Transition app_page to using app_confbridge internally for the conference bridge...
authorJoshua Colp <jcolp@digium.com>
Sat, 10 Mar 2012 20:06:46 +0000 (20:06 +0000)
committerJoshua Colp <jcolp@digium.com>
Sat, 10 Mar 2012 20:06:46 +0000 (20:06 +0000)
Review: https://reviewboard.asterisk.org/r/1754/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358730 65c4cc65-6c06-0410-ace0-fbb531ad65f3

CHANGES
apps/app_confbridge.c
apps/app_page.c
apps/confbridge/conf_config_parser.c
apps/confbridge/include/confbridge.h
configs/confbridge.conf.sample
include/asterisk/dial.h
main/dial.c

diff --git a/CHANGES b/CHANGES
index 10c179f..dccfb5c 100644 (file)
--- a/CHANGES
+++ b/CHANGES
@@ -39,6 +39,8 @@ ConfBridge
    occurs to be overriden using sound_participants_unmuted and sound_participants_muted.
  * Added menu action participant_count.  This will playback the number of current
    participants in a conference.
+ * Added announcement configuration option to user profile. If set the sound file will
+   be played to the user, and only the user, upon joining the conference bridge.
 
 Voicemail
 ------------------
index b5b27ed..89c4354 100644 (file)
@@ -997,6 +997,17 @@ static struct conference_bridge *join_conference_bridge(const char *name, struct
                ast_devstate_changed(AST_DEVICE_INUSE, "confbridge:%s", conference_bridge->name);
        }
 
+       /* If an announcement is to be played play it */
+       if (!ast_strlen_zero(conference_bridge_user->u_profile.announcement)) {
+               if (play_prompt_to_channel(conference_bridge,
+                                          conference_bridge_user->chan,
+                                          conference_bridge_user->u_profile.announcement)) {
+                       ao2_unlock(conference_bridge);
+                       leave_conference_bridge(conference_bridge, conference_bridge_user);
+                       return NULL;
+               }
+       }
+
        /* If the caller is a marked user or is waiting for a marked user to enter pass 'em off, otherwise pass them off to do regular joining stuff */
        if (ast_test_flag(&conference_bridge_user->u_profile, USER_OPT_MARKEDUSER | USER_OPT_WAITMARKED)) {
                if (post_join_marked(conference_bridge, conference_bridge_user)) {
index d96b4b7..f9170d4 100644 (file)
@@ -26,8 +26,7 @@
  */
 
 /*** MODULEINFO
-       <depend>dahdi</depend>
-       <depend>app_meetme</depend>
+       <depend>app_confbridge</depend>
        <support_level>core</support_level>
  ***/
 
@@ -76,7 +75,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
                                                <para>Quiet, do not play beep to caller</para>
                                        </option>
                                        <option name="r">
-                                               <para>Record the page into a file (meetme option <literal>r</literal>)</para>
+                                               <para>Record the page into a file (ConfBridge option <literal>r</literal>)</para>
                                        </option>
                                        <option name="s">
                                                <para>Only dial a channel if its device state says that it is <literal>NOT_INUSE</literal></para>
@@ -105,7 +104,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
                        destroyed when the original callers leaves.</para>
                </description>
                <see-also>
-                       <ref type="application">MeetMe</ref>
+                       <ref type="application">ConfBridge</ref>
                </see-also>
        </application>
  ***/
@@ -136,12 +135,46 @@ AST_APP_OPTIONS(page_opts, {
        AST_APP_OPTION('n', PAGE_NOCALLERANNOUNCE),
 });
 
+/* We use this structure as a way to pass this to all dialed channels */
+struct page_options {
+       char *opts[OPT_ARG_ARRAY_SIZE];
+       struct ast_flags flags;
+};
+
+static void page_state_callback(struct ast_dial *dial)
+{
+       struct ast_channel *chan;
+       struct page_options *options;
+
+       if (ast_dial_state(dial) != AST_DIAL_RESULT_ANSWERED ||
+           !(chan = ast_dial_answered(dial)) ||
+           !(options = ast_dial_get_user_data(dial))) {
+               return;
+       }
+
+       ast_func_write(chan, "CONFBRIDGE(bridge,template)", "default_bridge");
+
+       if (ast_test_flag(&options->flags, PAGE_RECORD)) {
+               ast_func_write(chan, "CONFBRIDGE(bridge,record_conference)", "yes");
+       }
+
+       ast_func_write(chan, "CONFBRIDGE(user,quiet)", "yes");
+       ast_func_write(chan, "CONFBRIDGE(user,end_marked)", "yes");
+
+       if (!ast_test_flag(&options->flags, PAGE_DUPLEX)) {
+               ast_func_write(chan, "CONFBRIDGE(user,startmuted)", "yes");
+       }
+
+       if (ast_test_flag(&options->flags, PAGE_ANNOUNCE) && !ast_strlen_zero(options->opts[OPT_ARG_ANNOUNCE])) {
+               ast_func_write(chan, "CONFBRIDGE(user,announcement)", options->opts[OPT_ARG_ANNOUNCE]);
+       }
+}
 
 static int page_exec(struct ast_channel *chan, const char *data)
 {
        char *tech, *resource, *tmp;
-       char meetmeopts[128], originator[AST_CHANNEL_NAME], *opts[OPT_ARG_ARRAY_SIZE];
-       struct ast_flags flags = { 0 };
+       char confbridgeopts[128], originator[AST_CHANNEL_NAME];
+       struct page_options options = { { 0, }, { 0, } };
        unsigned int confid = ast_random();
        struct ast_app *app;
        int res = 0, pos = 0, i = 0;
@@ -161,8 +194,8 @@ static int page_exec(struct ast_channel *chan, const char *data)
                return -1;
        }
 
-       if (!(app = pbx_findapp("MeetMe"))) {
-               ast_log(LOG_WARNING, "There is no MeetMe application available!\n");
+       if (!(app = pbx_findapp("ConfBridge"))) {
+               ast_log(LOG_WARNING, "There is no ConfBridge application available!\n");
                return -1;
        };
 
@@ -176,20 +209,14 @@ static int page_exec(struct ast_channel *chan, const char *data)
        }
 
        if (!ast_strlen_zero(args.options)) {
-               ast_app_parse_options(page_opts, &flags, opts, args.options);
+               ast_app_parse_options(page_opts, &options.flags, options.opts, args.options);
        }
 
        if (!ast_strlen_zero(args.timeout)) {
                timeout = atoi(args.timeout);
        }
 
-       if (ast_test_flag(&flags, PAGE_ANNOUNCE) && !ast_strlen_zero(opts[OPT_ARG_ANNOUNCE])) {
-               snprintf(meetmeopts, sizeof(meetmeopts), "MeetMe,%ud,%s%sqxdw(5)G(%s)", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "m"),
-                                                (ast_test_flag(&flags, PAGE_RECORD) ? "r" : ""), opts[OPT_ARG_ANNOUNCE] );
-       } else {
-               snprintf(meetmeopts, sizeof(meetmeopts), "MeetMe,%ud,%s%sqxdw(5)", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "m"),
-               (ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
-       }
+       snprintf(confbridgeopts, sizeof(confbridgeopts), "ConfBridge,%u", confid);
 
        /* Count number of extensions in list by number of ampersands + 1 */
        num_dials = 1;
@@ -222,7 +249,7 @@ static int page_exec(struct ast_channel *chan, const char *data)
                }
 
                /* Ensure device is not in use if skip option is enabled */
-               if (ast_test_flag(&flags, PAGE_SKIP)) {
+               if (ast_test_flag(&options.flags, PAGE_SKIP)) {
                        state = ast_device_state(tech);
                        if (state == AST_DEVICE_UNKNOWN) {
                                ast_log(LOG_WARNING, "Destination '%s' has device state '%s'. Paging anyway.\n", tech, ast_devstate2str(state));
@@ -247,16 +274,19 @@ static int page_exec(struct ast_channel *chan, const char *data)
                }
 
                /* Set ANSWER_EXEC as global option */
-               ast_dial_option_global_enable(dial, AST_DIAL_OPTION_ANSWER_EXEC, meetmeopts);
+               ast_dial_option_global_enable(dial, AST_DIAL_OPTION_ANSWER_EXEC, confbridgeopts);
 
                if (timeout) {
                        ast_dial_set_global_timeout(dial, timeout * 1000);
                }
 
-               if (ast_test_flag(&flags, PAGE_IGNORE_FORWARDS)) {
+               if (ast_test_flag(&options.flags, PAGE_IGNORE_FORWARDS)) {
                        ast_dial_option_global_enable(dial, AST_DIAL_OPTION_DISABLE_CALL_FORWARDING, NULL);
                }
 
+               ast_dial_set_state_callback(dial, &page_state_callback);
+               ast_dial_set_user_data(dial, &options);
+
                /* Run this dial in async mode */
                ast_dial_run(dial, chan, 1);
 
@@ -264,29 +294,32 @@ static int page_exec(struct ast_channel *chan, const char *data)
                dial_list[pos++] = dial;
        }
 
-       if (!ast_test_flag(&flags, PAGE_QUIET)) {
+       if (!ast_test_flag(&options.flags, PAGE_QUIET)) {
                res = ast_streamfile(chan, "beep", ast_channel_language(chan));
                if (!res)
                        res = ast_waitstream(chan, "");
        }
 
        if (!res) {
-               /* Default behaviour */
-               snprintf(meetmeopts, sizeof(meetmeopts), "%ud,A%s%sqxd", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "t"), 
-                       (ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
-               if (ast_test_flag(&flags, PAGE_ANNOUNCE) && !ast_strlen_zero(opts[OPT_ARG_ANNOUNCE]) &&
-                               !ast_test_flag(&flags, PAGE_NOCALLERANNOUNCE)) {
-                       snprintf(meetmeopts, sizeof(meetmeopts), "%ud,A%s%sqxdG(%s)", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "t"), 
-                         (ast_test_flag(&flags, PAGE_RECORD) ? "r" : ""), opts[OPT_ARG_ANNOUNCE] );
+               ast_func_write(chan, "CONFBRIDGE(bridge,template)", "default_bridge");
+
+               if (ast_test_flag(&options.flags, PAGE_RECORD)) {
+                       ast_func_write(chan, "CONFBRIDGE(bridge,record_conference)", "yes");
                }
-               pbx_exec(chan, app, meetmeopts);
+
+               ast_func_write(chan, "CONFBRIDGE(user,quiet)", "yes");
+               ast_func_write(chan, "CONFBRIDGE(user,marked)", "yes");
+
+               snprintf(confbridgeopts, sizeof(confbridgeopts), "%u", confid);
+
+               pbx_exec(chan, app, confbridgeopts);
        }
 
        /* Go through each dial attempt cancelling, joining, and destroying */
        for (i = 0; i < pos; i++) {
                struct ast_dial *dial = dial_list[i];
 
-               /* We have to wait for the async thread to exit as it's possible Meetme won't throw them out immediately */
+               /* We have to wait for the async thread to exit as it's possible ConfBridge won't throw them out immediately */
                ast_dial_join(dial);
 
                /* Hangup all channels */
index a73c658..0ae9ada 100644 (file)
@@ -187,6 +187,8 @@ static int set_user_option(const char *name, const char *value, struct user_prof
                ast_copy_string(u_profile->pin, value, sizeof(u_profile->pin));
        } else if (!strcasecmp(name, "music_on_hold_class")) {
                ast_copy_string(u_profile->moh_class, value, sizeof(u_profile->moh_class));
+       } else if (!strcasecmp(name, "announcement")) {
+               ast_copy_string(u_profile->announcement, value, sizeof(u_profile->announcement));
        } else if (!strcasecmp(name, "denoise")) {
                ast_set2_flag(u_profile, ast_true(value), USER_OPT_DENOISE);
        } else if (!strcasecmp(name, "dsp_talking_threshold")) {
@@ -515,6 +517,7 @@ static int parse_user(const char *cat, struct ast_config *cfg)
        u_profile->talking_threshold = DEFAULT_TALKING_THRESHOLD;
        memset(u_profile->pin, 0, sizeof(u_profile->pin));
        memset(u_profile->moh_class, 0, sizeof(u_profile->moh_class));
+       memset(u_profile->announcement, 0, sizeof(u_profile->announcement));
        for (var = ast_variable_browse(cfg, cat); var; var = var->next) {
                if (!strcasecmp(var->name, "type")) {
                        continue;
@@ -859,6 +862,8 @@ static char *handle_cli_confbridge_show_user_profile(struct ast_cli_entry *e, in
        ast_cli(a->fd,"MOH Class:               %s\n",
                ast_strlen_zero(u_profile.moh_class) ?
                "default" : u_profile.moh_class);
+       ast_cli(a->fd,"Announcement:            %s\n",
+               u_profile.announcement);
        ast_cli(a->fd,"Quiet:                   %s\n",
                u_profile.flags & USER_OPT_QUIET ?
                "enabled" : "disabled");
index 5337e22..dd4ceff 100644 (file)
@@ -128,6 +128,7 @@ struct user_profile {
        char name[128];
        char pin[MAX_PIN];
        char moh_class[128];
+       char announcement[PATH_MAX];
        unsigned int flags;
        unsigned int announce_user_count_all_after;
        /*! The time in ms of talking before a user is considered to be talking by the dsp. */
index d113825..7484b28 100644 (file)
@@ -126,6 +126,7 @@ type=user
                          ; the conference. This option is off by default.
 ;dtmf_passthrough=yes  ; Sets whether or not DTMF should pass through the conference.
                        ; This option is off by default.
+;announcement=</path/to/file> ; Play a sound file to the user when they join the conference.
 
 ; --- ConfBridge Bridge Profile Options ---
 [default_bridge]
index e023fb3..04721d2 100644 (file)
@@ -152,6 +152,19 @@ int ast_dial_option_disable(struct ast_dial *dial, int num, enum ast_dial_option
  */
 void ast_dial_set_state_callback(struct ast_dial *dial, ast_dial_state_callback callback);
 
+/*! \brief Set user data on a dial structure
+ * \param dial The dial structure to set a user data pointer on
+ * \param user_data The user data pointer
+ * \return nothing
+ */
+void ast_dial_set_user_data(struct ast_dial *dial, void *user_data);
+
+/*! \brief Return the user data on a dial structure
+ * \param dial The dial structure
+ * \return A pointer to the user data
+ */
+void *ast_dial_get_user_data(struct ast_dial *dial);
+
 /*! \brief Set the maximum time (globally) allowed for trying to ring phones
  * \param dial The dial structure to apply the time limit to
  * \param timeout Maximum time allowed in milliseconds
index 24dbf28..5384717 100644 (file)
@@ -47,6 +47,7 @@ struct ast_dial {
        enum ast_dial_result state;                        /*!< Status of dial */
        void *options[AST_DIAL_OPTION_MAX];                /*!< Global options */
        ast_dial_state_callback state_callback;            /*!< Status callback */
+       void *user_data;                                   /*!< Attached user data */
        AST_LIST_HEAD(, ast_dial_channel) channels; /*!< Channels being dialed */
        pthread_t thread;                                  /*!< Thread (if running in async) */
        ast_mutex_t lock;                                  /*! Lock to protect the thread information above */
@@ -1049,6 +1050,16 @@ void ast_dial_set_state_callback(struct ast_dial *dial, ast_dial_state_callback
        dial->state_callback = callback;
 }
 
+void ast_dial_set_user_data(struct ast_dial *dial, void *user_data)
+{
+       dial->user_data = user_data;
+}
+
+void *ast_dial_get_user_data(struct ast_dial *dial)
+{
+       return dial->user_data;
+}
+
 /*! \brief Set the maximum time (globally) allowed for trying to ring phones
  * \param dial The dial structure to apply the time limit to
  * \param timeout Maximum time allowed