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Update for 22.6.0

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2025-10-15 16:49:22 +00:00
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22.6.0-rc2 22.6.0

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ChangeLogs/ChangeLog-22.6.0-rc2.html ChangeLogs/ChangeLog-22.6.0.html

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<html><head><title>ChangeLog for asterisk-22.6.0-rc2</title></head><body>
<h2>Change Log for Release asterisk-22.6.0-rc2</h2>
<h3>Links:</h3>
<ul>
<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0-rc2.html">Full ChangeLog</a> </li>
<li><a href="https://github.com/asterisk/asterisk/compare/22.6.0-rc1...22.6.0-rc2">GitHub Diff</a> </li>
<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0-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-&gt;dtls in __rtp_sendto when rtcp mux is used.</li>
</ul>
<h3>Commit List:</h3>
<ul>
<li>res_rtp_asterisk.c: Use rtp-&gt;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-&gt;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-&gt;dtls structure instead of rtp-&gt;rtcp-&gt;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>

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## Change Log for Release asterisk-22.6.0-rc2
### Links:
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0-rc2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0-rc1...22.6.0-rc2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0-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

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@@ -1,17 +1,17 @@
<html><head><title>ChangeLog for asterisk-22.6.0-rc1</title></head><body> <html><head><title>ChangeLog for asterisk-22.6.0</title></head><body>
<h2>Change Log for Release asterisk-22.6.0-rc1</h2> <h2>Change Log for Release asterisk-22.6.0</h2>
<h3>Links:</h3> <h3>Links:</h3>
<ul> <ul>
<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0-rc1.html">Full ChangeLog</a> </li> <li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html">Full ChangeLog</a> </li>
<li><a href="https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0-rc1">GitHub Diff</a> </li> <li><a href="https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0">GitHub Diff</a> </li>
<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0-rc1.tar.gz">Tarball</a> </li> <li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz">Tarball</a> </li>
<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk">Downloads</a> </li> <li><a href="https://downloads.asterisk.org/pub/telephony/asterisk">Downloads</a> </li>
</ul> </ul>
<h3>Summary:</h3> <h3>Summary:</h3>
<ul> <ul>
<li>Commits: 51</li> <li>Commits: 54</li>
<li>Commit Authors: 22</li> <li>Commit Authors: 22</li>
<li>Issues Resolved: 37</li> <li>Issues Resolved: 40</li>
<li>Security Advisories Resolved: 0</li> <li>Security Advisories Resolved: 0</li>
</ul> </ul>
<h3>User Notes:</h3> <h3>User Notes:</h3>
@@ -124,7 +124,7 @@
<li>Allan Nathanson: (1)</li> <li>Allan Nathanson: (1)</li>
<li>Artem Umerov: (1)</li> <li>Artem Umerov: (1)</li>
<li>Ben Ford: (1)</li> <li>Ben Ford: (1)</li>
<li>George Joseph: (9)</li> <li>George Joseph: (12)</li>
<li>Igor Goncharovsky: (2)</li> <li>Igor Goncharovsky: (2)</li>
<li>Jaco Kroon: (1)</li> <li>Jaco Kroon: (1)</li>
<li>Joe Garlick: (1)</li> <li>Joe Garlick: (1)</li>
@@ -182,6 +182,9 @@
<li>1394: [improvement]: sig_analog: Skip Caller ID spill if Caller ID is disabled</li> <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>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>1401: [bug]: app_waitfornoise timeout is always less then configured because of time() usage</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>
</ul> </ul>
<h3>Commits By Author:</h3> <h3>Commits By Author:</h3>
<ul> <ul>
@@ -218,7 +221,7 @@
<p>res_rtp_asterisk: Don't send RTP before DTLS has negotiated.</p> <p>res_rtp_asterisk: Don't send RTP before DTLS has negotiated.</p>
</li> </li>
<li> <li>
<h4>George Joseph (9):</h4> <h4>George Joseph (12):</h4>
</li> </li>
<li>Media over Websocket Channel Driver</li> <li>Media over Websocket Channel Driver</li>
<li>app_mixmonitor: Update the documentation concerning the "D" option.</li> <li>app_mixmonitor: Update the documentation concerning the "D" option.</li>
@@ -228,8 +231,11 @@
<li>channelstorage_cpp_map_name_id.cc: Refactor iterators for thread-safety.</li> <li>channelstorage_cpp_map_name_id.cc: Refactor iterators for thread-safety.</li>
<li>xmldoc.c: Fix rendering of CLI output.</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: 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_ari: Ensure outbound websocket config has a websocket_client_id.</li>
<li>chan_websocket: Fix codec validation and add passthrough option.</li>
<li> <li>
<p>chan_websocket: Allow additional URI parameters to be added to the outgoing URI.</p> <p>res_rtp_asterisk.c: Use rtp-&gt;dtls in __rtp_sendto when rtcp mux is used.</p>
</li> </li>
<li> <li>
<h4>Igor Goncharovsky (2):</h4> <h4>Igor Goncharovsky (2):</h4>
@@ -347,6 +353,9 @@
</ul> </ul>
<h3>Commit List:</h3> <h3>Commit List:</h3>
<ul> <ul>
<li>res_rtp_asterisk.c: Use rtp-&gt;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>chan_websocket.c: Add DTMF messages</li>
<li>app_queue.c: Add new global 'log_unpause_on_reason_change'</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> <li>app_waitforsilence.c: Use milliseconds to calculate timeout time</li>
@@ -394,6 +403,40 @@
<li>res_musiconhold: Appropriately lock channel during start.</li> <li>res_musiconhold: Appropriately lock channel during start.</li>
</ul> </ul>
<h3>Commit Details:</h3> <h3>Commit Details:</h3>
<h4>res_rtp_asterisk.c: Use rtp-&gt;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-&gt;dtls structure instead of rtp-&gt;rtcp-&gt;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> <h4>chan_websocket.c: Add DTMF messages</h4>
<p>Author: Joe Garlick <p>Author: Joe Garlick
Date: 2025-09-04</p> Date: 2025-09-04</p>

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@@ -1,18 +1,18 @@
## Change Log for Release asterisk-22.6.0-rc1 ## Change Log for Release asterisk-22.6.0
### Links: ### Links:
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0-rc1.html) - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.6.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0-rc1) - [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.5.2...22.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0-rc1.tar.gz) - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
### Summary: ### Summary:
- Commits: 51 - Commits: 54
- Commit Authors: 22 - Commit Authors: 22
- Issues Resolved: 37 - Issues Resolved: 40
- Security Advisories Resolved: 0 - Security Advisories Resolved: 0
### User Notes: ### User Notes:
@@ -113,7 +113,7 @@
- Allan Nathanson: (1) - Allan Nathanson: (1)
- Artem Umerov: (1) - Artem Umerov: (1)
- Ben Ford: (1) - Ben Ford: (1)
- George Joseph: (9) - George Joseph: (12)
- Igor Goncharovsky: (2) - Igor Goncharovsky: (2)
- Jaco Kroon: (1) - Jaco Kroon: (1)
- Joe Garlick: (1) - Joe Garlick: (1)
@@ -172,6 +172,9 @@
- 1394: [improvement]: sig_analog: Skip Caller ID spill if Caller ID is disabled - 1394: [improvement]: sig_analog: Skip Caller ID spill if Caller ID is disabled
- 1396: [new-feature]: pbx_builtins: Make tone option for WaitExten configurable - 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 - 1401: [bug]: app_waitfornoise timeout is always less then configured because of time() usage
- 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: ### Commits By Author:
@@ -192,7 +195,7 @@
- #### Ben Ford (1): - #### Ben Ford (1):
- res_rtp_asterisk: Don't send RTP before DTLS has negotiated. - res_rtp_asterisk: Don't send RTP before DTLS has negotiated.
- #### George Joseph (9): - #### George Joseph (12):
- Media over Websocket Channel Driver - Media over Websocket Channel Driver
- app_mixmonitor: Update the documentation concerning the "D" option. - app_mixmonitor: Update the documentation concerning the "D" option.
- cdr.c: Set tenantid from party_a->base instead of chan->base. - cdr.c: Set tenantid from party_a->base instead of chan->base.
@@ -202,6 +205,9 @@
- xmldoc.c: Fix rendering of CLI output. - xmldoc.c: Fix rendering of CLI output.
- chan_websocket: Fix buffer overrun when processing TEXT websocket frames. - chan_websocket: Fix buffer overrun when processing TEXT websocket frames.
- chan_websocket: Allow additional URI parameters to be added to the outgoing URI. - chan_websocket: Allow additional URI parameters to be added to the outgoing URI.
- 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): - #### Igor Goncharovsky (2):
- app_waitforsilence.c: Use milliseconds to calculate timeout time - app_waitforsilence.c: Use milliseconds to calculate timeout time
@@ -273,6 +279,9 @@
### Commit List: ### 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.
- chan_websocket.c: Add DTMF messages - chan_websocket.c: Add DTMF messages
- app_queue.c: Add new global 'log_unpause_on_reason_change' - app_queue.c: Add new global 'log_unpause_on_reason_change'
- app_waitforsilence.c: Use milliseconds to calculate timeout time - app_waitforsilence.c: Use milliseconds to calculate timeout time
@@ -321,6 +330,47 @@
### Commit Details: ### 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
#### chan_websocket.c: Add DTMF messages #### chan_websocket.c: Add DTMF messages
Author: Joe Garlick Author: Joe Garlick
Date: 2025-09-04 Date: 2025-09-04

4
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@@ -1,4 +1,4 @@
<html><head><title>Readme for asterisk-22.6.0-rc2</title></head><body> <html><head><title>Readme for asterisk-22.6.0</title></head><body>
<h1>The Asterisk(R) Open Source PBX</h1> <h1>The Asterisk(R) Open Source PBX</h1>
<pre><code>By Mark Spencer &lt;markster@digium.com&gt; and the Asterisk.org developer community. <pre><code>By Mark Spencer &lt;markster@digium.com&gt; and the Asterisk.org developer community.
Copyright (C) 2001-2025 Sangoma Technologies Corporation and other copyright holders. 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 <p>If you are updating from a previous version of Asterisk, make sure you
read the Change Logs.</p> read the Change Logs.</p>
<!-- CHANGELOGS (the URL will change based on the location of this README) --> <!-- CHANGELOGS (the URL will change based on the location of this README) -->
<p><a href="ChangeLogs/ChangeLog-22.6.0-rc2.html">Change Logs</a></p> <p><a href="ChangeLogs/ChangeLog-22.6.0.html">Change Logs</a></p>
<!-- END-CHANGELOGS --> <!-- END-CHANGELOGS -->
<h3>NEW INSTALLATIONS</h3> <h3>NEW INSTALLATIONS</h3>

2
README.md
View File

@@ -55,7 +55,7 @@ If you are updating from a previous version of Asterisk, make sure you
read the Change Logs. read the Change Logs.
<!-- CHANGELOGS (the URL will change based on the location of this README) --> <!-- CHANGELOGS (the URL will change based on the location of this README) -->
[Change Logs](ChangeLogs/ChangeLog-22.6.0-rc2.html) [Change Logs](ChangeLogs/ChangeLog-22.6.0.html)
<!-- END-CHANGELOGS --> <!-- END-CHANGELOGS -->
### NEW INSTALLATIONS ### NEW INSTALLATIONS