19979/3 
&& git checkout -b change-19979 FETCH_HEAD
git fetch https://gerrit.asterisk.org/asterisk refs/changes/69/20069/1 
&& git checkout -b change-20069 FETCH_HEAD
git fetch https://gerrit.asterisk.org/asterisk refs/changes/38/20038/3 
&& git checkout -b change-20038 FETCH_HEAD
git fetch https://gerrit.asterisk.org/asterisk refs/changes/68/20068/1 
&& git checkout -b change-20068 FETCH_HEAD
git fetch https://gerrit.asterisk.org/asterisk refs/changes/58/20058/1 
&& git checkout -b change-20058 FETCH_HEAD
git fetch https://gerrit.asterisk.org/asterisk refs/changes/59/20059/1 
&& git checkout -b change-20059 FETCH_HEAD
git fetch https://gerrit.asterisk.org/asterisk refs/changes/33/20033/3 
&& git checkout -b change-20033 FETCH_HEAD
git fetch https://gerrit.asterisk.org/asterisk refs/changes/40/19740/3 
&& git checkout -b change-19740 FETCH_HEAD
git fetch https://gerrit.asterisk.org/testsuite refs/changes/83/20083/1 
&& git checkout -b change-20083 FETCH_HEAD
git fetch https://gerrit.asterisk.org/asterisk refs/changes/80/20080/1 
&& git checkout -b change-20080 FETCH_HEAD
git fetch https://gerrit.asterisk.org/asterisk refs/changes/19/17719/11 
&& git checkout -b change-17719 FETCH_HEAD
git fetch https://gerrit.asterisk.org/asterisk refs/changes/37/20037/2 
&& git checkout -b change-20037 FETCH_HEAD
git fetch https://gerrit.asterisk.org/asterisk refs/changes/41/19741/15 
&& git checkout -b change-19741 FETCH_HEAD
git fetch https://gerrit.asterisk.org/asterisk refs/changes/97/19797/3 
&& git checkout -b change-19797 FETCH_HEAD
git fetch https://gerrit.asterisk.org/asterisk refs/changes/21/19921/16 
&& git checkout -b change-19921 FETCH_HEAD
git fetch https://gerrit.asterisk.org/asterisk refs/changes/93/19793/7 
&& git checkout -b change-19793 FETCH_HEAD
git fetch https://gerrit.asterisk.org/asterisk refs/changes/29/18829/25 
&& git checkout -b change-18829 FETCH_HEAD
git fetch https://gerrit.asterisk.org/asterisk refs/changes/77/18577/4 
&& git checkout -b change-18577 FETCH_HEAD
git fetch https://gerrit.asterisk.org/asterisk refs/changes/64/19264/9 
&& git checkout -b change-19264 FETCH_HEAD
git fetch https://gerrit.asterisk.org/asterisk refs/changes/18/19718/4 
&& git checkout -b change-19718 FETCH_HEAD
git fetch https://gerrit.asterisk.org/asterisk refs/changes/49/19749/3 
&& git checkout -b change-19749 FETCH_HEAD
git fetch https://gerrit.asterisk.org/asterisk refs/changes/48/19748/1 
&& git checkout -b change-19748 FETCH_HEAD
git fetch https://gerrit.asterisk.org/asterisk refs/changes/67/19467/1 
&& git checkout -b change-19467 FETCH_HEAD
git fetch https://gerrit.asterisk.org/asterisk refs/changes/55/19655/1 
&& git checkout -b change-19655 FETCH_HEAD
git fetch https://gerrit.asterisk.org/asterisk refs/changes/72/19572/1 
&& git checkout -b change-19572 FETCH_HEAD
git fetch https://gerrit.asterisk.org/asterisk refs/changes/11/19211/3 
&& git checkout -b change-19211 FETCH_HEAD
git fetch https://gerrit.asterisk.org/asterisk refs/changes/34/19534/6 
&& git checkout -b change-19534 FETCH_HEAD
git fetch https://gerrit.asterisk.org/asterisk refs/changes/48/17948/5 
&& git checkout -b change-17948 FETCH_HEAD
git fetch https://gerrit.asterisk.o

Re: [asterisk-dev] Gerrit offline?

2023-08-17 Thread asterisk


On 8/17/2023 8:09 AM, George Joseph wrote:
On Wed, Aug 16, 2023 at 5:58 PM George Joseph <mailto:gjos...@sangoma.com>> wrote:


I'll bring it back up in the morning.


Gerrit is back up but will be permanently disabled on Monday at 1200Z.


Thanks George,
  Would it be at all possible to extend that possibly at least a couple 
days, perhaps through Wednesday at least?
I'm going to be out of the office and on the road a lot into early next 
week, I don't think I'm going to be able to get much done migration-wise 
by then. Wednesday at least I think I can plan to have things fully 
migrated. It's just a lot that's catching us off guard here without any 
advance warning beforehand. I think through Wednesday at least provides 
a good window for folks.
Additionally, is there any kind of archive of any of the stuff on Gerrit 
that is publicly accessible, or just what people have or will manually 
migrate by such time?

Thanks!

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[asterisk-dev] Gerrit offline?

2023-08-16 Thread asterisk

It seems like some time in the past day or so, Gerrit has gone offline, 
which is causing build failures and other issues since some patches are 
inaccessible.


Is this just temporary? I don't recall any announcement going out that 
Gerrit would be going offline imminently. The communication earlier this 
year was that it would remain online for some time and there would be 
communication ahead of any changes to allow people to prepare for this. 
Has there been any change with this?



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[asterisk-dev] libpri release 1.6.1

2023-08-16 Thread Asterisk Development Team

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

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

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

Thank You!


Change Log for Release libpri-1.6.1


Links:


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

Summary:


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

User Notes:



Upgrade Notes:



Closed Issues:


None

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Re: [asterisk-dev] [External] Re: Final Preview: docs.asterisk.org

2023-08-10 Thread asterisk

On 8/9/2023 6:43 PM, George Joseph wrote:
On Wed, Aug 9, 2023 at 4:05 PM George Joseph <mailto:gjos...@sangoma.com>> wrote:

On Wed, Aug 9, 2023 at 2:30 PM mailto:aster...@phreaknet.org>> wrote:

On 8/9/2023 11:12 AM, George Joseph wrote:

> Yeah, create an issue.  I can take a look in the coming
weeks.  If you
> constrict the width of your browser, at some point, the left
nav bar
> will collapse and you can get it back by clicking on the
"hamburger"
> button that then appears in the top-left of the page. 
There's no way
> to collapse it manually though so maybe we can find a way to
add that.
> Maybe we can also make the page table of contents
collapsible.  Both
> should give more space to the content.  I think we can also
override
> the viewport width of the content.    A tweak to the dynamic
> documentation generator might also help.

I don't think the issue here is collapsing the navigation. In
fact, I
really hate when you're on a large monitor and websites
collapse menus
like that, catering to mobile devices only is pure insanity,
making life
more difficult for everyone else by requiring yet more clicks
to do
anything. The issue is that the site seems to max out at a
certain
viewport; on a large monitor, the middle portion could take up
more
room, but there is vast whitespace to the left and right
margins. It's
possible that the style is assuming a max-width that it will
use for
presentation. Ideally, the middle content should expand to
take up the
space it can so it can use the full width of any monitor.

Yeah I get it.  I was just throwing out some additional ideas as well.

Simple change.  Check now.  You may have to clear your cache or at 
least look at a page that's not currently in your cache.

It's much better now. Takes up the full screen on a large monitor, on 
several different types of pages. Thanks, George!

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Re: [asterisk-dev] [External] Re: Final Preview: docs.asterisk.org

2023-08-09 Thread asterisk


On 8/9/2023 11:12 AM, George Joseph wrote:



On Wed, Aug 9, 2023 at 8:39 AM Joshua C. Colp <mailto:jc...@sangoma.com>> wrote:


On Wed, Aug 9, 2023 at 11:37 AM Floimair Florian
mailto:f.floim...@commend.com>> wrote:

Thanks Josh!

I went the same path actually but gave up, as CSS to me is
something completely out of my knowledge domain.

I also had a look at the other teams but so far Material for
mkdocs does still look like the best option out there readily
available.


Then I'd suggest filing an issue on the Github repo with your
comments so they don't get lost. No guarantee anything can be
done, but the docs repo is where issues should go.


Yeah, create an issue.  I can take a look in the coming weeks.  If you 
constrict the width of your browser, at some point, the left nav bar 
will collapse and you can get it back by clicking on the "hamburger" 
button that then appears in the top-left of the page.  There's no way 
to collapse it manually though so maybe we can find a way to add that. 
Maybe we can also make the page table of contents collapsible.  Both 
should give more space to the content.  I think we can also override 
the viewport width of the content.    A tweak to the dynamic 
documentation generator might also help.


I don't think the issue here is collapsing the navigation. In fact, I 
really hate when you're on a large monitor and websites collapse menus 
like that, catering to mobile devices only is pure insanity, making life 
more difficult for everyone else by requiring yet more clicks to do 
anything. The issue is that the site seems to max out at a certain 
viewport; on a large monitor, the middle portion could take up more 
room, but there is vast whitespace to the left and right margins. It's 
possible that the style is assuming a max-width that it will use for 
presentation. Ideally, the middle content should expand to take up the 
space it can so it can use the full width of any monitor.


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Re: [asterisk-dev] Final Preview: docs.asterisk.org

2023-08-04 Thread asterisk


On 8/4/2023 3:20 PM, George Joseph wrote:
On Fri, Aug 4, 2023 at 11:50 AM <mailto:aster...@phreaknet.org>> wrote:


Only thing I noticed when building this time around was warnings
like these:
INFO    -  Doc file

'Asterisk_20_Documentation/API_Documentation/Dialplan_Functions/SMDI_MSG_RETRIEVE.md'

contains an absolute link
'/Asterisk_20_Documentation/API_Documentation/Dialplan_Functions/SMDI_MSG',

it was left as is. Did you mean 'SMDI_MSG.md'?
INFO    -  Doc file

'Asterisk_20_Documentation/API_Documentation/Dialplan_Functions/STREAM_SILENCE.md'

contains an absolute link

'/Asterisk_20_Documentation/API_Documentation/Dialplan_Applications/ChanSpy',

it was left as is. Did you mean
    '../Dialplan_Applications/ChanSpy.md'?


Since I don't get those errors I assume they're your own added 
documentation?  I can't really help there.  All I can say is to look 
at how the standard documentation references other pages.  To make 
sure the links work in different versions of asterisk, they need to be 
relative, like the last error message above.


No, I was seeing this for everything (e.g. ChanSpy and SMDI_MSG_RETRIEVE 
above). But if you're not getting them, then maybe I (probably) missed 
something. Since everything worked regardless, I don't think we need to 
worry about this then. If I notice an actual issue down the line, I'll 
look into that and flag if needed, but right now all's good.


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Re: [asterisk-dev] Final Preview: docs.asterisk.org

2023-08-04 Thread asterisk


On 8/4/2023 9:42 AM, George Joseph wrote:
Well, I've made a few more changes and pushed them up.  I think this 
is as good as it's going to get for now.


I think it's perfect. Down from 230 MB to 140 MB for the same build. 40% 
size reduction just by removing whitespace I guess! Looking at a few 
pages manually, the HTML looks perfect. No visible issues.


Only thing I noticed when building this time around was warnings like these:
INFO    -  Doc file 
'Asterisk_20_Documentation/API_Documentation/Dialplan_Functions/SMDI_MSG_RETRIEVE.md' 
contains an absolute link
'/Asterisk_20_Documentation/API_Documentation/Dialplan_Functions/SMDI_MSG', 
it was left as is. Did you mean 'SMDI_MSG.md'?
INFO    -  Doc file 
'Asterisk_20_Documentation/API_Documentation/Dialplan_Functions/STREAM_SILENCE.md' 
contains an absolute link
'/Asterisk_20_Documentation/API_Documentation/Dialplan_Applications/ChanSpy', 
it was left as is. Did you mean

   '../Dialplan_Applications/ChanSpy.md'?

However I tested the site and things seem to work fine. The build did 
take longer, possibly due to the above checks, 90 seconds vs 18, but 
that's not really an issue. The links do appear to be relative to me - 
I'm not putting this in a domain root, but in a subfolder, and the links 
all seem to work correctly for me. So I don't think there's an issue and 
it seems like this can be ignored - perhaps it went ahead and converted 
it on the fly. Just wanted to point that out in case I'm wrong.


Everything seems to work well, I don't see any further issues with 
anything here. Thanks a lot George for looking into these issues, I'm 
looking forward to porting documentation over to this new generation method.


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Re: [asterisk-dev] issues-archive.asterisk.org is now available for preview

2023-08-04 Thread asterisk


On 8/4/2023 9:48 AM, George Joseph wrote:
We've done a dump of all the ASTERISK-* issues and their attachments 
from the issues.asterisk.org <http://issues.asterisk.org> Jira 
instance and made them available at 
https://issues-archive.asterisk.org. It'll be a few days before Google 
gets around to indexing the entire site so the search bar isn't 
working yet but you can browse the issues right now.  When the search 
is fully working we'll announce it on the asterisk-users list as well.


Take a look.
Looks good on the surface to me. My main concern was patch attachments 
being preserved and it looks like they are.


Just curious, how large is the export exactly? Certain things in JIRA 
were helpful like being able to filter issues by a keyword to "open", 
i.e. things which had and have not yet been resolved. The Google Search 
will help with indexing content but not with filtering, I'd think. If 
it's not too large, do you think this would be better suited to allowing 
folks to download an archive and then be able to use tools like grep to 
find the issues they want?


I don't know that the hosted archives needs to do this, I suspect very 
few people would have any need for being able to do that, and I don't 
want to add any work for anyone here.


Side question, more on the legal side. When everything was on JIRA, I 
think the policy was that any patches on JIRA could be taken through 
code review by anyone else, so long as the uploader had signed the CLA. 
Now that it is on GitHub, and there is a new CLA, and most people have 
not resigned the CLA for patches from the past ~20 years, how does this 
affect patches in the JIRA archive? Since the CLA was valid when they 
were provided to the project, can they still be taken through code 
review by anyone else? What is the status of such patches? Thanks.


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Re: [asterisk-dev] Final Preview: docs.asterisk.org

2023-07-28 Thread asterisk

On 7/28/2023 8:48 PM, George Joseph wrote:
On Fri, Jul 28, 2023 at 1:52 PM <mailto:aster...@phreaknet.org>> wrote:

It's at the very bottom of the README:
> If you're always going to build just 1 branch's dynamic
documentation,
> you can skip the Makefile..inc file and just place everything in
the
> main Makefile.inc:

The first Makefile..inc has an extra period world's least
important
typo.

Ahha! It's 'Makefile..inc' in the source README.md but the
'' is getting stripped. :)

Ah, probably should've noticed that, actually, Makefile.inc twice in a
row doesn't make any sense, if I was actually thinking about it...

Circling back to one other thing now that this seems good to go, what
exactly did you change for reducing the file sizes / is that
included in
the current iteration, without mike?

The addition of "navigation.prune" under features in mkdocs.yml did
most of it, and yes, it's currently included.

I'm still seeing a lot of
extraneous whitespace in the pages. 244 KB per page isn't huge,
but just
at a cursory glance,

Can you give me an example of an html page that's that big? Most I
see are in the 80-100k range

I think all of them - for example:
https://docs.asterisk.org/Asterisk_20_Documentation/API_Documentation/Dialplan_Applications/ADSIProg/

This one is "only" 131 KB, but if you go and view source, you can scroll
down a bit and see often hundreds of newlines, tabs, and spaces at a
time in a row. I can't work how that's creeping in from the markdown, so
I don't think it's from the markdown. That's why I thought we might need
to do it manually, e.g. using tr or something like that. So regardless
of page size, I think we could likely prune all the pages down just by
eliminating whitespace.

I think we could probably cut the size 10-20% just
by getting rid of the whitespace. In some places, there are just
hundreds of newlines in a row for no reason.

Give me an example page.

If this is just what the
tool generates, I understand that, we don't have any control over
that,
just wanted to know. We could remove it all pretty easily with sed
probably, and think could be a final "post processing" step in the
Makefile, run recursively on all files. What do you think?

I tried to do exactly that but it didn't result in much savings and I
got nervous about accidentally deleting multiple "blank" lines without
knowing whether you might be in a "" block or not. I was going
to try html-minifier but just haven't got to it yet.

Yeah, I guess that could be tricky. But how much is the tag
actually used? On the page linked above, for example, I only see it
once, and, ironically, there isn't any extraneous whitespace in it. I
took a look at a few different types of pages and this appears to be the
case for all of them. So in our particular case, it seems like it might
be okay to do a simple delete, since shouldn't be affected by
consecutive whitespace.

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Re: [asterisk-dev] Final Preview: docs.asterisk.org

2023-07-28 Thread asterisk

On 7/24/2023 8:27 AM, George Joseph wrote:

Only one at the time... I just found an extra period in
Makefile..inc in
the latest README, but haven't noticed anything else.

Not seeing it.  Probably fixed already.

It's at the very bottom of the README:
If you're always going to build just 1 branch's dynamic documentation, 
you can skip the Makefile..inc file and just place everything in the 
main Makefile.inc:

The first Makefile..inc has an extra period world's least important 
typo.

> Makefile.inc:
> ASTERISK_XML_FILE := /core-en_US.xml
> SKIP_ARI := yes
> BRANCHES := 20

Got it - this makes a lot more sense now! And yes, you read my
mind, I
don't care about ARI so that did the trick. I noticed the no-mike
branch
no longer exists, but looks like it was merged into main now, so I
gave
that a go and it got much further (sorry, I know it's been a while
and I
wasn't able to test this quickly).

Couldn't have asked for an easier process! It seems like I can just
clone the repo, copy my Makefile.inc in there, and run make. The
above
was on a relatively fast CPU, but it seems it shouldn't take
longer than
maybe 2 minutes.

The result is a 1.6 GB directory, 

Eh what?  When I build everything, temp/site is only 574M.  Maybe need 
to clean everything out or is your own stuff just that big?

No, I probably screwed it up somehow; "make clean" didn't remove any of 
the generated files, but it didn't give me a target error so I just 
assumed that would do what it needed to do. This time I explicitly did 
an rm -rf temp/site beforehand to ensure it would be clean.

 From what I tried initially, I should be able to solve this by
deleting
everything in the docs directory except index.md and the favicon,
which
ensures that there simply is no static content to build. That should
bring down both the size and the build time. I don't mind doing
that at
all, just wondering is there a good way to not build the static
content,
or would that be the best way?

Do a git pull :)
You should now be able to do...
"make BRANCH=master NO_STATIC=yes"

You can add NO_STATIC=yes to your makefile.inc instead.

Thanks George! This looks much more promising:

root@debian11:/usr/src/documentation# rm -rf temp/site/
root@debian11:/usr/src/documentation# make BRANCH=20
Creating ./temp/build-20
Setting Up Core Dynamic Documentation for branch '20'
  Generating markdown from Asterisk XML
Building to ./temp/site
INFO -  Cleaning site directory
INFO -  Building documentation to directory: 
/usr/src/documentation/temp/site

INFO -  Documentation built in 15.67 seconds

Now it's only 230 MB in total. The site builds quickly and it's exactly 
what I was looking for when I opened it up - perfect!

Circling back to one other thing now that this seems good to go, what 
exactly did you change for reducing the file sizes / is that included in 
the current iteration, without mike? I'm still seeing a lot of 
extraneous whitespace in the pages. 244 KB per page isn't huge, but just 
at a cursory glance, I think we could probably cut the size 10-20% just 
by getting rid of the whitespace. In some places, there are just 
hundreds of newlines in a row for no reason. If this is just what the 
tool generates, I understand that, we don't have any control over that, 
just wanted to know. We could remove it all pretty easily with sed 
probably, and think could be a final "post processing" step in the 
Makefile, run recursively on all files. What do you think?

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Re: [asterisk-dev] Final Preview: docs.asterisk.org

2023-07-23 Thread asterisk


On 7/12/2023 8:27 AM, George Joseph wrote:
On Tue, Jul 11, 2023 at 3:28 PM <mailto:aster...@phreaknet.org>> wrote:


How are the includes *supposed* to be handled, by the way i.e. what's
supposed to dereference the xincludes? Is it one of the Asterisk
build
scripts for the docs piecing everything together, or is it
expected that
whatever consumes the XML files is able to handle those?


XML processing is kinda spread out all over
* build_tools/get_documentation.py
* build_tools/make_xml_documentation
* loader.c:load_modules().

TBH, I haven't looked at that stuff in years.  It could probably be 
simplified quite a bit.


Thanks, I'll play with these further and see if I can solve the include 
omission I had before. Simplification aside, I think this type of stuff 
has been historically underdocumented, so maybe we could have a wiki 
page delving into these? If I come up with any useful notes, I could 
share there as well.



Only one??   Fixed.


Only one at the time... I just found an extra period in Makefile..inc in 
the latest README, but haven't noticed anything else.






root@debian11:/usr/src/documentation# cat Makefile*.inc
ASTERISK_XML_FILE := /usr/src/asterisk-20.3.0/doc/core-en_US.xml
BRANCHES := 20
ASTERISK_XML_FILE := /usr/src/asterisk-20.3.0/doc/core-en_US.xml


You're specifying ASTERISK_XML_FILE but not ASTERISK_ARI_DIR so the 
process is still going to try to use 'gh' to download the 
documentation source for ARI.  I just added the SKIP_ARI variable 
which can be set in any of the Makefile.inc files to skip processing 
of the ARI documentation altogether.  So your Makefile.inc could now 
look like this:


Makefile.inc:
ASTERISK_XML_FILE := /core-en_US.xml
SKIP_ARI := yes
BRANCHES := 20


Got it - this makes a lot more sense now! And yes, you read my mind, I 
don't care about ARI so that did the trick. I noticed the no-mike branch 
no longer exists, but looks like it was merged into main now, so I gave 
that a go and it got much further (sorry, I know it's been a while and I 
wasn't able to test this quickly).


Couldn't have asked for an easier process! It seems like I can just 
clone the repo, copy my Makefile.inc in there, and run make. The above 
was on a relatively fast CPU, but it seems it shouldn't take longer than 
maybe 2 minutes.


The result is a 1.6 GB directory, but it looks like there are 555M for 
latest_api and 511M for Asterisk 20. I guess I really only need "latest" 
(which I'm assuming is master) for the purposes of an application 
reference, since that should be a superset of everything (except stuff 
that's been removed obviously, which I don't care about).


It's also still generating the static docs, not just the dynamic docs, 
which is most of the other space. It looks like from the latest README, 
I can just use Makefile directly instead of Makefile.inc since I only 
need 1 branch, although I kind of like the Makefile.inc now to keep my 
stuff separate from the rest of the repo, and it doesn't look like that 
should make a difference. But if I do "make BRANCH=master" (with 
Makefile.20.inc duplicated to Makefile.master.inc), it doesn't seem to work:


root@debian11:/usr/src/documentation# make BRANCH=master
Creating ./temp/build-master
Setting Up Core Dynamic Documentation for branch 'master'
  Generating markdown from Asterisk XML
ln: failed to create symbolic link 
'./temp/docs/Asterisk_master_Documentation/API_Documentation': No such 
file or directory

make: *** [Makefile:105: dynamic-core-setup] Error 1
root@debian11:/usr/src/documentation# make BRANCH=20
Creating ./temp/build-20
Setting Up Core Dynamic Documentation for branch '20'
  Generating markdown from Asterisk XML
Building to ./temp/site
INFO -  Cleaning site directory
INFO -  Building documentation to directory: 
/usr/src/documentation/temp/site

INFO -  Documentation built in 85.57 seconds

From what I tried initially, I should be able to solve this by deleting 
everything in the docs directory except index.md and the favicon, which 
ensures that there simply is no static content to build. That should 
bring down both the size and the build time. I don't mind doing that at 
all, just wondering is there a good way to not build the static content, 
or would that be the best way?


This is already great by the way, for what I need to it to do - none of 
this is super important though, but if you have any thoughts I'll give 
it another go and see if I can get a more optimized build.


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[asterisk-dev] Asterisk Release 18.19.0

2023-07-20 Thread Asterisk Development Team

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

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

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

Thank You!

Change Log for Release 18.19.0

Links:

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

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

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

Summary:

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

User Notes:

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

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

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

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

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

[asterisk-dev] Asterisk Release 20.4.0

2023-07-20 Thread Asterisk Development Team

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

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

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

Thank You!

Change Log for Release 20.4.0

Links:

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

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

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

Summary:

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

User Notes:

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

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

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

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

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

[asterisk-dev] Asterisk Release 20.4.0-rc2

2023-07-13 Thread Asterisk Development Team

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

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

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

Thank You!


Change Log for Release 20.4.0-rc2


Links:


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

Summary:


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

User Notes:



Upgrade Notes:



Closed Issues:


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

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[asterisk-dev] Asterisk Release 18.19.0-rc2

2023-07-13 Thread Asterisk Development Team

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

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

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

Thank You!


Change Log for Release 18.19.0-rc2


Links:


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

Summary:


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

User Notes:



Upgrade Notes:



Closed Issues:


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

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Re: [asterisk-dev] Final Preview: docs.asterisk.org

2023-07-11 Thread asterisk


On 7/11/2023 4:34 PM, George Joseph wrote:
On Sat, Jul 8, 2023 at 4:24 PM <mailto:aster...@phreaknet.org>> wrote:


Hi, Geroge,
 Just had a chance to look at this this afternoon. The
instructions
for the dynamic doc generation definitely made my head hurt a little
bit, but I have a few thoughts after putzing around a little bit.


Your head should be much better now. :)


Much better... the A/C repairman just left too, which helps :)



My initial thought was that some of the make targets in the Makefile
could be split up a little bit. The version-dynamic target both
downloads documentation source and does the actual build of the
documentation. They could be split into different targets:

  * In my case, I don't want to download the upstream Asterisk
    documentation, I want to use the local core-en-us.xml, which is a
    superset of the documentation available upstream.


Uhm are you sure??   The core XML document generated during a build 
doesn't have the xinclude references de-referenced.  I'd be interested 
to know what you're seeing.


Mm... yes, you're right about that. I remember encountering that 
limitation when I wrote my converter. I planned to do something about 
it, but never did :)
My core-en_US.xml and full-en_US.xml files are very similar in size so I 
don't think which one I use would make any difference with this.


As far as "what I'm seeing", this is an example of what I get when I 
take my built XML file at the end of compiling and run it through my 
conversion script: https://asterisk.phreaknet.org/


I believe it handles most of the documentation except those 
xinclude/pointer references to other files. It's not fully on par with 
the real docs, but my main use case here was being able to view 
documentation for applications and functions that aren't upstreamed, and 
for that it's sufficed. Also, the parser I wrote is not super robust, an 
unintended benefit of which is occasionally it's caught bugs in the 
xmldocs that have needed to be fixed, that otherwise might have slipped 
through a more lenient parser.


I could keep using this, of course, since nothing has changed with the 
XML and there's no Confluence involved, but what you've done with the 
dynamic docs is way better (and the search capability is really nice), 
so I'd like to migrate to that for internal documentation.


How are the includes *supposed* to be handled, by the way i.e. what's 
supposed to dereference the xincludes? Is it one of the Asterisk build 
scripts for the docs piecing everything together, or is it expected that 
whatever consumes the XML files is able to handle those?


There are now facilities in the Makefile to get the dynamic sources 
from your

local system.   In that case, gh is not required. Details in the README.


Found a typo in the new README, seems to be just on that branch so I 
guess I can't submit a PR: "CreteDocs"




Then again, when you boil it down to that, it seems like it really
comes
down to just:

  * utils/astxml2markdown.py
  * mike deploy

So it seems I'm better off avoiding the makefile and just running the
individual commands needed. I'll end up wrapping it in a script
anyways,
but just wondering if there's anything else about the process here
I've
missed that wouldn't be conducive to that?


You don't need to avoid the Makefile.  You can specify where to get 
the dynamic sources from and which branches to build.  Details in the 
README.


I'm a little confused about the BRANCHES variable in Makefile.inc. I 
presume this is irrelevant if using a local XML doc? If only because 
there's only one version of Asterisk compiled (and from which the XML 
doc is sourced). It does specifically say if I'm using local sources I 
need separate Makefile.branch.inc's though so I think I'm not 
understanding something here.


I think I understand why the branches are needed - because now it's 
building all the docs for all the versions into the same site. I think 
my point of confusion is the dynamic doc builder only sees an XML file, 
it can't even possibly know what version of Asterisk that's for without 
being told, can it? I was thinking that make BRANCHES=XX just needs to 
match Makefile.XX.inc, but wasn't sure if there's any other significance 
to these values.


It seems to still be trying to download it for me despite adding the 
includes, though I just set up the includes and followed the 
instructions, haven't dug much deeper


root@debian11:/usr/src/documentation# nano Makefile.inc
root@debian11:/usr/src/documentation# nano Makefile.20.inc
root@debian11:/usr/src/documentation# make BRANCHES=20
Finding last CreateDocs job
/usr/bin/bash: line 1: gh: command not found
Setting Up Static Documentation
  Copying to temp build
  Applying link transformations
Building branch 20
Finding last CreateDocs job
/usr/bin/bash: line 1: gh: command not found
Creating ./

Re: [asterisk-dev] Final Preview: docs.asterisk.org

2023-07-10 Thread asterisk


On 7/10/2023 10:09 AM, George Joseph wrote:

I'm digesting this.  It may be a while. :)


Sorry, didn't mean to dump a novel on you there. Let me know if there's 
anything I should clarify.


On another note, after a couple months of being in the GitHub workflow, 
here's another suggestion for improvement, if it's an easy change to make:


The workflow that reminds people to post a cherry pick comment is 
obviously helpful. However, I've noticed that this runs on issues, 
regardless of whether or not a comment already exists on the issue for 
cherry picks. I usually post that comment before the bot kicks in, but 
it comments nonetheless. GitHub sends a voluminous amount of email, 
which isn't necessarily bad per se, but at this point, I find I've been 
conditioned to ignore these emails and notices, though every now and 
then I will forget and then it ends up being harder to keep track of due 
to all the "noise" mixed in.


Instead of "crying wolf" all the time, would it be possible to have 
these comment on issues only if there isn't already an existing comment? 
If that wouldn't be trivial to do, no biggie, just one (minor) painpoint 
I've noticed.


Other than the well known problem of the tests constantly failing to the 
point where I'm just ignoring test failures as well, no other issues 
I've noticed; everything has been pretty smooth. Kudos to George and 
everyone else for making this a really smooth transition.


 NA

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Re: [asterisk-dev] Final Preview: docs.asterisk.org

2023-07-08 Thread asterisk


On 6/29/2023 3:14 PM, George Joseph wrote:
I think we're at a point where the new documentation site can go 
live.  The dynamic documentation is integrated and the README file has 
been expanded greatly with information on how the site is built and 
how you can build it yourself.


Hi, Geroge,
    Just had a chance to look at this this afternoon. The instructions 
for the dynamic doc generation definitely made my head hurt a little 
bit, but I have a few thoughts after putzing around a little bit.


My initial thought was that some of the make targets in the Makefile 
could be split up a little bit. The version-dynamic target both 
downloads documentation source and does the actual build of the 
documentation. They could be split into different targets:


 * In my case, I don't want to download the upstream Asterisk
   documentation, I want to use the local core-en-us.xml, which is a
   superset of the documentation available upstream.
 * Since I'm not downloading the docs, I don't think I need the "gh"
   tool, and so it doesn't need to be installed for purely generating
   documentation. (Also, could be documented as a pre-req, if doing the
   download step)

Then again, when you boil it down to that, it seems like it really comes 
down to just:


 * utils/astxml2markdown.py
 * mike deploy

So it seems I'm better off avoiding the makefile and just running the 
individual commands needed. I'll end up wrapping it in a script anyways, 
but just wondering if there's anything else about the process here I've 
missed that wouldn't be conducive to that?


In particular, it doesn't seem that finagling with Git repositories is 
necessary at all to build the dynamic docs. I was able to get it to work 
with just this:


git clone https://github.com/asterisk/documentation.git --depth 1
cd documentation
pip3 install -r requirements.txt
mv docs/favicon.ico docs/index.md .
rm -rf docs/*
mv favicon.ico index.md docs
utils/astxml2markdown.py 
--file=/usr/src/asterisk-20.3.0/doc/core-en_US.xml 
--xslt=utils/astxml2markdown.xslt --directory=docs/ --branch=20 
--version=20.3.0

mike deploy -F mkdocs.yml 20
rm -rf /var/www/html/asterisk && mv site /var/www/html/asterisk
cd .. && rm -rf documentation

It seems to work really well. There were just a couple surprises or 
annoying aspects:


 * Even with --depth 1, the documentation repo takes a hot minute to
   clone, due to all the large PDF artifacts in it, though a tarball of
   the repo wouldn't help either if it came with all the static
   artifacts anyways. Could probably work around that by cloning it
   using svn instead of git, but I was too lazy to do that today.
 * For just turning markdown into HTML, mike is pretty slow, it takes
   over half a minute just for the dynamic docs (compared to ~1 second
   for my previous method, though that was from the original XML to
   HTML, not from intermediate markdown)
 * Most significantly, the webpages are *huge*. Even just the dynamic
   docs are a whopping 228 MB. On average, a documentation page is
   almost 250 KB (compared to 1.2 MB the old way for all the
   applications, functions, AMI/AGI, and configs on a single webpage -
   granted it's not a fair comparison since the menu and what not isn't
   repeated that way). Taking a look at the page source of an
   application[1], much of the page is whitespace. I know that docs
   hosted on GitHub don't need to worry about disk consumption, but for
   folks building the docs themselves, I think it might be worth trying
   to clean this a little bit. Bloated webpages are obviously also
   going to be slower to load as well.

I haven't yet figured out what's introducing all the extraneous 
whitespace. The markdown files seem okay, but things seem to blow up 
somewhere in the middle. Ideally we could prevent it from happening in 
the first place, but if not, then maybe some fancy recursive 
post-processing could strip it all out.


[1] https://docs.asterisk.org/20/_Dialplan_Applications/ADSIProg/


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[asterisk-dev] 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:

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

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

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

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

Upgrade Notes:

Closed Issues:

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

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

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-dev] 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-dev] 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-dev] 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-dev] Deprecating users.conf

2023-06-30 Thread asterisk


On 6/30/2023 8:32 AM, Jaco Kroon wrote:

On 2023/06/30 14:19, Sean Bright wrote:

On 6/30/2023 7:45 AM, aster...@phreaknet.org wrote:

I've put up a PR to deprecate users.conf[1], following a
discussion earlier this year about this, but I think that was on IRC so
wanted to discuss here as well.
Apologies - I realized after initially commenting on the PR that I 
could have stated my objection immediately rather than direct you here.


I and my users are using users.conf for nearly every install and 
removing support for it would be disruptive to 100s of installs.


Do you mind sharing what these use cases are and what 
functionality/modules you're using it for? As Henning said, maybe there 
is a better way. Either way, it would be good to understand what anyone 
might currently be doing with it.


I've also had some people reach out to me off-list expressing their 
concerns with it being removed.


Do you mind sharing what these concerns are exactly?

I am willing to take over all support for users.conf going forward. I 
can update the module deprecation page¹ indicating I am the maintainer.


If the deprecation warning remains I would need to be able to silence 
it with a command line flag or an option in asterisk.conf.


This would be silly, for a couple reasons:
- If something is deprecated, the idea is that it will be removed 
eventually. If it's not going to be removed, then it doesn't really make 
sense to deprecate to me. Adding an option is just temporarily adding 
integration for something that will be removed soon enough, otherwise.
- The fact that users.conf exists already currently throws warnings for 
users not using it. If it's not on track for deprecation, then we should 
at least silence these warnings so non-users are not confused.
- If this were really practical, then I would also like to see a similar 
flag or option to disable the deprecating warnings for app_adsiprog, 
app_getcpeid, and res_adsi, especially since those are not being 
removed, as those confuse my users. Currently, I patch the source on 
install to remove the deprecation warnings for these 3 modules. If you 
really "need" to silence something, you could theoretically do the same.


I don't have a specific objection against removal, we used this 
instead of dahdi.conf since we could get stuff working for dahdi 
channels that we could not get working in dahdi.conf.

(I'm assuming you mean chan_dahdi.conf, not dahdi.conf)

Can't remember the details but it has remained.  Fairly certain that I 
can dig that up again, and either get it migrated or can make a plan 
to get it sorted so that we can support it. 


Thanks for bringing this up. Do you know what is not working exactly? 
Independent of this issue, we should obviously get that fixed and all 
sorted out.


 NA

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[asterisk-dev] Asterisk Release 20.4.0-rc1

2023-06-30 Thread Asterisk Development Team

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

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

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

Thank You!

Change Log for Release 20.4.0-rc1

Links:

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

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

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

Summary:

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

User Notes:

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

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

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

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

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

[asterisk-dev] Asterisk Release 18.19.0-rc1

2023-06-30 Thread Asterisk Development Team

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

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

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

Thank You!

Change Log for Release 18.19.0-rc1

Links:

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

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

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

Summary:

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

User Notes:

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

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

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

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

- ### app_queue

[asterisk-dev] Deprecating users.conf

2023-06-30 Thread asterisk


Hi folks,

 I've put up a PR to deprecate users.conf[1], following a 
discussion earlier this year about this, but I think that was on IRC so 
wanted to discuss here as well.


Mark introduced users.conf at some point in the early 2000s with the 
idea of it being a "simple" way to configure certain things, but I think 
time has shown that to be its primary weakness. New modules haven't 
supported users.conf in a long time (such as PJSIP), and now that 
chan_sip is already gone, there is basically no point in keeping 
users.conf around anymore. The main two modules that still "support" it 
(if you can call the hack job parsing they do "support") are chan_dahdi 
and chan_iax2, and the configuration for both of these is almost 
entirely non-overlapping and really needs to be configured in the 
appropriate module config file anyways.


Therefore, I am proposing this be deprecated in 21 so that it can be 
removed in 23, in accordance with the Asterisk deprecation policy:


 * Support for users.conf has dwindled as new modules no longer support
   it and modules that did support it (e.g. chan_sip) have been largely
   removed
 * No real functionality has been added to the users.conf mechanism
   since it was introduced. New features are added to specific modules,
   but these are not supported in users.conf
 * users.conf was a super simplistic mechanism that in practice did not
   pan out. It's something that really should never have been added in
   the first place. Use of it has been widely discouraged since it was
   introduction, and caused confusion for Asterisk newcomers,
   particularly with a default install where users see warnings about
   users.conf. Users should not be using it and the fact that the
   sample config still exists continues to create confusion
 * Removing users.conf will help eliminate technical debt, allowing for
   simplification of the codebase and easier maintenance going forward

This is somewhat different as users.conf is not any single module, and 
there's no real process for deprecating a config file, but a warning 
message is added when the PBX loads here so that users will see a notice 
about it, just like with deprecated modules. It's also in the upgrade 
notes. This is a master only change, so it won't be removed until late 
2025 at the earliest.


Any objections or other thoughts on this matter?

[1] https://github.com/asterisk/asterisk/pull/184


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Re: [asterisk-dev] Issue with PJSIP contacts being "unavailable"

2023-06-27 Thread asterisk

On 6/27/2023 6:22 PM, Joshua C. Colp wrote:
On Tue, Jun 27, 2023 at 7:18 PM <mailto:aster...@phreaknet.org>> wrote:

On 6/27/2023 6:07 PM, Joshua C. Colp wrote:
>
> I'm not sure what exactly you are referring to with "using the
> transport used for registration". If "rewrite_contact" is set to
yes
> then the existing active connection should get used. If you are
> referring to Asterisk establishing a new outgoing connection to the
> registered Contact, then as long as it is optional and doesn't
break
> other behavior fine.

Basically, suppose a device registers on a port, associated with some
configured transport.

The reason I'm doing this now is that initially, calls out *to*
devices
would just use the default transport (the first one configured, or
something like that). Specifying a transport= in the endpoint
explicitly
ensured they'd only use the appropriate one. The problem still
remains
though that we don't necessarily know what transport a device is
going
to use in advance, and it could also change at any time.

  I don't know if this would be a "new" outgoing connection to the
contact or not... I was noticing this issue with outbound calls to
devices using the wrong transport (e.g. an ATA registered using
TLS, and
Asterisk would call the device using UDP, on a different port). The
description for "rewrite_contact" says "Allow Contact header to be
rewritten with the source IP address-port" which doesn't really
clarify
that, but if that means it'll always use the same transport out to
the
device that the device initiated a connection on, no matter what,
then I
think that will do the trick. I just want Asterisk to go along with
whatever the device wants to do. If there's a gap with
"rewrite_contact"
then I guess a new option is still needed to do the other half.

The source IP address, port, and transport type become the Contact. 
The Contact target is used for requests, and PJSIP looks for an 
existing active connection meeting that criteria. If such a connection 
is found then it is reused.

Thanks - just to clarify, if such a connection *isn't* found, this won't 
help me right now? It would still use the default transport even with 
rewrite_contact=yes?

In that case, I'm thinking the new option would add on to this by 
extending that behavior to if there isn't an active connection and it 
needs to set up a new one. Basically "use the contact to determine the 
transport, unconditionally" is essentially what it would do.

I guess for devices that don't register, you wouldn't necessarily have a 
contact so maybe that's why this isn't done all the time? But those are 
probably the cases where specifying a transport explicitly would 
probably make more sense anyways, and I'm not concerned about those, 
only things that register and as such a contact would always be available.

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Re: [asterisk-dev] Issue with PJSIP contacts being "unavailable"

2023-06-27 Thread asterisk


On 6/27/2023 6:07 PM, Joshua C. Colp wrote:


I'm not sure what exactly you are referring to with "using the 
transport used for registration". If "rewrite_contact" is set to yes 
then the existing active connection should get used. If you are 
referring to Asterisk establishing a new outgoing connection to the 
registered Contact, then as long as it is optional and doesn't break 
other behavior fine.


Basically, suppose a device registers on a port, associated with some 
configured transport.


The reason I'm doing this now is that initially, calls out *to* devices 
would just use the default transport (the first one configured, or 
something like that). Specifying a transport= in the endpoint explicitly 
ensured they'd only use the appropriate one. The problem still remains 
though that we don't necessarily know what transport a device is going 
to use in advance, and it could also change at any time.


 I don't know if this would be a "new" outgoing connection to the 
contact or not... I was noticing this issue with outbound calls to 
devices using the wrong transport (e.g. an ATA registered using TLS, and 
Asterisk would call the device using UDP, on a different port). The 
description for "rewrite_contact" says "Allow Contact header to be 
rewritten with the source IP address-port" which doesn't really clarify 
that, but if that means it'll always use the same transport out to the 
device that the device initiated a connection on, no matter what, then I 
think that will do the trick. I just want Asterisk to go along with 
whatever the device wants to do. If there's a gap with "rewrite_contact" 
then I guess a new option is still needed to do the other half.


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Re: [asterisk-dev] [asterisk-users] Issue with PJSIP contacts being "unavailable"

2023-06-27 Thread asterisk


On 6/27/2023 7:29 AM, Joshua C. Colp wrote:
On Tue, Jun 27, 2023 at 8:22 AM <mailto:aster...@phreaknet.org>> wrote:





Trace from an "unavailable" ATA (not working correctly):
https://paste.interlinked.us/iz07sapwrb.txt

Trace from an "available" ATA (working correctly):
https://paste.interlinked.us/ocutyjslmg.txt


The "unavailable" ATA no longer has a working TLS connection to 
Asterisk, resulting in OPTIONS failing, and going unreachable, and 
eventually the Contact going away. Actually examining the TLS side 
would be needed.


Thanks, Josh. Further troubleshooting supports that as being the problem 
as well. I'll have to figure out what's changed with that.


Replying to the dev list since this is not directly related, but it 
reminds me of a previous conversation we had about chan_pjsip 
automatically using the transport used for registration. This is not 
currently done; what would be your thoughts on perhaps adding an option 
to do this automatically? Currently, the provisioning system directs 
devices to the proper port based on the security options in the system 
and the TLS capabilities of the device. When something registers, I keep 
track of the port on which a device registers using AMI, logging it to a 
database. We have one port for UDP, one for TLS 1.0, and one for TLS 1.2 
(none for plain TCP at the moment). chan_sip isn't as flexible so the 
process is more straightforward there: just use the TLS 1.0 port if TLS 
is enabled, but for PJSIP, the transpiler assigns a transport based on 
the registration port. In theory, a client can toggle the transport for 
registration (TLS vs UDP), but that alone doesn't really work since 
pjsip.conf needs to be in agreement with that. It would be slightly more 
seamless if it could just sync up somehow, as right now I have to 
manually retranspile any time this occurs, and it seems kind of clunky 
to have to do all this for transports to work properly.


Would there be any consideration or problem with potentially doing 
something like this? After all, it seems like there's a 1:1 mapping and 
it should be fairly straightforward. Kind of like the "line" option for 
registrations, it would help in making things "just work" more of the 
time...


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Re: [asterisk-dev] New Asterisk Documentation website is available for preview

2023-06-21 Thread asterisk


On 6/20/2023 9:48 PM, Andrew Latham wrote:

https://github.com/asterisk/documentation/pull/2 for the binding topic

On Tue, Jun 20, 2023 at 7:42 PM Andrew Latham <mailto:lath...@gmail.com>> wrote:


Read https://github.com/mkdocs/mkdocs/issues/2108 and just have to
say wow...

This worked for me: `mkdocs serve --dev-addr 0.0.0.00:8000`



Thanks, Andrew, for looking into that, and putting those PRs up. Good to 
know that option works. I think he has a point about using a third-party 
server, which is what you'd do anyways, but `mkdocs serve` as a command 
is useless if it can't bind to all interfaces. I don't even use 
containers; to look at the website, it simply has to be bound to all 
interfaces if I want to look at it in a web browser, it's that simple. 
It's very narrow minded to assume that people's workstations are also 
the server that is running XYZ workload, which for me is *never* the case.


The only other thing I can think of at the moment is that it would be 
useful to put the `mkdocs build` command in the README. But this is 
mainly for when the dynamic docs are available, so maybe I'll wait until 
that is finalized as there will probably be more instructions and 
context needed.


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Re: [asterisk-dev] New Asterisk Documentation website is available for preview

2023-06-20 Thread asterisk

On 6/20/2023 8:33 PM, George Joseph wrote:
On Tue, Jun 20, 2023 at 5:06 PM Joshua C. Colp <mailto:jc...@sangoma.com>> wrote:

On Tue, Jun 20, 2023 at 3:51 PM mailto:aster...@phreaknet.org>> wrote:

On 6/20/2023 10:32 AM, George Joseph wrote:
> The one exception is the auto-generated documentation for
> AMI/ARI/Dialplan.  That I'm starting to work on now.
Thanks, George - I see from the README that one can run this
on a local
webserver. Will the auto-generated documentation aspect tie in
with this
as well? I wrote my own xmldoc to HTML generator a while back
so I can
view documentation for out of tree modules. If this can do
that out of
the box, then that would certainly be nice functionality to take
advantage of. Will it just be sourcing from a core xml file,
that we can
point elsewhere if we want, or is that going to remain more
upstream
specific like it was with Confluence?

I don't know how George plans to approach it fully, but ultimately
the reference documentation will also end up as markdown and
consumed with mkdocs. I do not expect those markdown files to be
checked into the tree but generated as part of the deploy process.
Any tooling to consume the XML and produce the markdown files will
be available, so if someone wanted it locally they could.

Each version of Asterisk generates between 800 and 900 pages of 
dynamic docs so it's going to take a bit of thought on how to 
integrate them.  As Josh noted, we don't want those markdown files 
checked into the repo but we do want mkdocs to integrate them 
seamlessly into the main docs site, including the search indexing.  
 We could run a full site build once a night to convert the static and 
dynamic pages into html and index them all BUT we don't have 
server-side searching available so it's done in the browser and right 
now, even without the dynamic pages, the search_index.json file is 
4.1MB.  This might make it prudent to create a "virtual" site with its 
own index just for the dynamic docs and link to it from the main 
site.   Maybe even a separate virtual site for each Asterisk version.  
 In fact, there are tools to create a versioned API site already 
available. Kind of like how Python does it with a drop down at the top 
of every page to select the Python version you want to see the 
documentation for.

Thanks, George - that helps!
I was a bit surprised by how fast the search results were on the new 
site. It seems to be a lot better than the old wiki (which doesn't seem 
to work anymore, either...)

There does seem to be an issue with the web hosting. It seems to be running:
root@debian11:/usr/src/documentation# mkdocs serve
INFO -  Building documentation...
INFO -  Cleaning site directory
INFO -  Documentation built in 16.96 seconds
INFO -  [20:42:02] Watching paths for changes: 'docs', 'mkdocs.yml'
INFO -  [20:42:02] Serving on http://127.0.0.1:8000/

But if I navigate to port 8000 on that machine in my browser, I get 
nothing... nothing even seems to be listening on that port.
It works if I curl localhost on that server, so it seems to be listening 
on just the loopback address. I don't really see how that's helpful - it 
should probably be listening on all interfaces, so one can see what it 
looks like graphically, no?

Realistically though, I wouldn't want to run a separate python server 
anyways, I just want static webpages I can serve in an Apache 
virtualhost, like my current doc generation process. It seems if I run 
"mkbuild docs" it does that. So if the dynamic docs have a similar 
process this seems like it will work great!

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Re: [asterisk-dev] New Asterisk Documentation website is available for preview

2023-06-20 Thread asterisk


On 6/20/2023 10:32 AM, George Joseph wrote:
The one exception is the auto-generated documentation for 
AMI/ARI/Dialplan.  That I'm starting to work on now.
Thanks, George - I see from the README that one can run this on a local 
webserver. Will the auto-generated documentation aspect tie in with this 
as well? I wrote my own xmldoc to HTML generator a while back so I can 
view documentation for out of tree modules. If this can do that out of 
the box, then that would certainly be nice functionality to take 
advantage of. Will it just be sourcing from a core xml file, that we can 
point elsewhere if we want, or is that going to remain more upstream 
specific like it was with Confluence?


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[asterisk-dev] Asterisk Release 20.3.0

2023-05-23 Thread Asterisk Development Team

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

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

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

Thank You!


Change Log for Release 20.3.0


Summary:


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

User Notes:


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

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

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

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

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

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

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

- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
  It is now possible to specify the MixMonitorID when calling
  the manager action:

[asterisk-dev] Asterisk Release 18.18.0

2023-05-23 Thread Asterisk Development Team

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

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

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

Thank You!


Change Log for Release 18.18.0


Summary:


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

User Notes:


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

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

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

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

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

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

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

- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
  It is now possible to specify the MixMonitorID when ca

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

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

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

User Notes:

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

[asterisk-dev] Asterisk Release 20.3.0-rc1

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

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

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

Thank You!


Change Log for Release 20.3.0-rc1


Summary:


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

User Notes:


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

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

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

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

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

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

[asterisk-dev] Asterisk Release 18.18.0-rc1

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

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

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

Thank You!


Change Log for Release 18.18.0-rc1


Summary:


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

User Notes:


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

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

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

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

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

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

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