From 6249324f40bff8d4ec3b529bb7ceb87974659679 Mon Sep 17 00:00:00 2001 From: Asterisk Development Team Date: Thu, 25 Sep 2025 13:49:56 +0000 Subject: [PATCH] Update for 23.0.0-rc2 --- .version | 2 +- CHANGES.html | 2 +- CHANGES.md | 2 +- ChangeLogs/ChangeLog-23.0.0-rc2.html | 81 ++++++++++++++++++++++++ ChangeLogs/ChangeLog-23.0.0-rc2.md | 95 ++++++++++++++++++++++++++++ README.html | 4 +- README.md | 2 +- 7 files changed, 182 insertions(+), 6 deletions(-) create mode 100644 ChangeLogs/ChangeLog-23.0.0-rc2.html create mode 100644 ChangeLogs/ChangeLog-23.0.0-rc2.md diff --git a/.version b/.version index b7c73efac8..e4997573a2 100644 --- a/.version +++ b/.version @@ -1 +1 @@ -23.0.0-rc1 +23.0.0-rc2 diff --git a/CHANGES.html b/CHANGES.html index 10a9546dc2..efb47bec74 120000 --- a/CHANGES.html +++ b/CHANGES.html @@ -1 +1 @@ -ChangeLogs/ChangeLog-23.0.0-rc1.html \ No newline at end of file +ChangeLogs/ChangeLog-23.0.0-rc2.html \ No newline at end of file diff --git a/CHANGES.md b/CHANGES.md index 9526b33867..75cb9dac21 120000 --- a/CHANGES.md +++ b/CHANGES.md @@ -1 +1 @@ -ChangeLogs/ChangeLog-23.0.0-rc1.md \ No newline at end of file +ChangeLogs/ChangeLog-23.0.0-rc2.md \ No newline at end of file diff --git a/ChangeLogs/ChangeLog-23.0.0-rc2.html b/ChangeLogs/ChangeLog-23.0.0-rc2.html new file mode 100644 index 0000000000..d9b143ed40 --- /dev/null +++ b/ChangeLogs/ChangeLog-23.0.0-rc2.html @@ -0,0 +1,81 @@ +ChangeLog for asterisk-23.0.0-rc2 +

Change Log for Release asterisk-23.0.0-rc2

+

Links:

+ +

Summary:

+ +

User Notes:

+

Upgrade Notes:

+

Developer Notes:

+

Commit Authors:

+ +

Issue and Commit Detail:

+

Closed Issues:

+ +

Commits By Author:

+ +

Commit List:

+ +

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

+ +

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

+ diff --git a/ChangeLogs/ChangeLog-23.0.0-rc2.md b/ChangeLogs/ChangeLog-23.0.0-rc2.md new file mode 100644 index 0000000000..1feb3732f9 --- /dev/null +++ b/ChangeLogs/ChangeLog-23.0.0-rc2.md @@ -0,0 +1,95 @@ + +## 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 + diff --git a/README.html b/README.html index 48dbe2cb60..35bf6fe948 100644 --- a/README.html +++ b/README.html @@ -1,4 +1,4 @@ -Readme for asterisk-23.0.0-rc1 +Readme for asterisk-23.0.0-rc2

The Asterisk(R) Open Source PBX

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.

 
If you are updating from a previous version of Asterisk, make sure you
 read the Change Logs.

 
-
Change Logs

+
Change Logs

 
 
 
NEW INSTALLATIONS

diff --git a/README.md b/README.md
index eefe89c9d8..92421abf1b 100644
--- a/README.md
+++ b/README.md
@@ -55,7 +55,7 @@ If you are updating from a previous version of Asterisk, make sure you
 read the Change Logs.
 
 
-[Change Logs](ChangeLogs/ChangeLog-23.0.0-rc1.html)
+[Change Logs](ChangeLogs/ChangeLog-23.0.0-rc2.html)
 
 
 ### NEW INSTALLATIONS