[asterisk-commits] app dial: Add the "Q" option to set the cause on unanswered... (asterisk[master])
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Oct 11 15:15:56 CDT 2016
Anonymous Coward #1000019 has submitted this change and it was merged.
Change subject: app_dial: Add the "Q" option to set the cause on unanswered channels
......................................................................
app_dial: Add the "Q" option to set the cause on unanswered channels
The "Q" option will set the cause on the unanswered channels when
another channel answers. It overrides the default of
ANSWERED_ELSEWHERE.
NOTE: chan_sip does not support setting the cause on a CANCEL to
anything other than ANSWERED_ELSEWHERE.
ASTERISK-26446 #close
Change-Id: I71742e0919aaa16784c30a2b2e73fbeed7672e47
---
M CHANGES
M apps/app_dial.c
2 files changed, 57 insertions(+), 5 deletions(-)
Approvals:
Richard Mudgett: Looks good to me, but someone else must approve
Anonymous Coward #1000019: Verified
Joshua Colp: Looks good to me, approved
diff --git a/CHANGES b/CHANGES
index be27987..96fdd13 100644
--- a/CHANGES
+++ b/CHANGES
@@ -103,6 +103,12 @@
instance, allows a channel to immediately exit the ConfBridge without having
to wait for a leave announcement to play.
+app_dial
+------------------
+ * Added the "Q" option which sets the Q.850/Q.931 cause on unanswered channels
+ when another channel answers the call. The default of ANSWERED_ELSEWHERE
+ is unchanged.
+
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13 to Asterisk 14 --------------------
------------------------------------------------------------------------------
diff --git a/apps/app_dial.c b/apps/app_dial.c
index 316d38d..b5d8f50 100644
--- a/apps/app_dial.c
+++ b/apps/app_dial.c
@@ -375,6 +375,25 @@
<para>Enable privacy mode. Use <replaceable>x</replaceable> as the family/key in the AstDB database if
it is provided. The current extension is used if a database family/key is not specified.</para>
</option>
+ <option name="Q">
+ <argument name="cause" required="true"/>
+ <para>Specify the Q.850/Q.931 <replaceable>cause</replaceable> to send on
+ unanswered channels when another channel answers the call.
+ As with <literal>Hangup()</literal>, <replaceable>cause</replaceable>
+ can be a numeric cause code or a name such as
+ <literal>NO_ANSWER</literal>,
+ <literal>USER_BUSY</literal>,
+ <literal>CALL_REJECTED</literal> or
+ <literal>ANSWERED_ELSEWHERE</literal> (the default if Q isn't specified).
+ You can also specify <literal>0</literal> or <literal>NONE</literal>
+ to send no cause. See the <filename>causes.h</filename> file for the
+ full list of valid causes and names.
+ </para>
+ <note>
+ <para>chan_sip does not support setting the cause on a CANCEL to anything
+ other than ANSWERED_ELSEWHERE.</para>
+ </note>
+ </option>
<option name="r">
<para>Default: Indicate ringing to the calling party, even if the called party isn't actually ringing. Pass no audio to the calling
party until the called channel has answered.</para>
@@ -519,6 +538,9 @@
</example>
<example title="Dial with call length limit">
same => n,Dial(PJSIP/alice,,L(60000:30000:10000))
+ </example>
+ <example title="Dial alice and bob and send NO_ANSWER to bob instead of ANSWERED_ELSEWHERE when alice answers">
+ same => n,Dial(PJSIP/alice&PJSIP/bob,,Q(NO_ANSWER))
</example>
<example title="Dial with pre-dial subroutines">
[default]
@@ -684,6 +706,7 @@
#define OPT_PREDIAL_CALLEE (1LLU << 41)
#define OPT_PREDIAL_CALLER (1LLU << 42)
#define OPT_RING_WITH_EARLY_MEDIA (1LLU << 43)
+#define OPT_HANGUPCAUSE (1LLU << 44)
enum {
OPT_ARG_ANNOUNCE = 0,
@@ -705,6 +728,7 @@
OPT_ARG_FORCE_CID_PRES,
OPT_ARG_PREDIAL_CALLEE,
OPT_ARG_PREDIAL_CALLER,
+ OPT_ARG_HANGUPCAUSE,
/* note: this entry _MUST_ be the last one in the enum */
OPT_ARG_ARRAY_SIZE
};
@@ -738,6 +762,7 @@
AST_APP_OPTION_ARG('O', OPT_OPERMODE, OPT_ARG_OPERMODE),
AST_APP_OPTION('p', OPT_SCREENING),
AST_APP_OPTION_ARG('P', OPT_PRIVACY, OPT_ARG_PRIVACY),
+ AST_APP_OPTION_ARG('Q', OPT_HANGUPCAUSE, OPT_ARG_HANGUPCAUSE),
AST_APP_OPTION_ARG('r', OPT_RINGBACK, OPT_ARG_RINGBACK),
AST_APP_OPTION('R', OPT_RING_WITH_EARLY_MEDIA),
AST_APP_OPTION_ARG('S', OPT_DURATION_STOP, OPT_ARG_DURATION_STOP),
@@ -796,7 +821,7 @@
ast_free(outgoing);
}
-static void hanguptree(struct dial_head *out_chans, struct ast_channel *exception, int answered_elsewhere)
+static void hanguptree(struct dial_head *out_chans, struct ast_channel *exception, int hangupcause)
{
/* Hang up a tree of stuff */
struct chanlist *outgoing;
@@ -804,9 +829,9 @@
while ((outgoing = AST_LIST_REMOVE_HEAD(out_chans, node))) {
/* Hangup any existing lines we have open */
if (outgoing->chan && (outgoing->chan != exception)) {
- if (answered_elsewhere) {
+ if (hangupcause >= 0) {
/* This is for the channel drivers */
- ast_channel_hangupcause_set(outgoing->chan, AST_CAUSE_ANSWERED_ELSEWHERE);
+ ast_channel_hangupcause_set(outgoing->chan, hangupcause);
}
ast_hangup(outgoing->chan);
}
@@ -2768,6 +2793,7 @@
} else {
const char *number;
int dial_end_raised = 0;
+ int cause = -1;
if (ast_test_flag64(&opts, OPT_CALLER_ANSWER))
ast_answer(chan);
@@ -2778,7 +2804,23 @@
/* Ah ha! Someone answered within the desired timeframe. Of course after this
we will always return with -1 so that it is hung up properly after the
conversation. */
- hanguptree(&out_chans, peer, 1);
+
+ if (ast_test_flag64(&opts, OPT_HANGUPCAUSE)
+ && !ast_strlen_zero(opt_args[OPT_ARG_HANGUPCAUSE])) {
+ cause = ast_str2cause(opt_args[OPT_ARG_HANGUPCAUSE]);
+ if (cause <= 0) {
+ if (!strcasecmp(opt_args[OPT_ARG_HANGUPCAUSE], "NONE")) {
+ cause = 0;
+ } else if (sscanf(opt_args[OPT_ARG_HANGUPCAUSE], "%30d", &cause) != 1
+ || cause < 0) {
+ ast_log(LOG_WARNING, "Invalid cause given to Dial(...Q(<cause>)): \"%s\"\n",
+ opt_args[OPT_ARG_HANGUPCAUSE]);
+ cause = -1;
+ }
+ }
+ }
+ hanguptree(&out_chans, peer, cause >= 0 ? cause : AST_CAUSE_ANSWERED_ELSEWHERE);
+
/* If appropriate, log that we have a destination channel and set the answer time */
if (ast_channel_name(peer))
pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", ast_channel_name(peer));
@@ -3182,7 +3224,11 @@
}
ast_channel_early_bridge(chan, NULL);
- hanguptree(&out_chans, NULL, ast_channel_hangupcause(chan)==AST_CAUSE_ANSWERED_ELSEWHERE || ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE) ? 1 : 0 ); /* forward 'answered elsewhere' if we received it */
+ /* forward 'answered elsewhere' if we received it */
+ hanguptree(&out_chans, NULL,
+ ast_channel_hangupcause(chan) == AST_CAUSE_ANSWERED_ELSEWHERE
+ || ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE)
+ ? AST_CAUSE_ANSWERED_ELSEWHERE : -1);
pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
ast_debug(1, "Exiting with DIALSTATUS=%s.\n", pa.status);
--
To view, visit https://gerrit.asterisk.org/4035
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-MessageType: merged
Gerrit-Change-Id: I71742e0919aaa16784c30a2b2e73fbeed7672e47
Gerrit-PatchSet: 7
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Richard Mudgett <rmudgett at digium.com>
More information about the asterisk-commits
mailing list