Implementation of a feature that will disable "missed calls" counters on SIP phones.

If the call is answered by another phone, other phones won't display the call as "missed".
You can also add an option to the dial command so that you can have a "followme"
scenario and not count the calls as "missed" when you cancel the call.

Thanks to Ramon and Frank for feedback on this feature.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@74024 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/apps/app_dial.c b/apps/app_dial.c
index e566f58..8751d65 100644 (file)
--- a/apps/app_dial.c
+++ b/apps/app_dial.c
@@ -99,6 +99,8 @@ static char *descrip =
 "  Options:\n"
 "    A(x) - Play an announcement to the called party, using 'x' as the file.\n"
 "    C    - Reset the CDR for this call.\n"
+"    c    - If DIAL cancels this call, always set the flag to tell the channel\n"
+"           driver that the call is answered elsewhere.\n"
 "    d    - Allow the calling user to dial a 1 digit extension while waiting for\n"
 "           a call to be answered. Exit to that extension if it exists in the\n"
 "           current context, or the context defined in the EXITCONTEXT variable,\n"
@@ -253,6 +255,7 @@ enum {
        OPT_CALLER_PARK =       (1 << 26),
        OPT_IGNORE_FORWARDING = (1 << 27),
        OPT_CALLEE_GOSUB =      (1 << 28),
+       OPT_CANCEL_ELSEWHERE =  (1 << 29),
 };
 
 #define DIAL_STILLGOING                        (1 << 30)
@@ -276,6 +279,7 @@ enum {
 AST_APP_OPTIONS(dial_exec_options, {
        AST_APP_OPTION_ARG('A', OPT_ANNOUNCE, OPT_ARG_ANNOUNCE),
        AST_APP_OPTION('C', OPT_RESETCDR),
+       AST_APP_OPTION('c', OPT_CANCEL_ELSEWHERE),
        AST_APP_OPTION('d', OPT_DTMF_EXIT),
        AST_APP_OPTION_ARG('D', OPT_SENDDTMF, OPT_ARG_SENDDTMF),
        AST_APP_OPTION('f', OPT_FORCECLID),
@@ -315,14 +319,17 @@ struct chanlist {
 };
 
 
-static void hanguptree(struct chanlist *outgoing, struct ast_channel *exception)
+static void hanguptree(struct chanlist *outgoing, struct ast_channel *exception, int answered_elsewhere)
 {
        /* Hang up a tree of stuff */
        struct chanlist *oo;
        while (outgoing) {
                /* Hangup any existing lines we have open */
-               if (outgoing->chan && (outgoing->chan != exception))
+               if (outgoing->chan && (outgoing->chan != exception)) {
+                       if (answered_elsewhere)
+                               ast_set_flag(outgoing->chan, AST_FLAG_ANSWERED_ELSEWHERE);
                        ast_hangup(outgoing->chan);
+               }
                oo = outgoing;
                outgoing=outgoing->next;
                ast_free(oo);
@@ -1314,6 +1321,7 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
                        goto out;
                if (opts.flags) {
                        ast_copy_flags(tmp, &opts,
+                                      OPT_CANCEL_ELSEWHERE |
                                       OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
                                       OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
                                       OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
@@ -1513,7 +1521,7 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
                /* 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(outgoing, peer);
+               hanguptree(outgoing, peer, 1);
                outgoing = NULL;
                /* If appropriate, log that we have a destination channel */
                if (chan->cdr)
@@ -1562,7 +1570,7 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
                        ast_parseable_goto(peer, opt_args[OPT_ARG_GOTO]);
                        peer->priority++;
                        ast_pbx_start(peer);
-                       hanguptree(outgoing, NULL);
+                       hanguptree(outgoing, NULL, ast_test_flag(&opts, OPT_CANCEL_ELSEWHERE ? 1 : 0));
                        if (continue_exec)
                                *continue_exec = 1;
                        res = 0;
@@ -1800,7 +1808,7 @@ out:
                ast_indicate(chan, -1);
        }
        ast_channel_early_bridge(chan, NULL);
-       hanguptree(outgoing, NULL);
+       hanguptree(outgoing, NULL, 0);  /* In this case, there's no answer anywhere */
        pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
        senddialendevent(chan, pa.status);
        ast_debug(1, "Exiting with DIALSTATUS=%s.\n", pa.status);

diff --git a/channels/chan_local.c b/channels/chan_local.c
index beac406..b4072ce 100644 (file)
--- a/channels/chan_local.c
+++ b/channels/chan_local.c
@@ -495,6 +495,8 @@ static int local_hangup(struct ast_channel *ast)
                return -1;
 
        ast_mutex_lock(&p->lock);
+       if (p->chan && ast_test_flag(ast, AST_FLAG_ANSWERED_ELSEWHERE)) 
+               ast_set_flag(p->chan, AST_FLAG_ANSWERED_ELSEWHERE);
        isoutbound = IS_OUTBOUND(ast, p);
        if (isoutbound) {
                const char *status = pbx_builtin_getvar_helper(p->chan, "DIALSTATUS");

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 6c9ea33..9cebd9d 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -752,7 +752,7 @@ struct sip_auth {
 #define SIP_REALTIME           (1 << 11)       /*!< P: Flag for realtime users */
 #define SIP_USECLIENTCODE      (1 << 12)       /*!< DP: Trust X-ClientCode info message */
 #define SIP_OUTGOING           (1 << 13)       /*!< D: Direction of the last transaction in this dialog */
-#define SIP_FREE_BIT           (1 << 14)       /*!< ---- */
+#define SIP_DIALOG_ANSWEREDELSEWHERE   (1 << 14)       /*!< D: This call is cancelled due to answer on another channel */
 #define SIP_DEFER_BYE_ON_TRANSFER      (1 << 15)       /*!< D: Do not hangup at first ast_hangup */
 #define SIP_DTMF               (3 << 16)       /*!< DP: DTMF Support: four settings, uses two bits */
 #define SIP_DTMF_RFC2833       (0 << 16)       /*!< DP: DTMF Support: RTP DTMF - "rfc2833" */
@@ -3674,6 +3674,12 @@ static int sip_hangup(struct ast_channel *ast)
                ast_debug(1, "Asked to hangup channel that was not connected\n");
                return 0;
        }
+       if (ast_test_flag(ast, AST_FLAG_ANSWERED_ELSEWHERE)) {
+               if (option_debug)
+                       ast_log(LOG_DEBUG, "This call was answered elsewhere");
+               append_history(p, "Cancel", "Call answered elsewhere");
+               ast_set_flag(&p->flags[0], SIP_DIALOG_ANSWEREDELSEWHERE);
+       }
 
        if (ast_test_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER)) {
                if (ast_test_flag(&p->flags[0], SIP_INC_COUNT)) {
@@ -8140,6 +8146,9 @@ static int transmit_request(struct sip_pvt *p, int sipmethod, int seqno, enum xm
                p->invitestate = INV_CONFIRMED;
 
        reqprep(&resp, p, sipmethod, seqno, newbranch);
+       if (sipmethod == SIP_CANCEL && ast_test_flag(&p->flags[0], SIP_DIALOG_ANSWEREDELSEWHERE)) 
+               add_header(&resp, "Reason:", "SIP;cause=200;text=\"Call completed elsewhere\"");
+
        add_header_contentLength(&resp, 0);
        return send_request(p, &resp, reliable, seqno ? seqno : p->ocseq);
 }

diff --git a/include/asterisk/channel.h b/include/asterisk/channel.h
index 8c0b6a5..139a8b1 100644 (file)
--- a/include/asterisk/channel.h
+++ b/include/asterisk/channel.h
@@ -539,6 +539,9 @@ enum {
        /*! This is set to tell the channel not to generate DTMF begin frames, and
         *  to instead only generate END frames. */
        AST_FLAG_END_DTMF_ONLY = (1 << 14),
+       /*! Flag to show channels that this call is hangup due to the fact that the call
+           was indeed anwered, but in another channel */
+       AST_FLAG_ANSWERED_ELSEWHERE = (1 << 15),
 };
 
 /*! \brief ast_bridge_config flags */