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1 Commits
23.0.0-rc2
...
23.0.0
| Author | SHA1 | Date | |
|---|---|---|---|
|
3bb594c9ca |
@@ -1 +1 @@
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ChangeLogs/ChangeLog-23.0.0-rc2.html
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ChangeLogs/ChangeLog-23.0.0.html
|
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@@ -1 +1 @@
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ChangeLogs/ChangeLog-23.0.0-rc2.md
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ChangeLogs/ChangeLog-23.0.0.md
|
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@@ -1,81 +0,0 @@
|
||||
<html><head><title>ChangeLog for asterisk-23.0.0-rc2</title></head><body>
|
||||
<h2>Change Log for Release asterisk-23.0.0-rc2</h2>
|
||||
<h3>Links:</h3>
|
||||
<ul>
|
||||
<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0-rc2.html">Full ChangeLog</a> </li>
|
||||
<li><a href="https://github.com/asterisk/asterisk/compare/23.0.0-rc1...23.0.0-rc2">GitHub Diff</a> </li>
|
||||
<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0-rc2.tar.gz">Tarball</a> </li>
|
||||
<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk">Downloads</a> </li>
|
||||
</ul>
|
||||
<h3>Summary:</h3>
|
||||
<ul>
|
||||
<li>Commits: 3</li>
|
||||
<li>Commit Authors: 1</li>
|
||||
<li>Issues Resolved: 3</li>
|
||||
<li>Security Advisories Resolved: 0</li>
|
||||
</ul>
|
||||
<h3>User Notes:</h3>
|
||||
<h3>Upgrade Notes:</h3>
|
||||
<h3>Developer Notes:</h3>
|
||||
<h3>Commit Authors:</h3>
|
||||
<ul>
|
||||
<li>George Joseph: (3)</li>
|
||||
</ul>
|
||||
<h2>Issue and Commit Detail:</h2>
|
||||
<h3>Closed Issues:</h3>
|
||||
<ul>
|
||||
<li>1457: [bug]: segmentation fault because of a wrong ari config</li>
|
||||
<li>1462: [bug]: chan_websocket isn't handling the "opus" codec correctly.</li>
|
||||
<li>1474: [bug]: Media doesn't flow for video conference after res_rtp_asterisk change to stop media flow before DTLS completes</li>
|
||||
</ul>
|
||||
<h3>Commits By Author:</h3>
|
||||
<ul>
|
||||
<li>
|
||||
<h4>George Joseph (3):</h4>
|
||||
</li>
|
||||
<li>res_ari: Ensure outbound websocket config has a websocket_client_id.</li>
|
||||
<li>chan_websocket: Fix codec validation and add passthrough option.</li>
|
||||
<li>res_rtp_asterisk.c: Use rtp->dtls in __rtp_sendto when rtcp mux is used.</li>
|
||||
</ul>
|
||||
<h3>Commit List:</h3>
|
||||
<ul>
|
||||
<li>res_rtp_asterisk.c: Use rtp->dtls in __rtp_sendto when rtcp mux is used.</li>
|
||||
<li>chan_websocket: Fix codec validation and add passthrough option.</li>
|
||||
<li>res_ari: Ensure outbound websocket config has a websocket_client_id.</li>
|
||||
</ul>
|
||||
<h3>Commit Details:</h3>
|
||||
<h4>res_rtp_asterisk.c: Use rtp->dtls in __rtp_sendto when rtcp mux is used.</h4>
|
||||
<p>Author: George Joseph
|
||||
Date: 2025-09-23</p>
|
||||
<p>In __rtp_sendto(), the check for DTLS negotiation completion for rtcp packets
|
||||
needs to use the rtp->dtls structure instead of rtp->rtcp->dtls when
|
||||
AST_RTP_INSTANCE_RTCP_MUX is set.</p>
|
||||
<p>Resolves: #1474</p>
|
||||
<h4>chan_websocket: Fix codec validation and add passthrough option.</h4>
|
||||
<p>Author: George Joseph
|
||||
Date: 2025-09-17</p>
|
||||
<ul>
|
||||
<li>Fixed an issue in webchan_write() where we weren't detecting equivalent
|
||||
codecs properly.</li>
|
||||
<li>Added the "p" dialstring option that puts the channel driver in
|
||||
"passthrough" mode where it will not attempt to re-frame or re-time
|
||||
media coming in over the websocket from the remote app. This can be used
|
||||
for any codec but MUST be used for codecs that use packet headers or whose
|
||||
data stream can't be broken up on arbitrary byte boundaries. In this case,
|
||||
the remote app is fully responsible for correctly framing and timing media
|
||||
sent to Asterisk and the MEDIA text commands that could be sent over the
|
||||
websocket are disabled. Currently, passthrough mode is automatically set
|
||||
for the opus, speex and g729 codecs.</li>
|
||||
<li>Now calling ast_set_read_format() after ast_channel_set_rawreadformat() to
|
||||
ensure proper translation paths are set up when switching between native
|
||||
frames and slin silence frames. This fixes an issue with codec errors
|
||||
when transcode_via_sln=yes.</li>
|
||||
</ul>
|
||||
<p>Resolves: #1462</p>
|
||||
<h4>res_ari: Ensure outbound websocket config has a websocket_client_id.</h4>
|
||||
<p>Author: George Joseph
|
||||
Date: 2025-09-12</p>
|
||||
<p>Added a check to outbound_websocket_apply() that makes sure an outbound
|
||||
websocket config object in ari.conf has a websocket_client_id parameter.</p>
|
||||
<p>Resolves: #1457</p>
|
||||
</body></html>
|
||||
@@ -1,95 +0,0 @@
|
||||
|
||||
## Change Log for Release asterisk-23.0.0-rc2
|
||||
|
||||
### Links:
|
||||
|
||||
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0-rc2.html)
|
||||
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-rc1...23.0.0-rc2)
|
||||
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0-rc2.tar.gz)
|
||||
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
|
||||
|
||||
### Summary:
|
||||
|
||||
- Commits: 3
|
||||
- Commit Authors: 1
|
||||
- Issues Resolved: 3
|
||||
- Security Advisories Resolved: 0
|
||||
|
||||
### User Notes:
|
||||
|
||||
|
||||
### Upgrade Notes:
|
||||
|
||||
|
||||
### Developer Notes:
|
||||
|
||||
|
||||
### Commit Authors:
|
||||
|
||||
- George Joseph: (3)
|
||||
|
||||
## Issue and Commit Detail:
|
||||
|
||||
### Closed Issues:
|
||||
|
||||
- 1457: [bug]: segmentation fault because of a wrong ari config
|
||||
- 1462: [bug]: chan_websocket isn't handling the "opus" codec correctly.
|
||||
- 1474: [bug]: Media doesn't flow for video conference after res_rtp_asterisk change to stop media flow before DTLS completes
|
||||
|
||||
### Commits By Author:
|
||||
|
||||
- #### George Joseph (3):
|
||||
- res_ari: Ensure outbound websocket config has a websocket_client_id.
|
||||
- chan_websocket: Fix codec validation and add passthrough option.
|
||||
- res_rtp_asterisk.c: Use rtp->dtls in __rtp_sendto when rtcp mux is used.
|
||||
|
||||
|
||||
### Commit List:
|
||||
|
||||
- res_rtp_asterisk.c: Use rtp->dtls in __rtp_sendto when rtcp mux is used.
|
||||
- chan_websocket: Fix codec validation and add passthrough option.
|
||||
- res_ari: Ensure outbound websocket config has a websocket_client_id.
|
||||
|
||||
### Commit Details:
|
||||
|
||||
#### res_rtp_asterisk.c: Use rtp->dtls in __rtp_sendto when rtcp mux is used.
|
||||
Author: George Joseph
|
||||
Date: 2025-09-23
|
||||
|
||||
In __rtp_sendto(), the check for DTLS negotiation completion for rtcp packets
|
||||
needs to use the rtp->dtls structure instead of rtp->rtcp->dtls when
|
||||
AST_RTP_INSTANCE_RTCP_MUX is set.
|
||||
|
||||
Resolves: #1474
|
||||
|
||||
#### chan_websocket: Fix codec validation and add passthrough option.
|
||||
Author: George Joseph
|
||||
Date: 2025-09-17
|
||||
|
||||
* Fixed an issue in webchan_write() where we weren't detecting equivalent
|
||||
codecs properly.
|
||||
* Added the "p" dialstring option that puts the channel driver in
|
||||
"passthrough" mode where it will not attempt to re-frame or re-time
|
||||
media coming in over the websocket from the remote app. This can be used
|
||||
for any codec but MUST be used for codecs that use packet headers or whose
|
||||
data stream can't be broken up on arbitrary byte boundaries. In this case,
|
||||
the remote app is fully responsible for correctly framing and timing media
|
||||
sent to Asterisk and the MEDIA text commands that could be sent over the
|
||||
websocket are disabled. Currently, passthrough mode is automatically set
|
||||
for the opus, speex and g729 codecs.
|
||||
* Now calling ast_set_read_format() after ast_channel_set_rawreadformat() to
|
||||
ensure proper translation paths are set up when switching between native
|
||||
frames and slin silence frames. This fixes an issue with codec errors
|
||||
when transcode_via_sln=yes.
|
||||
|
||||
Resolves: #1462
|
||||
|
||||
#### res_ari: Ensure outbound websocket config has a websocket_client_id.
|
||||
Author: George Joseph
|
||||
Date: 2025-09-12
|
||||
|
||||
Added a check to outbound_websocket_apply() that makes sure an outbound
|
||||
websocket config object in ari.conf has a websocket_client_id parameter.
|
||||
|
||||
Resolves: #1457
|
||||
|
||||
@@ -1,17 +1,17 @@
|
||||
<html><head><title>ChangeLog for asterisk-23.0.0-rc1</title></head><body>
|
||||
<h2>Change Log for Release asterisk-23.0.0-rc1</h2>
|
||||
<html><head><title>ChangeLog for asterisk-23.0.0</title></head><body>
|
||||
<h2>Change Log for Release asterisk-23.0.0</h2>
|
||||
<h3>Links:</h3>
|
||||
<ul>
|
||||
<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0-rc1.html">Full ChangeLog</a> </li>
|
||||
<li><a href="https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0-rc1">GitHub Diff</a> </li>
|
||||
<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0-rc1.tar.gz">Tarball</a> </li>
|
||||
<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html">Full ChangeLog</a> </li>
|
||||
<li><a href="https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0">GitHub Diff</a> </li>
|
||||
<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz">Tarball</a> </li>
|
||||
<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk">Downloads</a> </li>
|
||||
</ul>
|
||||
<h3>Summary:</h3>
|
||||
<ul>
|
||||
<li>Commits: 41</li>
|
||||
<li>Commit Authors: 13</li>
|
||||
<li>Issues Resolved: 32</li>
|
||||
<li>Commits: 45</li>
|
||||
<li>Commit Authors: 14</li>
|
||||
<li>Issues Resolved: 36</li>
|
||||
<li>Security Advisories Resolved: 1</li>
|
||||
<li><a href="https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j">GHSA-64qc-9x89-rx5j</a>: A specifically malformed Authorization header in an incoming SIP request can cause Asterisk to crash</li>
|
||||
</ul>
|
||||
@@ -130,9 +130,10 @@
|
||||
<ul>
|
||||
<li>Alexei Gradinari: (1)</li>
|
||||
<li>Alexey Khabulyak: (1)</li>
|
||||
<li>Allan Nathanson: (1)</li>
|
||||
<li>Artem Umerov: (1)</li>
|
||||
<li>Ben Ford: (2)</li>
|
||||
<li>George Joseph: (4)</li>
|
||||
<li>George Joseph: (7)</li>
|
||||
<li>Igor Goncharovsky: (2)</li>
|
||||
<li>Joe Garlick: (1)</li>
|
||||
<li>Jose Lopes: (1)</li>
|
||||
@@ -177,6 +178,10 @@
|
||||
<li>1394: [improvement]: sig_analog: Skip Caller ID spill if Caller ID is disabled</li>
|
||||
<li>1396: [new-feature]: pbx_builtins: Make tone option for WaitExten configurable</li>
|
||||
<li>1401: [bug]: app_waitfornoise timeout is always less then configured because of time() usage</li>
|
||||
<li>1451: [bug]: ast_config_text_file_save2(): incorrect handling of deep/wide template inheritance</li>
|
||||
<li>1457: [bug]: segmentation fault because of a wrong ari config</li>
|
||||
<li>1462: [bug]: chan_websocket isn't handling the "opus" codec correctly.</li>
|
||||
<li>1474: [bug]: Media doesn't flow for video conference after res_rtp_asterisk change to stop media flow before DTLS completes</li>
|
||||
<li>ASTERISK-30370: config: Template inheritance is incorrect for ast_variable_retrieve</li>
|
||||
</ul>
|
||||
<h3>Commits By Author:</h3>
|
||||
@@ -206,13 +211,16 @@
|
||||
<p>res_rtp_asterisk: Don't send RTP before DTLS has negotiated.</p>
|
||||
</li>
|
||||
<li>
|
||||
<h4>George Joseph (4):</h4>
|
||||
<h4>George Joseph (7):</h4>
|
||||
</li>
|
||||
<li>xmldoc.c: Fix rendering of CLI output.</li>
|
||||
<li>chan_websocket: Fix buffer overrun when processing TEXT websocket frames.</li>
|
||||
<li>chan_websocket: Allow additional URI parameters to be added to the outgoing URI.</li>
|
||||
<li>res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.</li>
|
||||
<li>res_ari: Ensure outbound websocket config has a websocket_client_id.</li>
|
||||
<li>chan_websocket: Fix codec validation and add passthrough option.</li>
|
||||
<li>
|
||||
<p>res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.</p>
|
||||
<p>res_rtp_asterisk.c: Use rtp->dtls in __rtp_sendto when rtcp mux is used.</p>
|
||||
</li>
|
||||
<li>
|
||||
<h4>Igor Goncharovsky (2):</h4>
|
||||
@@ -278,6 +286,10 @@
|
||||
<li>logger.c: Remove deprecated/redundant configuration option.</li>
|
||||
<li>func_dialplan: Remove deprecated/redundant function.</li>
|
||||
<li>Update version for Asterisk 23</li>
|
||||
<li>config.c: fix saving of deep/wide template configurations</li>
|
||||
<li>res_rtp_asterisk.c: Use rtp->dtls in __rtp_sendto when rtcp mux is used.</li>
|
||||
<li>chan_websocket: Fix codec validation and add passthrough option.</li>
|
||||
<li>res_ari: Ensure outbound websocket config has a websocket_client_id.</li>
|
||||
<li>chan_websocket.c: Add DTMF messages</li>
|
||||
<li>app_queue.c: Add new global 'log_unpause_on_reason_change'</li>
|
||||
<li>app_waitforsilence.c: Use milliseconds to calculate timeout time</li>
|
||||
@@ -465,6 +477,54 @@
|
||||
<h4>Update version for Asterisk 23</h4>
|
||||
<p>Author: Ben Ford
|
||||
Date: 2025-08-13</p>
|
||||
<h4>config.c: fix saving of deep/wide template configurations</h4>
|
||||
<p>Author: Allan Nathanson
|
||||
Date: 2025-09-10</p>
|
||||
<p>Follow-on to #244 and #960 regarding how the ast_config_XXX APIs
|
||||
handle template inheritance.</p>
|
||||
<p>ast_config_text_file_save2() incorrectly suppressed variables if they
|
||||
matched any ancestor template. This broke deep chains (dropping values
|
||||
based on distant parents) and wide inheritance (ignoring last-wins order
|
||||
across multiple parents).</p>
|
||||
<p>The function now inspects the full template hierarchy to find the nearest
|
||||
effective parent (last occurrence wins). Earlier inherited duplicates are
|
||||
collapsed, explicit overrides are kept unless they exactly match the parent,
|
||||
and PreserveEffectiveContext avoids writing redundant lines.</p>
|
||||
<p>Resolves: #1451</p>
|
||||
<h4>res_rtp_asterisk.c: Use rtp->dtls in __rtp_sendto when rtcp mux is used.</h4>
|
||||
<p>Author: George Joseph
|
||||
Date: 2025-09-23</p>
|
||||
<p>In __rtp_sendto(), the check for DTLS negotiation completion for rtcp packets
|
||||
needs to use the rtp->dtls structure instead of rtp->rtcp->dtls when
|
||||
AST_RTP_INSTANCE_RTCP_MUX is set.</p>
|
||||
<p>Resolves: #1474</p>
|
||||
<h4>chan_websocket: Fix codec validation and add passthrough option.</h4>
|
||||
<p>Author: George Joseph
|
||||
Date: 2025-09-17</p>
|
||||
<ul>
|
||||
<li>Fixed an issue in webchan_write() where we weren't detecting equivalent
|
||||
codecs properly.</li>
|
||||
<li>Added the "p" dialstring option that puts the channel driver in
|
||||
"passthrough" mode where it will not attempt to re-frame or re-time
|
||||
media coming in over the websocket from the remote app. This can be used
|
||||
for any codec but MUST be used for codecs that use packet headers or whose
|
||||
data stream can't be broken up on arbitrary byte boundaries. In this case,
|
||||
the remote app is fully responsible for correctly framing and timing media
|
||||
sent to Asterisk and the MEDIA text commands that could be sent over the
|
||||
websocket are disabled. Currently, passthrough mode is automatically set
|
||||
for the opus, speex and g729 codecs.</li>
|
||||
<li>Now calling ast_set_read_format() after ast_channel_set_rawreadformat() to
|
||||
ensure proper translation paths are set up when switching between native
|
||||
frames and slin silence frames. This fixes an issue with codec errors
|
||||
when transcode_via_sln=yes.</li>
|
||||
</ul>
|
||||
<p>Resolves: #1462</p>
|
||||
<h4>res_ari: Ensure outbound websocket config has a websocket_client_id.</h4>
|
||||
<p>Author: George Joseph
|
||||
Date: 2025-09-12</p>
|
||||
<p>Added a check to outbound_websocket_apply() that makes sure an outbound
|
||||
websocket config object in ari.conf has a websocket_client_id parameter.</p>
|
||||
<p>Resolves: #1457</p>
|
||||
<h4>chan_websocket.c: Add DTMF messages</h4>
|
||||
<p>Author: Joe Garlick
|
||||
Date: 2025-09-04</p>
|
||||
@@ -1,18 +1,18 @@
|
||||
|
||||
## Change Log for Release asterisk-23.0.0-rc1
|
||||
## Change Log for Release asterisk-23.0.0
|
||||
|
||||
### Links:
|
||||
|
||||
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0-rc1.html)
|
||||
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0-rc1)
|
||||
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0-rc1.tar.gz)
|
||||
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.0.0.html)
|
||||
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0-pre1...23.0.0)
|
||||
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.0.0.tar.gz)
|
||||
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
|
||||
|
||||
### Summary:
|
||||
|
||||
- Commits: 41
|
||||
- Commit Authors: 13
|
||||
- Issues Resolved: 32
|
||||
- Commits: 45
|
||||
- Commit Authors: 14
|
||||
- Issues Resolved: 36
|
||||
- Security Advisories Resolved: 1
|
||||
- [GHSA-64qc-9x89-rx5j](https://github.com/asterisk/asterisk/security/advisories/GHSA-64qc-9x89-rx5j): A specifically malformed Authorization header in an incoming SIP request can cause Asterisk to crash
|
||||
|
||||
@@ -115,9 +115,10 @@
|
||||
|
||||
- Alexei Gradinari: (1)
|
||||
- Alexey Khabulyak: (1)
|
||||
- Allan Nathanson: (1)
|
||||
- Artem Umerov: (1)
|
||||
- Ben Ford: (2)
|
||||
- George Joseph: (4)
|
||||
- George Joseph: (7)
|
||||
- Igor Goncharovsky: (2)
|
||||
- Joe Garlick: (1)
|
||||
- Jose Lopes: (1)
|
||||
@@ -163,6 +164,10 @@
|
||||
- 1394: [improvement]: sig_analog: Skip Caller ID spill if Caller ID is disabled
|
||||
- 1396: [new-feature]: pbx_builtins: Make tone option for WaitExten configurable
|
||||
- 1401: [bug]: app_waitfornoise timeout is always less then configured because of time() usage
|
||||
- 1451: [bug]: ast_config_text_file_save2(): incorrect handling of deep/wide template inheritance
|
||||
- 1457: [bug]: segmentation fault because of a wrong ari config
|
||||
- 1462: [bug]: chan_websocket isn't handling the "opus" codec correctly.
|
||||
- 1474: [bug]: Media doesn't flow for video conference after res_rtp_asterisk change to stop media flow before DTLS completes
|
||||
- ASTERISK-30370: config: Template inheritance is incorrect for ast_variable_retrieve
|
||||
|
||||
### Commits By Author:
|
||||
@@ -179,11 +184,14 @@
|
||||
- #### Ben Ford (1):
|
||||
- res_rtp_asterisk: Don't send RTP before DTLS has negotiated.
|
||||
|
||||
- #### George Joseph (4):
|
||||
- #### George Joseph (7):
|
||||
- xmldoc.c: Fix rendering of CLI output.
|
||||
- chan_websocket: Fix buffer overrun when processing TEXT websocket frames.
|
||||
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI.
|
||||
- res_pjsip_authenticator_digest: Fix SEGV if get_authorization_hdr returns NULL.
|
||||
- res_ari: Ensure outbound websocket config has a websocket_client_id.
|
||||
- chan_websocket: Fix codec validation and add passthrough option.
|
||||
- res_rtp_asterisk.c: Use rtp->dtls in __rtp_sendto when rtcp mux is used.
|
||||
|
||||
- #### Igor Goncharovsky (2):
|
||||
- app_waitforsilence.c: Use milliseconds to calculate timeout time
|
||||
@@ -233,6 +241,10 @@
|
||||
- logger.c: Remove deprecated/redundant configuration option.
|
||||
- func_dialplan: Remove deprecated/redundant function.
|
||||
- Update version for Asterisk 23
|
||||
- config.c: fix saving of deep/wide template configurations
|
||||
- res_rtp_asterisk.c: Use rtp->dtls in __rtp_sendto when rtcp mux is used.
|
||||
- chan_websocket: Fix codec validation and add passthrough option.
|
||||
- res_ari: Ensure outbound websocket config has a websocket_client_id.
|
||||
- chan_websocket.c: Add DTMF messages
|
||||
- app_queue.c: Add new global 'log_unpause_on_reason_change'
|
||||
- app_waitforsilence.c: Use milliseconds to calculate timeout time
|
||||
@@ -481,6 +493,66 @@
|
||||
Date: 2025-08-13
|
||||
|
||||
|
||||
#### config.c: fix saving of deep/wide template configurations
|
||||
Author: Allan Nathanson
|
||||
Date: 2025-09-10
|
||||
|
||||
Follow-on to #244 and #960 regarding how the ast_config_XXX APIs
|
||||
handle template inheritance.
|
||||
|
||||
ast_config_text_file_save2() incorrectly suppressed variables if they
|
||||
matched any ancestor template. This broke deep chains (dropping values
|
||||
based on distant parents) and wide inheritance (ignoring last-wins order
|
||||
across multiple parents).
|
||||
|
||||
The function now inspects the full template hierarchy to find the nearest
|
||||
effective parent (last occurrence wins). Earlier inherited duplicates are
|
||||
collapsed, explicit overrides are kept unless they exactly match the parent,
|
||||
and PreserveEffectiveContext avoids writing redundant lines.
|
||||
|
||||
Resolves: #1451
|
||||
|
||||
#### res_rtp_asterisk.c: Use rtp->dtls in __rtp_sendto when rtcp mux is used.
|
||||
Author: George Joseph
|
||||
Date: 2025-09-23
|
||||
|
||||
In __rtp_sendto(), the check for DTLS negotiation completion for rtcp packets
|
||||
needs to use the rtp->dtls structure instead of rtp->rtcp->dtls when
|
||||
AST_RTP_INSTANCE_RTCP_MUX is set.
|
||||
|
||||
Resolves: #1474
|
||||
|
||||
#### chan_websocket: Fix codec validation and add passthrough option.
|
||||
Author: George Joseph
|
||||
Date: 2025-09-17
|
||||
|
||||
* Fixed an issue in webchan_write() where we weren't detecting equivalent
|
||||
codecs properly.
|
||||
* Added the "p" dialstring option that puts the channel driver in
|
||||
"passthrough" mode where it will not attempt to re-frame or re-time
|
||||
media coming in over the websocket from the remote app. This can be used
|
||||
for any codec but MUST be used for codecs that use packet headers or whose
|
||||
data stream can't be broken up on arbitrary byte boundaries. In this case,
|
||||
the remote app is fully responsible for correctly framing and timing media
|
||||
sent to Asterisk and the MEDIA text commands that could be sent over the
|
||||
websocket are disabled. Currently, passthrough mode is automatically set
|
||||
for the opus, speex and g729 codecs.
|
||||
* Now calling ast_set_read_format() after ast_channel_set_rawreadformat() to
|
||||
ensure proper translation paths are set up when switching between native
|
||||
frames and slin silence frames. This fixes an issue with codec errors
|
||||
when transcode_via_sln=yes.
|
||||
|
||||
Resolves: #1462
|
||||
|
||||
#### res_ari: Ensure outbound websocket config has a websocket_client_id.
|
||||
Author: George Joseph
|
||||
Date: 2025-09-12
|
||||
|
||||
Added a check to outbound_websocket_apply() that makes sure an outbound
|
||||
websocket config object in ari.conf has a websocket_client_id parameter.
|
||||
|
||||
Resolves: #1457
|
||||
|
||||
#### chan_websocket.c: Add DTMF messages
|
||||
Author: Joe Garlick
|
||||
Date: 2025-09-04
|
||||
@@ -1,4 +1,4 @@
|
||||
<html><head><title>Readme for asterisk-23.0.0-rc2</title></head><body>
|
||||
<html><head><title>Readme for asterisk-23.0.0</title></head><body>
|
||||
<h1>The Asterisk(R) Open Source PBX</h1>
|
||||
<pre><code>By Mark Spencer <markster@digium.com> and the Asterisk.org developer community.
|
||||
Copyright (C) 2001-2025 Sangoma Technologies Corporation and other copyright holders.
|
||||
@@ -37,7 +37,7 @@ hardware.</p>
|
||||
<p>If you are updating from a previous version of Asterisk, make sure you
|
||||
read the Change Logs.</p>
|
||||
<!-- CHANGELOGS (the URL will change based on the location of this README) -->
|
||||
<p><a href="ChangeLogs/ChangeLog-23.0.0-rc2.html">Change Logs</a></p>
|
||||
<p><a href="ChangeLogs/ChangeLog-23.0.0.html">Change Logs</a></p>
|
||||
<!-- END-CHANGELOGS -->
|
||||
|
||||
<h3>NEW INSTALLATIONS</h3>
|
||||
|
||||
@@ -55,7 +55,7 @@ If you are updating from a previous version of Asterisk, make sure you
|
||||
read the Change Logs.
|
||||
|
||||
<!-- CHANGELOGS (the URL will change based on the location of this README) -->
|
||||
[Change Logs](ChangeLogs/ChangeLog-23.0.0-rc2.html)
|
||||
[Change Logs](ChangeLogs/ChangeLog-23.0.0.html)
|
||||
<!-- END-CHANGELOGS -->
|
||||
|
||||
### NEW INSTALLATIONS
|
||||
|
||||
Reference in New Issue
Block a user