From 0399a3dc730ae639636bcd053193b37751f8df4d Mon Sep 17 00:00:00 2001
From: Andre Heider <a.heider@gmail.com>
Date: Tue, 28 Sep 2021 20:11:44 +0200
Subject: [PATCH] asterisk: bump to 18.7.1

The removed patch has been applied upstream and is part of this release.

Switch to AUTORELEASE while at it.

Signed-off-by: Andre Heider <a.heider@gmail.com>
---
 net/asterisk/Makefile                         |   6 +-
 ...-A-option-to-add-caller-announcement.patch | 147 ------------------
 .../patches/140-use-default-lua.patch         |   2 +-
 3 files changed, 4 insertions(+), 151 deletions(-)
 delete mode 100644 net/asterisk/patches/090-app_dial-expanded-A-option-to-add-caller-announcement.patch

diff --git a/net/asterisk/Makefile b/net/asterisk/Makefile
index 7cbcf9b..b2d9f34 100644
--- a/net/asterisk/Makefile
+++ b/net/asterisk/Makefile
@@ -8,12 +8,12 @@
 include $(TOPDIR)/rules.mk
 
 PKG_NAME:=asterisk
-PKG_VERSION:=18.5.1
-PKG_RELEASE:=1
+PKG_VERSION:=18.7.1
+PKG_RELEASE:=$(AUTORELEASE)
 
 PKG_SOURCE:=asterisk-$(PKG_VERSION).tar.gz
 PKG_SOURCE_URL:=https://downloads.asterisk.org/pub/telephony/asterisk/releases
-PKG_HASH:=e7d78716a0deeadf24b7d537cd24c11c2d9a096265eefc9470565c4da0fc54c7
+PKG_HASH:=11c8728718b59af74440276ba888484bc9201217719d56436e1e0e85bf533c17
 
 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
deleted file mode 100644
index bd20730..0000000
--- a/net/asterisk/patches/090-app_dial-expanded-A-option-to-add-caller-announcement.patch
+++ /dev/null
@@ -1,147 +0,0 @@
-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.
-+
diff --git a/net/asterisk/patches/140-use-default-lua.patch b/net/asterisk/patches/140-use-default-lua.patch
index bd7041f..c71aa8a 100644
--- a/net/asterisk/patches/140-use-default-lua.patch
+++ b/net/asterisk/patches/140-use-default-lua.patch
@@ -1,6 +1,6 @@
 --- a/configure.ac
 +++ b/configure.ac
-@@ -2555,7 +2555,7 @@ if test -z "$__opus_include" -o x"$__opu
+@@ -2556,7 +2556,7 @@ if test -z "$__opus_include" -o x"$__opu
  fi
  AST_EXT_LIB_CHECK([OPUSFILE], [opusfile], [op_open_callbacks], [opus/opusfile.h], [], [$__opus_include])
  
-- 
2.30.2