From: Daniel Golle <daniel@makrotopia.org>
Date: Wed, 21 Jul 2021 13:17:09 +0000 (+0100)
Subject: asterisk: update to version 18.5.0 and backport patch
X-Git-Url: http://git.openwrt.org/?a=commitdiff_plain;h=20e8c3b2d9150ed0bba02cfeb2e22e7c43e366c8;p=feed%2Ftelephony.git
asterisk: update to version 18.5.0 and backport patch
Update to upstream version 18.5.0 and backport commit[1] completing the
announcements in Dial command to behave like it is stated in the
documenation[2].
[1]: https://github.com/asterisk/asterisk/commit/1e5a2cfe3037823b17dd4ac47b071f02d6f9825f
[2]: https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Application_Dial
Signed-off-by: Daniel Golle <daniel@makrotopia.org>
---
diff --git a/net/asterisk/Makefile b/net/asterisk/Makefile
index 269cd25..9348c8b 100644
--- a/net/asterisk/Makefile
+++ b/net/asterisk/Makefile
@@ -8,12 +8,12 @@
include $(TOPDIR)/rules.mk
PKG_NAME:=asterisk
-PKG_VERSION:=18.4.0
+PKG_VERSION:=18.5.0
PKG_RELEASE:=1
PKG_SOURCE:=asterisk-$(PKG_VERSION).tar.gz
PKG_SOURCE_URL:=https://downloads.asterisk.org/pub/telephony/asterisk/releases
-PKG_HASH:=91c0c35a0c53097647ab3e7b29d34dd0861fed0a8d52e3617408e9eced404db1
+PKG_HASH:=6266a2542ae50225fadf0a3eebb1d11e74b1d1d5cd767cd7b4ff4e65568e8f35
PKG_BUILD_DEPENDS:=libxml2/host
diff --git a/net/asterisk/patches/090-app_dial-expanded-A-option-to-add-caller-announcement.patch b/net/asterisk/patches/090-app_dial-expanded-A-option-to-add-caller-announcement.patch
new file mode 100644
index 0000000..bd20730
--- /dev/null
+++ b/net/asterisk/patches/090-app_dial-expanded-A-option-to-add-caller-announcement.patch
@@ -0,0 +1,147 @@
+From 1e5a2cfe3037823b17dd4ac47b071f02d6f9825f Mon Sep 17 00:00:00 2001
+From: Naveen Albert <mail@interlinked.x10host.com>
+Date: Fri, 21 May 2021 20:08:58 -0400
+Subject: [PATCH] app_dial: Expanded A option to add caller announcement
+
+Hitherto, the A option has made it possible to play
+audio upon answer to the called party only. This option
+is expanded to allow for playback of an audio file to
+the caller instead of or in addition to the audio
+played to the answerer.
+
+ASTERISK-29442
+
+Change-Id: If6eed3ff5c341dc8c588c8210987f2571e891e5e
+---
+ apps/app_dial.c | 79 +++++++++++++++----
+ doc/CHANGES-staging/app_dial_announcement.txt | 6 ++
+ 2 files changed, 68 insertions(+), 17 deletions(-)
+ create mode 100644 doc/CHANGES-staging/app_dial_announcement.txt
+
+--- a/apps/app_dial.c
++++ b/apps/app_dial.c
+@@ -93,11 +93,17 @@
+ </parameter>
+ <parameter name="options" required="false">
+ <optionlist>
+- <option name="A">
+- <argument name="x" required="true">
++ <option name="A" argsep=":">
++ <argument name="x">
+ <para>The file to play to the called party</para>
+ </argument>
+- <para>Play an announcement to the called party, where <replaceable>x</replaceable> is the prompt to be played</para>
++ <argument name="y">
++ <para>The file to play to the calling party</para>
++ </argument>
++ <para>Play an announcement to the called and/or calling parties, where <replaceable>x</replaceable>
++ is the prompt to be played to the called party and <replaceable>y</replaceable> is the prompt
++ to be played to the caller. The files may be different and will be played to each party
++ simultaneously.</para>
+ </option>
+ <option name="a">
+ <para>Immediately answer the calling channel when the called channel answers in
+@@ -2941,33 +2947,71 @@ static int dial_exec_full(struct ast_cha
+ int digit = 0;
+ struct ast_channel *chans[2];
+ struct ast_channel *active_chan;
++ char *calledfile = NULL, *callerfile = NULL;
++ int calledstream = 0, callerstream = 0;
+
+ chans[0] = chan;
+ chans[1] = peer;
+
+- /* we need to stream the announcement to the called party when the OPT_ARG_ANNOUNCE (-A) is setted */
+-
+- /* stream the file */
+- res = ast_streamfile(peer, opt_args[OPT_ARG_ANNOUNCE], ast_channel_language(peer));
+- if (res) {
+- res = 0;
+- ast_log(LOG_ERROR, "error streaming file '%s' to callee\n", opt_args[OPT_ARG_ANNOUNCE]);
++ /* we need to stream the announcement(s) when the OPT_ARG_ANNOUNCE (-A) is set */
++ callerfile = opt_args[OPT_ARG_ANNOUNCE];
++ calledfile = strsep(&callerfile, ":");
++
++ /* stream the file(s) */
++ if (!ast_strlen_zero(calledfile)) {
++ res = ast_streamfile(peer, calledfile, ast_channel_language(peer));
++ if (res) {
++ res = 0;
++ ast_log(LOG_ERROR, "error streaming file '%s' to callee\n", calledfile);
++ } else {
++ calledstream = 1;
++ }
++ }
++ if (!ast_strlen_zero(callerfile)) {
++ res = ast_streamfile(chan, callerfile, ast_channel_language(chan));
++ if (res) {
++ res = 0;
++ ast_log(LOG_ERROR, "error streaming file '%s' to caller\n", callerfile);
++ } else {
++ callerstream = 1;
++ }
+ }
+
++ /* can't use ast_waitstream, because we're streaming two files at once, and can't block
++ We'll need to handle both channels at once. */
++
+ ast_channel_set_flag(peer, AST_FLAG_END_DTMF_ONLY);
+- while (ast_channel_stream(peer)) {
+- int ms;
++ while (ast_channel_stream(peer) || ast_channel_stream(chan)) {
++ int mspeer, mschan;
+
+- ms = ast_sched_wait(ast_channel_sched(peer));
++ mspeer = ast_sched_wait(ast_channel_sched(peer));
++ mschan = ast_sched_wait(ast_channel_sched(chan));
+
+- if (ms < 0 && !ast_channel_timingfunc(peer)) {
+- ast_stopstream(peer);
++ if (calledstream) {
++ if (mspeer < 0 && !ast_channel_timingfunc(peer)) {
++ ast_stopstream(peer);
++ calledstream = 0;
++ }
++ }
++ if (callerstream) {
++ if (mschan < 0 && !ast_channel_timingfunc(chan)) {
++ ast_stopstream(chan);
++ callerstream = 0;
++ }
++ }
++
++ if (!calledstream && !callerstream) {
+ break;
+ }
+- if (ms < 0)
+- ms = 1000;
+
+- active_chan = ast_waitfor_n(chans, 2, &ms);
++ if (mspeer < 0)
++ mspeer = 1000;
++
++ if (mschan < 0)
++ mschan = 1000;
++
++ /* wait for the lowest maximum of the two */
++ active_chan = ast_waitfor_n(chans, 2, (mspeer > mschan ? &mschan : &mspeer));
+ if (active_chan) {
+ struct ast_channel *other_chan;
+ struct ast_frame *fr = ast_read(active_chan);
+@@ -3017,6 +3061,7 @@ static int dial_exec_full(struct ast_cha
+ ast_frfree(fr);
+ }
+ ast_sched_runq(ast_channel_sched(peer));
++ ast_sched_runq(ast_channel_sched(chan));
+ }
+ ast_channel_clear_flag(peer, AST_FLAG_END_DTMF_ONLY);
+ }
+--- /dev/null
++++ b/doc/CHANGES-staging/app_dial_announcement.txt
+@@ -0,0 +1,6 @@
++Subject: app_dial announcement option
++
++The A option for Dial now supports
++playing audio to the caller as well
++as the called party.
++