[Asterisk-code-review] app_mixmonitor: Add option to use real Caller ID for voicemail. (asterisk[20])

Friendly Automation asteriskteam at digium.com
Thu Dec 8 08:04:43 CST 2022


Friendly Automation has submitted this change. ( https://gerrit.asterisk.org/c/asterisk/+/19549 )

Change subject: app_mixmonitor: Add option to use real Caller ID for voicemail.
......................................................................

app_mixmonitor: Add option to use real Caller ID for voicemail.

MixMonitor currently uses the Connected Line as the Caller ID
for voicemails. This is due to the implementation being written
this way for use with Digium phones. However, in general this
is not correct for generic usage in the dialplan, and people
may need the real Caller ID instead. This adds an option to do that.

ASTERISK-30286 #close

Change-Id: I3d0ce76dfe75e2a614e0f709ab27acbd2478267c
---
M apps/app_mixmonitor.c
A doc/CHANGES-staging/app_mixmonitor_clid.txt
2 files changed, 56 insertions(+), 10 deletions(-)

Approvals:
  Joshua Colp: Looks good to me, but someone else must approve
  George Joseph: Looks good to me, approved
  Friendly Automation: Approved for Submit




diff --git a/apps/app_mixmonitor.c b/apps/app_mixmonitor.c
index e2b9e8c..1592c40 100644
--- a/apps/app_mixmonitor.c
+++ b/apps/app_mixmonitor.c
@@ -90,6 +90,11 @@
 						<para>Play a periodic beep while this call is being recorded.</para>
 						<argument name="interval"><para>Interval, in seconds. Default is 15.</para></argument>
 					</option>
+					<option name="c">
+						<para>Use the real Caller ID from the channel for the voicemail Caller ID.</para>
+						<para>By default, the Connected Line is used. If you want the channel caller's
+						real number, you may need to specify this option.</para>
+					</option>
 					<option name="d">
 						<para>Delete the recording file as soon as MixMonitor is done with it.</para>
 						<para>For example, if you use the m option to dispatch the recording to a voicemail box,
@@ -413,6 +418,7 @@
 	MUXFLAG_DEPRECATED_RWSYNC = (1 << 14),
 	MUXFLAG_NO_RWSYNC = (1 << 15),
 	MUXFLAG_AUTO_DELETE = (1 << 16),
+	MUXFLAG_REAL_CALLERID = (1 << 17),
 };
 
 enum mixmonitor_args {
@@ -433,6 +439,7 @@
 	AST_APP_OPTION('a', MUXFLAG_APPEND),
 	AST_APP_OPTION('b', MUXFLAG_BRIDGED),
 	AST_APP_OPTION_ARG('B', MUXFLAG_BEEP, OPT_ARG_BEEP_INTERVAL),
+	AST_APP_OPTION('c', MUXFLAG_REAL_CALLERID),
 	AST_APP_OPTION('d', MUXFLAG_AUTO_DELETE),
 	AST_APP_OPTION('p', MUXFLAG_BEEP_START),
 	AST_APP_OPTION('P', MUXFLAG_BEEP_STOP),
@@ -1035,20 +1042,37 @@
 
 	if (!ast_strlen_zero(recipients)) {
 		char callerid[256];
-		struct ast_party_connected_line *connected;
 
 		ast_channel_lock(chan);
 
-		/* We use the connected line of the invoking channel for caller ID. */
+		/* We use the connected line of the invoking channel for caller ID,
+		 * unless we've been told to use the Caller ID.
+		 * The initial use for this relied on Connected Line to get the
+		 * actual number for recording with Digium phones,
+		 * but in generic use the Caller ID is likely what people want.
+		 */
 
-		connected = ast_channel_connected(chan);
-		ast_debug(3, "Connected Line CID = %d - %s : %d - %s\n", connected->id.name.valid,
-			connected->id.name.str, connected->id.number.valid,
-			connected->id.number.str);
-		ast_callerid_merge(callerid, sizeof(callerid),
-			S_COR(connected->id.name.valid, connected->id.name.str, NULL),
-			S_COR(connected->id.number.valid, connected->id.number.str, NULL),
-			"Unknown");
+		if (ast_test_flag(mixmonitor, MUXFLAG_REAL_CALLERID)) {
+			struct ast_party_caller *caller;
+			caller = ast_channel_caller(chan);
+			ast_debug(3, "Caller ID = %d - %s : %d - %s\n", caller->id.name.valid,
+				caller->id.name.str, caller->id.number.valid,
+				caller->id.number.str);
+			ast_callerid_merge(callerid, sizeof(callerid),
+				S_COR(caller->id.name.valid, caller->id.name.str, NULL),
+				S_COR(caller->id.number.valid, caller->id.number.str, NULL),
+				"Unknown");
+		} else {
+			struct ast_party_connected_line *connected;
+			connected = ast_channel_connected(chan);
+			ast_debug(3, "Connected Line CID = %d - %s : %d - %s\n", connected->id.name.valid,
+				connected->id.name.str, connected->id.number.valid,
+				connected->id.number.str);
+			ast_callerid_merge(callerid, sizeof(callerid),
+				S_COR(connected->id.name.valid, connected->id.name.str, NULL),
+				S_COR(connected->id.number.valid, connected->id.number.str, NULL),
+				"Unknown");
+		}
 
 		ast_string_field_set(mixmonitor, call_context, ast_channel_context(chan));
 		ast_string_field_set(mixmonitor, call_macrocontext, ast_channel_macrocontext(chan));
diff --git a/doc/CHANGES-staging/app_mixmonitor_clid.txt b/doc/CHANGES-staging/app_mixmonitor_clid.txt
new file mode 100644
index 0000000..a8331ec
--- /dev/null
+++ b/doc/CHANGES-staging/app_mixmonitor_clid.txt
@@ -0,0 +1,5 @@
+Subject: app_mixmonitor
+
+Adds the c option to use the real Caller ID on
+the channel in voicemail recordings as opposed
+to the Connected Line.

-- 
To view, visit https://gerrit.asterisk.org/c/asterisk/+/19549
To unsubscribe, or for help writing mail filters, visit https://gerrit.asterisk.org/settings

Gerrit-Project: asterisk
Gerrit-Branch: 20
Gerrit-Change-Id: I3d0ce76dfe75e2a614e0f709ab27acbd2478267c
Gerrit-Change-Number: 19549
Gerrit-PatchSet: 2
Gerrit-Owner: N A <asterisk at phreaknet.org>
Gerrit-Reviewer: Friendly Automation
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Joshua Colp <jcolp at sangoma.com>
Gerrit-MessageType: merged
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-code-review/attachments/20221208/31191501/attachment.html>


More information about the asterisk-code-review mailing list