Merge branch 'fsorig'

Conflicts:
	libs/freetdm/src/ftmod/ftmod_sangoma_isdn/ftmod_sangoma_isdn_stack_cfg.c
	libs/freetdm/src/ftmod/ftmod_sangoma_isdn/ftmod_sangoma_isdn_stack_hndl.c

src/include/private/switch_core_pvt.h

@@ -195,9 +195,9 @@ struct switch_media_bug {
 	char *target;
 	switch_codec_implementation_t read_impl;
 	switch_codec_implementation_t write_impl;
-	switch_timer_t timer;
-	uint32_t last_read_ts;
-	uint32_t last_write_ts;
+	uint32_t record_frame_size;
+	uint32_t record_pre_buffer_count;
+	uint32_t record_pre_buffer_max;
 	switch_frame_t *ping_frame;
 	struct switch_media_bug *next;
 };
@@ -260,6 +260,7 @@ struct switch_runtime {
 	uint32_t max_db_handles;
 	uint32_t db_handle_timeout;
 	int cpu_count;
+	uint32_t time_sync;
 };
 
 extern struct switch_runtime runtime;

src/include/switch_channel.h

@@ -403,6 +403,7 @@ SWITCH_DECLARE(void) switch_channel_clear_flag_recursive(switch_channel_t *chann
 SWITCH_DECLARE(switch_status_t) switch_channel_perform_answer(switch_channel_t *channel, const char *file, const char *func, int line);
 
 SWITCH_DECLARE(switch_status_t) switch_channel_perform_mark_answered(switch_channel_t *channel, const char *file, const char *func, int line);
+SWITCH_DECLARE(void) switch_channel_check_zrtp(switch_channel_t *channel);
 
 /*!
   \brief Answer a channel (initiate/acknowledge a successful connection)

src/include/switch_core.h

@@ -217,6 +217,8 @@ SWITCH_DECLARE(uint32_t) switch_core_media_bug_clear_flag(_In_ switch_media_bug_
 */
 SWITCH_DECLARE(void) switch_core_media_bug_set_read_replace_frame(_In_ switch_media_bug_t *bug, _In_ switch_frame_t *frame);
 
+SWITCH_DECLARE(uint32_t) switch_core_cpu_count(void);
+
 /*!
   \brief Remove a media bug from the session
   \param session the session to remove the bug from
@@ -270,6 +272,8 @@ SWITCH_DECLARE(void) switch_core_media_bug_flush(_In_ switch_media_bug_t *bug);
 */
 SWITCH_DECLARE(switch_status_t) switch_core_media_bug_flush_all(_In_ switch_core_session_t *session);
 
+SWITCH_DECLARE(switch_status_t) switch_core_media_bug_set_pre_buffer_framecount(switch_media_bug_t *bug, uint32_t framecount);
+
 ///\}
 
 ///\defgroup pa1 Port Allocation

src/include/switch_rtp.h

@@ -196,6 +196,7 @@ SWITCH_DECLARE(switch_status_t) switch_rtp_set_local_address(switch_rtp_t *rtp_s
 SWITCH_DECLARE(void) switch_rtp_kill_socket(switch_rtp_t *rtp_session);
 
 SWITCH_DECLARE(void) switch_rtp_break(switch_rtp_t *rtp_session);
+SWITCH_DECLARE(void) switch_rtp_flush(switch_rtp_t *rtp_session);
 
 /*! 
   \brief Test if an RTP session is ready

src/include/switch_types.h

@@ -314,7 +314,8 @@ typedef enum {
 	SCF_CLEAR_SQL = (1 << 17),
 	SCF_THREADED_SYSTEM_EXEC = (1 << 18),
 	SCF_SYNC_CLOCK_REQUESTED = (1 << 19),
-	SCF_CORE_ODBC_REQ = (1 << 20)
+	SCF_CORE_ODBC_REQ = (1 << 20),
+	SCF_DEBUG_SQL = (1 << 21)
 } switch_core_flag_enum_t;
 typedef uint32_t switch_core_flag_t;
 
@@ -1208,6 +1209,8 @@ typedef enum {
 	CF_VIDEO_REFRESH_REQ,
 	CF_SERVICE_AUDIO,
 	CF_SERVICE_VIDEO,
+	CF_ZRTP_HASH,
+	CF_ZRTP_PASS,
 	/* WARNING: DO NOT ADD ANY FLAGS BELOW THIS LINE */
 	/* IF YOU ADD NEW ONES CHECK IF THEY SHOULD PERSIST OR ZERO THEM IN switch_core_session.c switch_core_session_request_xml() */
 	CF_FLAG_MAX
@@ -1741,7 +1744,8 @@ typedef enum {
 	SCSC_PAUSE_CHECK,
 	SCSC_READY_CHECK,
 	SCSC_THREADED_SYSTEM_EXEC,
-	SCSC_SYNC_CLOCK_WHEN_IDLE
+	SCSC_SYNC_CLOCK_WHEN_IDLE,
+	SCSC_DEBUG_SQL,
 } switch_session_ctl_t;
 
 typedef enum {

src/mod/applications/mod_abstraction/conf/autoload_configs/abstraction.conf.xml

@@ -0,0 +1,5 @@
+<configuration name="abstraction.conf" description="Abstraction">
+<apis>
+	<api name="user_name" description="Return Name for extension" syntax="<exten>" parse="(.*)" destination="user_data" argument="$1@default var effective_caller_id_name"/>
+</apis>
+</configuration>

src/mod/applications/mod_blacklist/conf/autoload_configs/blacklist.conf.xml

@@ -0,0 +1,11 @@
+<configuration name="mod_blacklist.conf" description="Blacklist module">
+<lists>
+	<!--
+		Example blacklist, the referenced file contains blacklisted items, one entry per line
+
+		NOTE: make sure the file exists and is readable by FreeSWITCH.
+
+	<list name="example" filename="/usr/local/freeswitch/conf/blacklists/example.list"/>
+	-->
+</lists>
+</configuration>

src/mod/applications/mod_callcenter/conf/autoload_configs/callcenter.conf.xml

@@ -0,0 +1,38 @@
+<configuration name="callcenter.conf" description="CallCenter">
+  <settings>
+    <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
+    <!--<param name="dbname" value="/dev/shm/callcenter.db"/>-->
+  </settings>
+
+  <queues>
+
+    <queue name="support@default">
+      <param name="strategy" value="longest-idle-agent"/>
+      <param name="moh-sound" value="$${hold_music}"/>
+      <!--<param name="record-template" value="$${base_dir}/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}.${destination_number}.${caller_id_number}.${uuid}.wav"/>-->
+      <param name="time-base-score" value="system"/>
+      <param name="max-wait-time" value="0"/>
+      <param name="max-wait-time-with-no-agent" value="0"/>
+      <param name="max-wait-time-with-no-agent-time-reached" value="5"/>
+      <param name="tier-rules-apply" value="false"/>
+      <param name="tier-rule-wait-second" value="300"/>
+      <param name="tier-rule-wait-multiply-level" value="true"/>
+      <param name="tier-rule-no-agent-no-wait" value="false"/>
+      <param name="discard-abandoned-after" value="60"/>
+      <param name="abandoned-resume-allowed" value="false"/>
+    </queue>
+
+  </queues>
+
+<!-- WARNING: Configuration of XML Agents will be updated into the DB upon restart. -->
+<!-- WARNING: Configuration of XML Tiers will reset the level and position if those were supplied. -->
+<!-- WARNING: Agents and Tiers XML config shouldn't be used in a multi FS shared DB setup (Not currently supported anyway) -->
+  <agents>
+    <!--<agent name="1000@default" type="callback" contact="[call_timeout=10]user/1000@default" status="Available" max-no-answer="3" wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" />-->
+  </agents>
+  <tiers>
+    <!-- If no level or position is provided, they will default to 1.  You should do this to keep db value on restart. -->
+    <!-- <tier agent="1000@default" queue="support@default" level="1" position="1"/> -->
+  </tiers>
+
+</configuration> 

src/mod/applications/mod_callcenter/mod_callcenter.c

@@ -1371,6 +1371,22 @@ end:
 	return status;
 }
 
+static void playback_array(switch_core_session_t *session, const char *str) {
+	if (str && !strncmp(str, "ARRAY::", 7)) {
+		char *i = (char*) str + 7, *j = i;
+		while (1) {
+			if ((j = strstr(i, "::"))) {
+				*j = 0;
+			}
+			switch_ivr_play_file(session, NULL, i, NULL);
+			if (!j) break;
+			i = j + 2;
+		}
+	} else {
+		switch_ivr_play_file(session, NULL, str, NULL);
+	}
+}
+
 static void *SWITCH_THREAD_FUNC outbound_agent_thread_run(switch_thread_t *thread, void *obj)
 {
 	struct call_helper *h = (struct call_helper *) obj;
@@ -1429,6 +1445,14 @@ static void *SWITCH_THREAD_FUNC outbound_agent_thread_run(switch_thread_t *threa
 
 	/* CallBack Mode */
 	if (!strcasecmp(h->agent_type, CC_AGENT_TYPE_CALLBACK)) {
+		switch_channel_t *member_channel = switch_core_session_get_channel(member_session);
+		char *cid_name = NULL;
+		const char *cid_name_prefix = NULL;
+		if ((cid_name_prefix = switch_channel_get_variable(member_channel, "cc_outbound_cid_name_prefix"))) {
+			cid_name = switch_mprintf("%s%s", cid_name_prefix, h->member_cid_name);
+			switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(member_session), SWITCH_LOG_DEBUG, "Setting outbound caller_id_name to: %s\n", cid_name);
+		}
+
 		switch_event_create(&ovars, SWITCH_EVENT_REQUEST_PARAMS);
 		switch_event_add_header(ovars, SWITCH_STACK_BOTTOM, "cc_queue", "%s", h->queue_name);
 		switch_event_add_header(ovars, SWITCH_STACK_BOTTOM, "cc_member_uuid", "%s", h->member_uuid);
@@ -1444,9 +1468,9 @@ static void *SWITCH_THREAD_FUNC outbound_agent_thread_run(switch_thread_t *threa
 
 		t_agent_called = local_epoch_time_now(NULL);
 		dialstr = switch_mprintf("%s", h->originate_string);
-		status = switch_ivr_originate(NULL, &agent_session, &cause, dialstr, 60, NULL, h->member_cid_name, h->member_cid_number, NULL, ovars, SOF_NONE, NULL);
-
+		status = switch_ivr_originate(NULL, &agent_session, &cause, dialstr, 60, NULL, cid_name ? cid_name : h->member_cid_name, h->member_cid_number, NULL, ovars, SOF_NONE, NULL);
 		switch_safe_free(dialstr);
+		switch_safe_free(cid_name);
 
 		switch_event_destroy(&ovars);
 	/* UUID Standby Mode */
@@ -1489,6 +1513,7 @@ static void *SWITCH_THREAD_FUNC outbound_agent_thread_run(switch_thread_t *threa
 		switch_channel_t *member_channel = switch_core_session_get_channel(member_session);
 		switch_channel_t *agent_channel = switch_core_session_get_channel(agent_session);
 		const char *other_loopback_leg_uuid = switch_channel_get_variable(agent_channel, "other_loopback_leg_uuid");
+		const char *o_announce = NULL;
 
 		switch_channel_set_variable(agent_channel, "cc_member_pre_answer_uuid", NULL);
 
@@ -1616,6 +1641,10 @@ static void *SWITCH_THREAD_FUNC outbound_agent_thread_run(switch_thread_t *threa
 		switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(member_session), SWITCH_LOG_DEBUG, "Agent %s answered \"%s\" <%s> from queue %s%s\n",
 				h->agent_name, h->member_cid_name, h->member_cid_number, h->queue_name, (h->record_template?" (Recorded)":""));
 
+		if ((o_announce = switch_channel_get_variable(member_channel, "cc_outbound_announce"))) {
+			playback_array(agent_session, o_announce);
+		}
+
 		switch_ivr_uuid_bridge(h->member_session_uuid, switch_core_session_get_uuid(agent_session));
 
 		switch_channel_set_variable(member_channel, "cc_agent_uuid", agent_uuid);
@@ -2340,6 +2369,7 @@ void *SWITCH_THREAD_FUNC cc_member_thread_run(switch_thread_t *thread, void *obj
 
 struct moh_dtmf_helper {
 	const char *queue_name;
+	const char *exit_keys;
 	char dtmf;
 };
 
@@ -2348,10 +2378,9 @@ static switch_status_t moh_on_dtmf(switch_core_session_t *session, void *input,
 
 	switch (itype) {
 		case SWITCH_INPUT_TYPE_DTMF:
-			{
-				/* Just laywork for people who want to get some DTMF actions */
+			if (h->exit_keys && *(h->exit_keys)) {
 				switch_dtmf_t *dtmf = (switch_dtmf_t *) input;
-				if (strchr("#", dtmf->digit)) {
+				if (strchr(h->exit_keys, dtmf->digit)) {
 					h->dtmf = dtmf->digit;
 					return SWITCH_STATUS_BREAK;
 				}
@@ -2552,6 +2581,7 @@ SWITCH_STANDARD_APP(callcenter_function)
 		switch_input_args_t args = { 0 };
 		struct moh_dtmf_helper ht;
 
+		ht.exit_keys = switch_channel_get_variable(member_channel, "cc_exit_keys");
 		ht.dtmf = '\0';
 		args.input_callback = moh_on_dtmf;
 		args.buf = (void *) &ht;
@@ -2575,17 +2605,21 @@ SWITCH_STANDARD_APP(callcenter_function)
 				/* Sadly, there doesn't seem to be a return to switch_ivr_play_file that tell you the file wasn't found.  FALSE also mean that the channel got switch to BRAKE state, so we check for read acceptable */
 				switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(member_session), SWITCH_LOG_WARNING, "Couldn't play file '%s', continuing wait with no audio\n", cur_moh);
 				moh_valid = SWITCH_FALSE;
-
+			} else if (status == SWITCH_STATUS_BREAK) {
+				char buf[2] = { ht.dtmf, 0 };
+				switch_channel_set_variable(member_channel, "cc_exit_key", buf);
+				break;
 			} else if (!SWITCH_READ_ACCEPTABLE(status)) {
 				break;
 			}
-
 		} else {
-			switch_ivr_collect_digits_callback(session, &args, 0, 0);
+			if ((switch_ivr_collect_digits_callback(member_session, &args, 0, 0)) == SWITCH_STATUS_BREAK) {
+				char buf[2] = { ht.dtmf, 0 };
+				switch_channel_set_variable(member_channel, "cc_exit_key", buf);
+				break;
+			}
 		}
-
 		switch_yield(1000);
-
 	}
 
 	/* Make sure an agent was found, as we might break above without setting it */

src/mod/applications/mod_cidlookup/conf/autoload_configs/cidlookup.conf.xml

@@ -0,0 +1,33 @@
+<configuration name="cidlookup.conf" description="cidlookup Configuration">
+  <settings>
+    <!-- comment out url to not setup a url based lookup -->
+    <param name="url" value="http://query.voipcnam.com/query.php?api_key=MYAPIKEY&number=${caller_id_number}"/>
+
+    <!-- comment out whitepages-apikey to not use whitepages.com, you must
+         get an API key from http://developer.whitepages.com/ -->
+    <param name="whitepages-apikey" value="MYAPIKEY"/>
+
+    <!-- set to false to not cache (in memcache) results from the url query -->
+    <param name="cache" value="true"/>
+    <!-- expire is in seconds -->
+    <param name="cache-expire" value="86400"/>
+
+    <param name="odbc-dsn" value="phone:phone:phone"/>
+
+    <!-- comment out sql to not setup a database (directory) lookup -->
+    <param name="sql" value="
+     SELECT name||' ('||type||')' AS name 
+      FROM phonebook p JOIN numbers n ON p.id = n.phonebook_id
+      WHERE n.number='${caller_id_number}' 
+      LIMIT 1
+      "/>
+    <!-- comment out citystate-sql to not setup a database (city/state) 
+         lookup -->
+    <param name="citystate-sql" value="
+     SELECT ratecenter||' '||state as name
+      FROM npa_nxx_company_ocn
+      WHERE npa = ${caller_id_number:1:3} AND nxx = ${caller_id_number:4:3}
+      LIMIT 1
+      "/>
+  </settings>
+</configuration>

src/mod/applications/mod_commands/mod_commands.c

@@ -546,24 +546,19 @@ SWITCH_STANDARD_API(timer_test_function)
 		goto end;
 	}
 
-	switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CONSOLE, "Timer Test: samplecount after init: %d\n", timer.samplecount);
+	switch_core_timer_next(&timer); /* Step timer once before testing results below, to get first timestamp as accurate as possible */
 
-	/* Step timer once before testing results below, to get first timestamp as accurate as possible */
-	switch_core_timer_next(&timer);
+	start = then = switch_time_ref();
 
-	switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CONSOLE, "Timer Test: samplecount after first step: %d\n", timer.samplecount);
-
-	start = switch_time_ref();
 	for (x = 1; x <= max; x++) {
-		then = switch_time_ref();
 		switch_core_timer_next(&timer);
 		now = switch_time_ref();
 		diff = (int) (now - then);
-		//stream->write_function(stream, "test %d sleep %ld %d\n", x, mss, diff);
-		switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CONSOLE, "Timer Test: %d sleep %d %d\n", x, mss, diff);
 		total += diff;
+		then = now;
+		switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CONSOLE, "Timer Test: %d sleep %d %d\n", x, mss, diff);
 	}
-	end = switch_time_ref();
+	end = then;
 
 	switch_yield(250000);
 
@@ -1913,6 +1908,12 @@ SWITCH_STANDARD_API(ctl_function)
 			}
 			switch_core_session_ctl(command, &arg);
 			stream->write_function(stream, "+OK\n");
+
+		} else if (!strcasecmp(argv[0], "debug_sql")) {
+			int x = 0;
+			switch_core_session_ctl(SCSC_DEBUG_SQL, &x);
+			stream->write_function(stream, "+OK SQL DEBUG [%s]\n", x ? "on" : "off");			
+
 		} else if (!strcasecmp(argv[0], "reclaim_mem")) {
 			switch_core_session_ctl(SCSC_RECLAIM, &arg);
 			stream->write_function(stream, "+OK\n");
@@ -2379,7 +2380,7 @@ SWITCH_STANDARD_API(dual_transfer_function)
 		}
 	}
 
-	if ((dp2 = strstr(dest1, "/inline")) && *(dp2 + 7) == '\0') {
+	if ((dp2 = strstr(dest2, "/inline")) && *(dp2 + 7) == '\0') {
 		*dp2++ = '\0';
 	} else {
 		if ((dp2 = strchr(dest2, '/'))) {
@@ -5312,6 +5313,27 @@ SWITCH_STANDARD_API(log_function)
 	return SWITCH_STATUS_SUCCESS;
 }
 
+SWITCH_STANDARD_API(file_exists_function)
+{
+	if (!zstr(cmd)) {
+		switch_memory_pool_t *pool;
+
+		switch_core_new_memory_pool(&pool);
+
+		if (switch_file_exists(cmd, pool) == SWITCH_STATUS_SUCCESS) {
+			stream->write_function(stream, "true");
+		} else {
+			stream->write_function(stream, "false");
+		}
+
+		switch_core_destroy_memory_pool(&pool);
+	} else {
+		stream->write_function(stream, "false");
+	}
+
+	return SWITCH_STATUS_SUCCESS;
+}
+
 SWITCH_MODULE_LOAD_FUNCTION(mod_commands_load)
 {
 	switch_api_interface_t *commands_api_interface;
@@ -5436,6 +5458,7 @@ SWITCH_MODULE_LOAD_FUNCTION(mod_commands_load)
 	SWITCH_ADD_API(commands_api_interface, "xml_flush_cache", "clear xml cache", xml_flush_function, "<id> <key> <val>");
 	SWITCH_ADD_API(commands_api_interface, "xml_locate", "find some xml", xml_locate_function, "[root | <section> <tag> <tag_attr_name> <tag_attr_val>]");
 	SWITCH_ADD_API(commands_api_interface, "xml_wrap", "Wrap another api command in xml", xml_wrap_api_function, "<command> <args>");
+	SWITCH_ADD_API(commands_api_interface, "file_exists", "check if a file exists on server", file_exists_function, "<file>");
 
 
 	switch_console_set_complete("add alias add");
@@ -5444,6 +5467,7 @@ SWITCH_MODULE_LOAD_FUNCTION(mod_commands_load)
 	switch_console_set_complete("add complete del");
 	switch_console_set_complete("add db_cache status");
 	switch_console_set_complete("add fsctl debug_level");
+	switch_console_set_complete("add fsctl debug_sql");
 	switch_console_set_complete("add fsctl last_sps");
 	switch_console_set_complete("add fsctl default_dtmf_duration");
 	switch_console_set_complete("add fsctl hupall");
@@ -5577,6 +5601,7 @@ SWITCH_MODULE_LOAD_FUNCTION(mod_commands_load)
 	switch_console_set_complete("add version");
 	switch_console_set_complete("add uuid_warning ::console::list_uuid");
 	switch_console_set_complete("add ...");
+	switch_console_set_complete("add file_exists");
 
 
 	/* indicate that the module should continue to be loaded */

src/mod/applications/mod_conference/conf/autoload_configs/conference.conf.xml

@@ -0,0 +1,209 @@
+<!-- http://wiki.freeswitch.org/wiki/Mod_conference --> 
+<!-- None of these paths are real if you want any of these options you need to really set them up -->
+<configuration name="conference.conf" description="Audio Conference">
+  <!-- Advertise certain presence on startup . -->
+  <advertise>
+    <room name="3001@$${domain}" status="FreeSWITCH"/>
+  </advertise>
+
+  <!-- These are the default keys that map when you do not specify a caller control group -->	
+  <!-- Note: none and default are reserved names for group names.  Disabled if dist-dtmf member flag is set. -->	
+  <caller-controls>
+    <group name="default">
+      <control action="mute" digits="0"/>
+      <control action="deaf mute" digits="*"/>
+      <control action="energy up" digits="9"/>
+      <control action="energy equ" digits="8"/>
+      <control action="energy dn" digits="7"/>
+      <control action="vol talk up" digits="3"/>
+      <control action="vol talk zero" digits="2"/>
+      <control action="vol talk dn" digits="1"/>
+      <control action="vol listen up" digits="6"/>
+      <control action="vol listen zero" digits="5"/>
+      <control action="vol listen dn" digits="4"/>
+      <control action="hangup" digits="#"/>
+    </group>
+  </caller-controls>
+
+  <!-- Profiles are collections of settings you can reference by name. -->
+  <profiles>
+    <!--If no profile is specified it will default to "default"-->
+    <profile name="default">
+      <!-- Directory to drop CDR's 
+	   'auto' means $PREFIX/logs/conference_cdr/<confernece_uuid>.cdr.xml
+	   a non-absolute path means $PREFIX/logs/<value>/<confernece_uuid>.cdr.xml
+	   absolute path means <value>/<confernece_uuid>.cdr.xml
+      -->
+      <!-- <param name="cdr-log-dir" value="auto"/> -->
+      
+      <!-- Domain (for presence) -->
+      <param name="domain" value="$${domain}"/>
+      <!-- Sample Rate-->
+      <param name="rate" value="8000"/>
+      <!-- Number of milliseconds per frame -->
+      <param name="interval" value="20"/>
+      <!-- Energy level required for audio to be sent to the other users -->
+      <param name="energy-level" value="300"/>
+
+      <!--Can be | delim of waste|mute|deaf|dist-dtmf waste will always transmit data to each channel
+          even during silence.  dist-dtmf propagates dtmfs to all other members, but channel controls
+	  via dtmf will be disabled. -->
+      <!--<param name="member-flags" value="waste"/>-->
+
+      <!-- Name of the caller control group to use for this profile -->
+      <!-- <param name="caller-controls" value="some name"/> -->
+      <!-- Name of the caller control group to use for the moderator in this profile -->
+      <!-- <param name="moderator-controls" value="some name"/> -->
+      <!-- TTS Engine to use -->
+      <!--<param name="tts-engine" value="cepstral"/>-->
+      <!-- TTS Voice to use -->
+      <!--<param name="tts-voice" value="david"/>-->
+
+      <!-- If TTS is enabled all audio-file params beginning with -->
+      <!-- 'say:' will be considered text to say with TTS -->
+      <!-- Override the default path here, after which you use relative paths in the other sound params -->
+      <!-- Note: The default path is the conference's first caller's sound_prefix -->
+      <!--<param name="sound-prefix" value="$${sounds_dir}/en/us/callie"/>-->
+      <!-- File to play to acknowledge succees -->
+      <!--<param name="ack-sound" value="beep.wav"/>-->
+      <!-- File to play to acknowledge failure -->
+      <!--<param name="nack-sound" value="beeperr.wav"/>-->
+      <!-- File to play to acknowledge muted -->
+      <param name="muted-sound" value="conference/conf-muted.wav"/>
+      <!-- File to play to acknowledge unmuted -->
+      <param name="unmuted-sound" value="conference/conf-unmuted.wav"/>
+      <!-- File to play if you are alone in the conference -->
+      <param name="alone-sound" value="conference/conf-alone.wav"/>
+      <!-- File to play endlessly (nobody will ever be able to talk) -->
+      <!--<param name="perpetual-sound" value="perpetual.wav"/>-->
+      <!-- File to play when you're alone (music on hold)-->
+      <param name="moh-sound" value="$${hold_music}"/>
+      <!-- File to play when you join the conference -->
+      <param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/>
+      <!-- File to play when you leave the conference -->
+      <param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/>
+      <!-- File to play when you ae ejected from the conference -->
+      <param name="kicked-sound" value="conference/conf-kicked.wav"/>
+      <!-- File to play when the conference is locked -->
+      <param name="locked-sound" value="conference/conf-locked.wav"/>
+      <!-- File to play when the conference is locked during the call-->
+      <param name="is-locked-sound" value="conference/conf-is-locked.wav"/>
+      <!-- File to play when the conference is unlocked during the call-->
+      <param name="is-unlocked-sound" value="conference/conf-is-unlocked.wav"/>
+      <!-- File to play to prompt for a pin -->
+      <param name="pin-sound" value="conference/conf-pin.wav"/>
+      <!-- File to play to when the pin is invalid -->
+      <param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/>
+      <!-- Conference pin -->
+      <!--<param name="pin" value="12345"/>-->
+      <!--<param name="moderator-pin" value="54321"/>-->
+      <!-- Max number of times the user can be prompted for PIN -->
+      <!--<param name="pin-retries" value="3"/>-->
+      <!-- Default Caller ID Name for outbound calls -->
+      <param name="caller-id-name" value="$${outbound_caller_name}"/>
+      <!-- Default Caller ID Number for outbound calls -->
+      <param name="caller-id-number" value="$${outbound_caller_id}"/>
+      <!-- Suppress start and stop talking events -->
+      <!-- <param name="suppress-events" value="start-talking,stop-talking"/> -->
+      <!-- enable comfort noise generation -->
+      <param name="comfort-noise" value="true"/>
+      <!-- Uncomment auto-record to toggle recording every conference call. -->
+      <!-- Another valid value is   shout://user:pass@server.com/live.mp3   -->
+      <!--
+      <param name="auto-record" value="$${recordings_dir}/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
+      -->
+
+      <!-- IVR digit machine timeouts -->
+      <!-- How much to wait between DTMF digits to match caller-controls -->
+      <!-- <param name="ivr-dtmf-timeout" value="500"/> -->
+      <!-- How much to wait for the first DTMF, 0 forever -->
+      <!-- <param name="ivr-input-timeout" value="0" /> -->
+      <!-- Delay before a conference is asked to be terminated -->
+      <!-- <param name="endconf-grace-time" value="120" /> -->
+    </profile>
+
+    <profile name="wideband">
+      <param name="domain" value="$${domain}"/>
+      <param name="rate" value="16000"/>
+      <param name="interval" value="20"/>
+      <param name="energy-level" value="300"/>
+      <!--<param name="sound-prefix" value="$${sounds_dir}/en/us/callie"/>-->
+      <param name="muted-sound" value="conference/conf-muted.wav"/>
+      <param name="unmuted-sound" value="conference/conf-unmuted.wav"/>
+      <param name="alone-sound" value="conference/conf-alone.wav"/>
+      <param name="moh-sound" value="$${hold_music}"/>
+      <param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/>
+      <param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/>
+      <param name="kicked-sound" value="conference/conf-kicked.wav"/>
+      <param name="locked-sound" value="conference/conf-locked.wav"/>
+      <param name="is-locked-sound" value="conference/conf-is-locked.wav"/>
+      <param name="is-unlocked-sound" value="conference/conf-is-unlocked.wav"/>
+      <param name="pin-sound" value="conference/conf-pin.wav"/>
+      <param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/>
+      <param name="caller-id-name" value="$${outbound_caller_name}"/>
+      <param name="caller-id-number" value="$${outbound_caller_id}"/>
+      <param name="comfort-noise" value="true"/>
+      <!--<param name="tts-engine" value="flite"/>-->
+      <!--<param name="tts-voice" value="kal16"/>-->
+    </profile>
+
+    <profile name="ultrawideband">
+      <param name="domain" value="$${domain}"/>
+      <param name="rate" value="32000"/>
+      <param name="interval" value="20"/>
+      <param name="energy-level" value="300"/>
+      <!--<param name="sound-prefix" value="$${sounds_dir}/en/us/callie"/>-->
+      <param name="muted-sound" value="conference/conf-muted.wav"/>
+      <param name="unmuted-sound" value="conference/conf-unmuted.wav"/>
+      <param name="alone-sound" value="conference/conf-alone.wav"/>
+      <param name="moh-sound" value="$${hold_music}"/>
+      <param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/>
+      <param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/>
+      <param name="kicked-sound" value="conference/conf-kicked.wav"/>
+      <param name="locked-sound" value="conference/conf-locked.wav"/>
+      <param name="is-locked-sound" value="conference/conf-is-locked.wav"/>
+      <param name="is-unlocked-sound" value="conference/conf-is-unlocked.wav"/>
+      <param name="pin-sound" value="conference/conf-pin.wav"/>
+      <param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/>
+      <param name="caller-id-name" value="$${outbound_caller_name}"/>
+      <param name="caller-id-number" value="$${outbound_caller_id}"/>
+      <param name="comfort-noise" value="true"/>
+      <!--<param name="tts-engine" value="flite"/>-->
+      <!--<param name="tts-voice" value="kal16"/>-->
+    </profile>
+
+    <profile name="cdquality">
+      <param name="domain" value="$${domain}"/>
+      <param name="rate" value="48000"/>
+      <param name="interval" value="10"/>
+      <param name="energy-level" value="300"/>
+      <!--<param name="sound-prefix" value="$${sounds_dir}/en/us/callie"/>-->
+      <param name="muted-sound" value="conference/conf-muted.wav"/>
+      <param name="unmuted-sound" value="conference/conf-unmuted.wav"/>
+      <param name="alone-sound" value="conference/conf-alone.wav"/>
+      <param name="moh-sound" value="$${hold_music}"/>
+      <param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/>
+      <param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/>
+      <param name="kicked-sound" value="conference/conf-kicked.wav"/>
+      <param name="locked-sound" value="conference/conf-locked.wav"/>
+      <param name="is-locked-sound" value="conference/conf-is-locked.wav"/>
+      <param name="is-unlocked-sound" value="conference/conf-is-unlocked.wav"/>
+      <param name="pin-sound" value="conference/conf-pin.wav"/>
+      <param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/>
+      <param name="caller-id-name" value="$${outbound_caller_name}"/>
+      <param name="caller-id-number" value="$${outbound_caller_id}"/>
+      <param name="comfort-noise" value="true"/>
+    </profile>
+
+    <profile name="sla">
+      <param name="domain" value="$${domain}"/>
+      <param name="rate" value="16000"/>
+      <param name="interval" value="20"/>
+      <param name="caller-controls" value="none"/>
+      <param name="energy-level" value="200"/>
+      <param name="moh-sound" value="silence"/>
+      <param name="comfort-noise" value="true"/>
+    </profile>
+
+  </profiles>
+</configuration>

src/mod/applications/mod_conference/conf/dialplan/conference_dialplan.xml

@@ -0,0 +1,31 @@
+	<!--
+		start a dynamic conference with the settings of the "default" conference profile in conference.conf.xml
+	-->
+	<extension name="nb_conferences">
+		<condition field="destination_number" expression="^(30\d{2})$">
+			<action application="answer"/>
+			<action application="conference" data="$1-${domain_name}@default"/>
+		</condition>
+	</extension>
+
+	<extension name="wb_conferences">
+		<condition field="destination_number" expression="^(31\d{2})$">
+			<action application="answer"/>
+			<action application="conference" data="$1-${domain_name}@wideband"/>
+		</condition>
+	</extension>
+
+	<extension name="uwb_conferences">
+		<condition field="destination_number" expression="^(32\d{2})$">
+			<action application="answer"/>
+			<action application="conference" data="$1-${domain_name}@ultrawideband"/>
+		</condition>
+	</extension>
+
+	<!-- MONO 48kHz conferences -->
+	<extension name="cdquality_conferences">
+		<condition field="destination_number" expression="^(33\d{2})$">
+			<action application="answer"/>
+			<action application="conference" data="$1-${domain_name}@cdquality"/>
+		</condition>
+	</extension>

src/mod/applications/mod_conference/mod_conference.c

@@ -1104,6 +1104,7 @@ static switch_status_t conference_del_member(conference_obj_t *conference, confe
 
 		if (test_eflag(conference, EFLAG_FLOOR_CHANGE)) {
 			switch_event_create_subclass(&event, SWITCH_EVENT_CUSTOM, CONF_EVENT_MAINT);
+			conference_add_event_data(conference, event); 
 			switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "Action", "floor-change");
 			switch_event_add_header(event, SWITCH_STACK_BOTTOM, "Old-ID", "%d", member->id);
 			switch_event_add_header(event, SWITCH_STACK_BOTTOM, "New-ID", "none");
@@ -1257,15 +1258,17 @@ static void *SWITCH_THREAD_FUNC conference_video_thread_run(switch_thread_t *thr
 		}
 
 		session = conference->floor_holder->session;
-		switch_core_session_read_lock(session);
-		switch_mutex_unlock(conference->mutex);
-		if (!switch_channel_ready(switch_core_session_get_channel(session))) {
-			status = SWITCH_STATUS_FALSE;
-		} else {
-			status = switch_core_session_read_video_frame(session, &vid_frame, SWITCH_IO_FLAG_NONE, 0);
+
+		if ((status = switch_core_session_read_lock(session)) == SWITCH_STATUS_SUCCESS) {
+			switch_mutex_unlock(conference->mutex);
+			if (!switch_channel_ready(switch_core_session_get_channel(session))) {
+				status = SWITCH_STATUS_FALSE;
+			} else {
+				status = switch_core_session_read_video_frame(session, &vid_frame, SWITCH_IO_FLAG_NONE, 0);
+			}
+			switch_mutex_lock(conference->mutex);
+			switch_core_session_rwunlock(session);
 		}
-		switch_mutex_lock(conference->mutex);
-		switch_core_session_rwunlock(session);
 
 		if (!SWITCH_READ_ACCEPTABLE(status)) {
 			yield = 100000;
@@ -6954,7 +6957,7 @@ static conference_obj_t *conference_new(char *name, conf_xml_cfg_t cfg, switch_c
 	switch_uuid_t uuid;
 	switch_codec_implementation_t read_impl = { 0 };
 	switch_channel_t *channel = NULL;
-	const char *force_rate = NULL, *force_interval = NULL;
+	const char *force_rate = NULL, *force_interval = NULL, *presence_id = NULL;
 	uint32_t force_rate_i = 0, force_interval_i = 0;
 
 	/* Validate the conference name */
@@ -6969,6 +6972,8 @@ static conference_obj_t *conference_new(char *name, conf_xml_cfg_t cfg, switch_c
 		switch_core_session_get_read_impl(session, &read_impl);
 		channel = switch_core_session_get_channel(session);
 		
+		presence_id = switch_channel_get_variable(channel, "presence_id");
+
 		if ((force_rate = switch_channel_get_variable(channel, "conference_force_rate"))) {
 			if (!strcasecmp(force_rate, "auto")) {
 				force_rate_i = read_impl.actual_samples_per_second;
@@ -7362,7 +7367,11 @@ static conference_obj_t *conference_new(char *name, conf_xml_cfg_t cfg, switch_c
 
 	conference->name = switch_core_strdup(conference->pool, name);
 
-	if ((name_domain = strchr(conference->name, '@'))) {
+	if (presence_id && (name_domain = strchr(presence_id, '@'))) {
+		name_domain++;
+		conference->domain = switch_core_strdup(conference->pool, name_domain);
+	} else if ((name_domain = strchr(conference->name, '@'))) {
+		name_domain++;
 		conference->domain = switch_core_strdup(conference->pool, name_domain);
 	} else if (domain) {
 		conference->domain = switch_core_strdup(conference->pool, domain);

src/mod/applications/mod_curl/mod_curl.2010.vcxproj

@@ -136,6 +136,12 @@
       <Project>{9778f1c0-09bc-4698-8ebc-bd982247209a}</Project>
       <ReferenceOutputAssembly>false</ReferenceOutputAssembly>
     </ProjectReference>
+    <ProjectReference Include="..\..\..\..\libs\win32\openssl\libeay32.2010.vcxproj">
+      <Project>{d331904d-a00a-4694-a5a3-fcff64ab5dbe}</Project>
+    </ProjectReference>
+    <ProjectReference Include="..\..\..\..\libs\win32\openssl\ssleay32.2010.vcxproj">
+      <Project>{b4b62169-5ad4-4559-8707-3d933ac5db39}</Project>
+    </ProjectReference>
     <ProjectReference Include="..\..\..\..\w32\Library\FreeSwitchCore.2010.vcxproj">
       <Project>{202d7a4e-760d-4d0e-afa1-d7459ced30ff}</Project>
       <ReferenceOutputAssembly>false</ReferenceOutputAssembly>

src/mod/applications/mod_db/conf/autoload_configs/db.conf.xml

@@ -0,0 +1,5 @@
+<configuration name="db.conf" description="LIMIT DB Configuration">
+  <settings>
+    <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
+  </settings>
+</configuration>

src/mod/applications/mod_distributor/conf/autoload_configs/distributor.conf.xml

@@ -0,0 +1,10 @@
+<configuration name="distributor.conf" description="Distributor Configuration">
+  <lists>
+    <!-- every 10 calls to test you will get foo1 once and foo2 9 times...yes NINE TIMES! -->
+    <!-- this is not the same as 100 with 10 and 90 that would do foo1 10 times in a row then foo2 90 times in a row -->
+    <list name="test" total-weight="10">
+      <node name="foo1" weight="1"/>
+      <node name="foo2" weight="9"/>
+    </list>
+  </lists>
+</configuration>

src/mod/applications/mod_distributor/conf/dialplan/distributor.xml

@@ -0,0 +1,8 @@
+<?xml version="1.0" encoding="utf-8"?>
+<include>  
+  <extension name="test gateway distro">
+      <condition field="destination_number" expression="^(.*)$">
+          <action application="bridge" data="sofia/gateway/${distributor(test)}/$1"/>
+      </condition>
+   </extension>
+</include>

src/mod/applications/mod_dptools/mod_dptools.c

@@ -2902,6 +2902,12 @@ SWITCH_STANDARD_APP(audio_bridge_function)
 										v_campon_fallback_exten,
 										switch_channel_get_variable(caller_channel, "campon_fallback_dialplan"),
 										switch_channel_get_variable(caller_channel, "campon_fallback_context"));
+
+			if (peer_session) {
+				switch_channel_hangup(switch_core_session_get_channel(peer_session), SWITCH_CAUSE_ORIGINATOR_CANCEL);
+				switch_core_session_rwunlock(peer_session);
+			}
+
 			return;
 		}
 
@@ -3038,6 +3044,11 @@ SWITCH_STANDARD_APP(audio_bridge_function)
 	} else {
 
 		if (switch_channel_test_flag(caller_channel, CF_PROXY_MODE)) {
+			switch_channel_t *peer_channel = switch_core_session_get_channel(peer_session);
+			if (switch_true(switch_channel_get_variable(caller_channel, SWITCH_BYPASS_MEDIA_AFTER_BRIDGE_VARIABLE)) ||
+				switch_true(switch_channel_get_variable(peer_channel, SWITCH_BYPASS_MEDIA_AFTER_BRIDGE_VARIABLE))) {
+				switch_channel_set_flag(caller_channel, CF_BYPASS_MEDIA_AFTER_BRIDGE);
+			}
 			switch_ivr_signal_bridge(session, peer_session);
 		} else {
 			switch_channel_t *channel = switch_core_session_get_channel(session);
@@ -3240,7 +3251,7 @@ static switch_call_cause_t user_outgoing_channel(switch_core_session_t *session,
 												 switch_call_cause_t *cancel_cause)
 {
 	switch_xml_t x_user = NULL, x_param, x_params;
-	char *user = NULL, *domain = NULL, *dup_domain = NULL;
+	char *user = NULL, *domain = NULL, *dup_domain = NULL, *dialed_user = NULL;
 	const char *dest = NULL;
 	switch_call_cause_t cause = SWITCH_CAUSE_NONE;
 	unsigned int timelimit = 60;
@@ -3310,9 +3321,14 @@ static switch_call_cause_t user_outgoing_channel(switch_core_session_t *session,
 		}
 	}
 
+	dialed_user = (char *)switch_xml_attr(x_user, "id");
+
 	if (var_event) {
-		switch_event_add_header_string(var_event, SWITCH_STACK_BOTTOM, "dialed_user", user);
+		switch_event_add_header_string(var_event, SWITCH_STACK_BOTTOM, "dialed_user", dialed_user);
 		switch_event_add_header_string(var_event, SWITCH_STACK_BOTTOM, "dialed_domain", domain);
+		if (!zstr(dest) && !strstr(dest, "presence_id=")) {
+			switch_event_add_header(var_event, SWITCH_STACK_BOTTOM, "presence_id", "%s@%s", dialed_user, domain);
+		}
 	}
 
 	if (!dest) {
@@ -3338,7 +3354,7 @@ static switch_call_cause_t user_outgoing_channel(switch_core_session_t *session,
 				timelimit = atoi(varval);
 			}
 
-			switch_channel_set_variable(channel, "dialed_user", user);
+			switch_channel_set_variable(channel, "dialed_user", dialed_user);
 			switch_channel_set_variable(channel, "dialed_domain", domain);
 
 			d_dest = switch_channel_expand_variables(channel, dest);
@@ -3359,7 +3375,7 @@ static switch_call_cause_t user_outgoing_channel(switch_core_session_t *session,
 				switch_assert(event);
 			}
 
-			switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "dialed_user", user);
+			switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "dialed_user", dialed_user);
 			switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "dialed_domain", domain);
 			d_dest = switch_event_expand_headers(event, dest);
 			switch_event_destroy(&event);
@@ -3374,7 +3390,7 @@ static switch_call_cause_t user_outgoing_channel(switch_core_session_t *session,
 		}
 
 
-		switch_snprintf(stupid, sizeof(stupid), "user/%s", user);
+		switch_snprintf(stupid, sizeof(stupid), "user/%s", dialed_user);
 		if (switch_stristr(stupid, d_dest)) {
 			switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_WARNING, "Waddya Daft? You almost called '%s' in an infinate loop!\n",
 							  stupid);

src/mod/applications/mod_easyroute/conf/autoload_configs/easyroute.conf.xml

@@ -0,0 +1,28 @@
+<configuration name="easyroute.conf" description="EasyRoute Module">
+  <settings>
+    <!-- These are kind Obvious -->
+    <param name="db-username" value="root"/>
+    <param name="db-password" value="password"/>
+    <param name="db-dsn" value="easyroute"/>
+
+    <!-- Default Technology and profile -->
+    <param name="default-techprofile" value="sofia/default"/>
+    
+    <!-- IP or Hostname of Default Route -->
+    <param name="default-gateway" value="192.168.66.6"/>
+
+    <!-- Number of times to retry ODBC connection on connection problems, default is 120 -->
+    <param name="odbc-retries" value="120"/>
+
+    <!-- Customer Query. Use this with Care!!! We are not responsible if you mess
+         This up!!! Query *MUST* return columns in the following order!
+	 gateway varchar(128) - contains destination gateway host:port pair (ex: 192.168.1.1:5060 )
+	 group varchar(128) - contains optional group name
+	 call_limit varchar(16) - contains optional call limit
+	 tech_prefix varchar(128) - tech prefix used to build dial string (ex: sofia/default )
+	 acctcode varchar(128) - used to set channel variable acctcode for logging into the CDRs
+	 destination_number varchar(16) - Number returning for the query for building the dial string. (ex: 18005551212) 
+	 See Documentation on the Wiki for further information -->
+    <!-- <param name="custom-query" value="call  FS_GET_SIP_LOCATION(%s);"/> -->
+  </settings>
+</configuration>

src/mod/applications/mod_easyroute/conf/dialplan/easyroute.xml

@@ -0,0 +1,27 @@
+<?xml version="1.0" encoding="utf-8"?>
+<include>
+	<!-- This example will test the limit value for the DID number being checked -->
+	<!-- If over limit, send 503 -->
+	<extension name="easyroute">
+		<condition field="destination_number" expression="^1?(\d{10})$" break="on-true">
+			<action application="easyroute" data="$1"/>
+			<action application="limit" data="db easyroute ${easy_group} ${easy_limit} LE-$1"/>
+			<action application="bridge" data="${easy_dialstring}"/>
+		</condition>
+	</extension>
+
+	<extension name="easyroute_limit_exceeded">
+		<condition field="destination_number" expression="^LE-(\d{10})$" break="on-true">
+			<action application="respond" data="503 Buzz Off You Ran Out of Channels"/>
+		</condition>
+	</extension>
+
+	<!-- Simple example - just grab the dialstring and send the call -->
+	<extension name="easyroute">
+		<condition field="destination_number" expression="^1?(\d{10})$" break="on-true">
+			<action application="set" data="dialstring={easyroute($1 dialstring)"/>
+			<action application="bridge" data="${easy_dialstring}"/>
+		</condition>
+	</extension>
+
+<include>

src/mod/applications/mod_easyroute/mod_easyroute.c

@@ -43,7 +43,7 @@ typedef struct easyroute_results {
 	char dialstring[256];
 	char group[16];
 	char acctcode[17];
-	char translated[17];
+	char translated[61];
 } easyroute_results_t;
 
 
@@ -53,7 +53,7 @@ typedef struct route_callback {
 	char techprofile[129];
 	char limit[129];
 	char acctcode[129];
-	char translated[17];
+	char translated[61];
 } route_callback_t;
 
 static struct {
@@ -88,7 +88,7 @@ static int route_callback(void *pArg, int argc, char **argv, char **columnNames)
 	switch_copy_string(cbt->limit, argv[2], 128);
 	switch_copy_string(cbt->techprofile, argv[3], 128);
 	switch_copy_string(cbt->acctcode, argv[4], 128);
-	switch_copy_string(cbt->translated, argv[5], 16);
+	switch_copy_string(cbt->translated, argv[5], 60);
 
 	return 0;
 }

src/mod/applications/mod_enum/conf/autoload_configs/enum.conf.xml

@@ -0,0 +1,13 @@
+<configuration name="enum.conf" description="ENUM Module">
+  <settings>
+    <param name="default-root" value="e164.org"/>
+    <param name="default-isn-root" value="freenum.org"/>
+    <param name="query-timeout" value="10"/>
+    <param name="auto-reload" value="true"/>
+  </settings>
+
+  <routes>
+    <route service="E2U+SIP" regex="sip:(.*)" replace="sofia/${use_profile}/$1"/>
+    <!--<route service="E2U+XMPP" regex="XMPP:(.*)" replace="dingaling/$${xmpp_server_profile}/$1"/>-->
+  </routes>
+</configuration>

src/mod/applications/mod_enum/conf/dialplan/99999_enum.xml

@@ -0,0 +1,8 @@
+<include>
+	<extension name="enum">
+		<condition field="${module_exists(mod_enum)}" expression="true"/>
+		<condition field="destination_number" expression="^(.*)$">
+			<action application="transfer" data="$1 enum"/>
+		</condition>
+	</extension>
+</include>

src/mod/applications/mod_fifo/conf/autoload_configs/fifo.conf.xml

@@ -0,0 +1,10 @@
+<configuration name="fifo.conf" description="FIFO Configuration">
+  <settings>
+    <param name="delete-all-outbound-member-on-startup" value="false"/>
+  </settings>
+  <fifos>
+    <fifo name="cool_fifo@$${domain}" importance="0">
+      <!--<member timeout="60" simo="1" lag="20">{member_wait=nowait}user/1005@$${domain}</member>-->
+    </fifo>
+  </fifos>
+</configuration>

src/mod/applications/mod_fifo/conf/dialplan/fifo.xml

@@ -0,0 +1,19 @@
+	<!-- 
+		Parking extensions... transferring calls to 5900 will park them in a queue.
+	-->
+	<extension name="park">
+		<condition field="destination_number" expression="^5900$">
+			<action application="set" data="fifo_music=$${hold_music}"/>
+			<action application="fifo" data="5900@${domain_name} in"/>
+		</condition>
+	</extension>
+
+	<!-- 
+		Parking pickup extension.  Calling 5901 will pickup the call.
+	-->
+	<extension name="unpark">
+		<condition field="destination_number" expression="^5901$">
+			<action application="answer"/>
+			<action application="fifo" data="5900@${domain_name} out nowait"/>
+		</condition>
+	</extension>

src/mod/applications/mod_hash/conf/autoload_configs/hash.conf.xml

@@ -0,0 +1,6 @@
+<configuration name="hash.conf" description="Hash Configuration">
+  <remotes>
+	<!-- List of hosts from where to pull usage data -->
+	<!-- <remote name="Test1" host="10.0.0.10" port="8021" password="ClueCon" interval="1000" /> -->
+  </remotes>
+</configuration>

src/mod/applications/mod_hash/conf/dialplan/hash.xml

@@ -0,0 +1,7 @@
+	<extension name="hash_demo">
+		<condition field="destination_number" expression="^hash_demo$">
+			<action application="hash" data="insert/realm/key/value"/>
+			<action application="hash" data="delete/realm/key"/>
+			<action application="set" data="var=${hash(select/realm/key)}"/>
+		</condition>
+	</extension>

src/mod/applications/mod_httapi/docs/mod_httapi_doc.txt

@@ -80,6 +80,21 @@ asr-engine          : ASR engine to use
 asr-grammar         : ASR grammar to use
 
 
+<vmname file name action digit-timeout input-timeout loops><bind action strip>*EXPR*</bind></playback>
+		    : Plays a vm name and optionally collects input.
+
+ATTRS:
+file                : The file
+name                : Param name to save result.
+error-file          : Error file to play on invalid input.
+action              : Change the new target url.
+temp-action	    : Change url to submit to. just for the next loop.
+digit-timeout       : Timeout waiting for digits after file plays (when input bindings are present)
+input-timeout       : Timeout waiting for more digits in a multi-digit input.
+loops               : max times to play the file when input bindings are present.
+
+
+
 
 
 <record file name error-file action digit-timeout input-timeout><bind action strip>*EXPR*</bind></record>

src/mod/applications/mod_httapi/mod_httapi.c

@@ -395,6 +395,7 @@ static switch_status_t parse_playback(const char *tag_name, client_t *client, sw
 	const char *say_gender = NULL;
 	const char *sp_engine = NULL;
 	const char *sp_grammar = NULL;
+	char *free_string = NULL;
 
 	if (!strcasecmp(tag_name, "say")) {
 		say_lang = switch_xml_attr(tag, "language");
@@ -431,6 +432,48 @@ static switch_status_t parse_playback(const char *tag_name, client_t *client, sw
 		} else {
 			play = 1;
 		}
+	} else if (!strcasecmp(tag_name, "vmname")) {
+		const char *id = switch_xml_attr(tag, "id");
+		char *cmd, *resp;
+		switch_stream_handle_t stream = { 0 };
+		char *p;
+
+		SWITCH_STANDARD_STREAM(stream);
+		
+		cmd = switch_mprintf("%s|name_path", id);
+
+		switch_api_execute("vm_prefs", cmd, NULL, &stream);
+
+		resp = (char *) stream.data;
+
+		if (!zstr(resp)) {
+			if (switch_stristr("://", resp) || switch_file_exists(resp, NULL) == SWITCH_STATUS_SUCCESS) {
+				play = 1;
+				file = free_string = resp;
+				resp = NULL;
+			} else {
+				switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Invalid file! [%s]\n", resp);
+				switch_safe_free(resp);
+				return SWITCH_STATUS_FALSE;
+			}
+		} else {
+			switch_safe_free(resp);
+			say_lang = switch_xml_attr(tag, "language");
+			say_gender = switch_xml_attr(tag, "gender");
+			say_type = "name_spelled";
+			say_method = "pronounced";
+			free_string = strdup(id);
+			if ((p = strchr(free_string, '@'))) {
+				*p = '\0';
+			}
+			say = 1;
+
+			body = free_string;
+			switch_ivr_play_file(client->session, NULL, "voicemail/vm-person.wav", &nullargs);
+			
+		}
+
+		switch_safe_free(resp);
 	}
 
 
@@ -545,7 +588,7 @@ static switch_status_t parse_playback(const char *tag_name, client_t *client, sw
 
 		if (input_timeout && status == SWITCH_STATUS_SUCCESS) {
 			if ((status = switch_ivr_sleep(client->session, input_timeout, SWITCH_TRUE, args)) == SWITCH_STATUS_SUCCESS) {
-				status = SWITCH_STATUS_BREAK;
+				status = (input || !pause) ? SWITCH_STATUS_BREAK : SWITCH_STATUS_SUCCESS;
 			}
 		}
 
@@ -618,11 +661,13 @@ static switch_status_t parse_playback(const char *tag_name, client_t *client, sw
 			}
 		}
 	}
-
+	
 	if (dmachine) {
 		switch_ivr_dmachine_destroy(&dmachine);
 	}
 
+	switch_safe_free(free_string);
+	
 	return status;
 }
 
@@ -2790,6 +2835,7 @@ SWITCH_MODULE_LOAD_FUNCTION(mod_httapi_load)
 	bind_parser("record", parse_record);
 	bind_parser("recordCall", parse_record_call);
 	bind_parser("playback", parse_playback);
+	bind_parser("vmName", parse_playback);
 	bind_parser("speak", parse_playback);
 	bind_parser("say", parse_playback);
 	bind_parser("conference", parse_conference);

src/mod/applications/mod_http_cache/conf/autoload_configs/http_cache.conf.xml

@@ -0,0 +1,8 @@
+<configuration name="http_cache.conf" description="HTTP GET cache">
+  <settings>
+    <param name="max-urls" value="10000"/>
+    <param name="location" value="$${base_dir}/http_cache"/>
+    <param name="default-max-age" value="86400"/>
+  </settings>
+</configuration>
+

src/mod/applications/mod_http_cache/conf/dialplan/http_cache.xml

@@ -0,0 +1,7 @@
+<include>
+	<extension "http_cache_demo">
+		<condition field="destination_number" expression="^(.*)$">
+			<action application="playback" data="${http_get(http://myserver.yo/media/hello_world.wav)}"/>
+		</condition>
+	</extension>
+</include>

src/mod/applications/mod_ladspa/conf/dialplan/00_ladspa.xml

@@ -0,0 +1,77 @@
+<include>
+  
+  <X-PRE-PROCESS cmd="set" data="AT_EPENT1=0 0 0 -1 -1 0 -1 0 -1 -1 0 -1"/>
+  <X-PRE-PROCESS cmd="set" data="AT_EPENT2=1 1 1 -1 -1 1 -1 1 -1 -1 1 -1"/>
+  <X-PRE-PROCESS cmd="set" data="AT_CPENT1=0 -1 -1 0 -1 0 0 0 -1 -1 0 -1"/>
+  <X-PRE-PROCESS cmd="set" data="AT_CPENT2=1 -1 -1 1 -1 1 1 1 -1 -1 1 -1"/>
+  <X-PRE-PROCESS cmd="set" data="AT_CMAJ1=0 -1 0 0 -1 0 -1 0 0 -1 0 -1"/>
+  <X-PRE-PROCESS cmd="set" data="AT_CMAJ2=1 -1 1 1 -1 1 -1 1 1 -1 1 -1"/>
+  <X-PRE-PROCESS cmd="set" data="AT_BBLUES=1 -1 1 -1 -1 1 -1 1 1 1 -1 -1"/>
+  <X-PRE-PROCESS cmd="set" data="ATGPENT2=-1 1 -1 1 -1 1 -1 -1 1 -1 1 -1"/>
+  
+  <extension name="101"> 
+    <condition field="destination_number" expression="^101$"> 
+      <!-- AUTOTALENT DEFAULTS -->
+
+      <!--
+      <action application="set" data="AT_TUNE=440"/>
+      <action application="set" data="AT_FIXED=0"/>
+      <action application="set" data="AT_PULL=0"/>
+      <action application="set" data="AT_A=0"/>
+      <action application="set" data="AT_Bb=-1"/>
+      <action application="set" data="AT_B=0"/>
+      <action application="set" data="AT_C=0"/>
+      <action application="set" data="AT_Db=-1"/>
+      <action application="set" data="AT_D=0"/>
+      <action application="set" data="AT_Eb=-1"/>
+      <action application="set" data="AT_E=0"/>
+      <action application="set" data="AT_F=0"/>
+      <action application="set" data="AT_Gb=-1"/>
+      <action application="set" data="AT_G=0"/>
+      <action application="set" data="AT_Ab=-1"/>
+      <action application="set" data="AT_AMOUNT=1"/>
+      <action application="set" data="AT_SMOOTH=0"/>
+      <action application="set" data="AT_SHIFT=0"/>
+      <action application="set" data="AT_OUTSCALE=0"/>
+      <action application="set" data="AT_LFODEPTH=0"/>
+      <action application="set" data="AT_LFORATE=5"/>
+      <action application="set" data="AT_LFOSHAPE=0"/>
+      <action application="set" data="AT_LFOSYMM=0"/>
+      <action application="set" data="AT_LFOQUANT=0"/>
+      <action application="set" data="AT_FCORR=0"/>
+      <action application="set" data="AT_FWARP=0"/>
+      <action application="set" data="AT_MIX=1"/>
+      -->
+
+
+      <action application="set" data="AT_TUNE=440"/>
+      <action application="set" data="AT_FIXED=0"/>
+      <action application="set" data="AT_PULL=0"/>
+
+      <action application="set" data="AT_AMOUNT=1"/>
+      <action application="set" data="AT_SMOOTH=0"/>
+      <action application="set" data="AT_SHIFT=1"/>
+      <action application="set" data="AT_OUTSCALE=0"/>
+      <action application="set" data="AT_LFODEPTH=0"/>
+      <action application="set" data="AT_LFORATE=5"/>
+      <action application="set" data="AT_LFOSHAPE=0"/>
+      <action application="set" data="AT_LFOSYMM=0"/>
+      <action application="set" data="AT_LFOQUANT=0"/>
+      <action application="set" data="AT_FCORR=0"/>
+      <action application="set" data="AT_FWARP=0"/>
+      <action application="set" data="AT_MIX=1"/>
+
+      <!-- you have to download the ladspa package and the desired plugins from their desired site -->
+
+
+      <action application="set"><![CDATA[ladspa_params=${AT_TUNE} ${AT_FIXED} ${AT_PULL} ${AT_EPENT2} ${AT_AMOUNT} ${AT_SMOOTH} ${AT_SHIFT} ${AT_OUTSCALE} ${AT_LFODEPTH} ${AT_LFORATE} ${AT_LFOSHAPE} ${AT_LFOSYMM} ${AT_LFOQUANT} ${AT_FCORR} ${AT_FWARP} ${AT_MIX}]]></action>
+      
+      <action application="ladspa_run" data="r|autotalent||${ladspa_params}"/>
+      <action application="ladspa_run" data="r|tap_chorusflanger||"/>
+      <action application="ladspa_run" data="r|phasers_1217.so|autoPhaser|"/>
+      <action application="bridge" data="sofia/internal/888@conference.freeswitch.org"/>
+
+      </condition> 
+  </extension> 
+
+</include>

src/mod/applications/mod_ladspa/docs/README

@@ -0,0 +1,4 @@
+mod_ladspa is a module that allow to use Linux Audio Developer's Simple Plugin API inside freeswitch in realtime. 
+see http://wiki.freeswitch.org/wiki/Mod_ladspa for installation and configuration documentation.
+
+Please Note Currently this module only works on Linux.

src/mod/applications/mod_lcr/conf/autoload_configs/lcr.conf.xml

@@ -0,0 +1,89 @@
+<configuration name="lcr.conf" description="LCR Configuration">
+  <settings>
+    <param name="odbc-dsn" value="freeswitch-mysql:freeswitch:Fr33Sw1tch"/>
+<!--    <param name="odbc-dsn" value="freeswitch-pgsql:freeswitch:Fr33Sw1tch"/> -->
+  </settings>
+  <profiles>
+    <profile name="default">
+      <param name="id" value="0"/>
+      <param name="order_by" value="rate,quality,reliability"/>
+    </profile>
+    <profile name="qual_rel">
+      <param name="id" value="1"/>
+      <param name="order_by" value="quality,reliability"/>
+    </profile>
+    <profile name="rel_qual">
+      <param name="id" value="2"/>
+      <param name="order_by" value="reliability,quality"/>
+    </profile>
+<!-- 
+  Some samples of how to do custom SQL:
+
+    =============================================================
+    PostgreSQL with contrib prefix module which supports fast
+    prefix queries.  Ideal option.
+    =============================================================
+    <profile name="pg_prefix">
+      <param name="custom_sql" value="
+SELECT l.digits AS lcr_digits,
+       c.carrier_name AS lcr_carrier_name,
+       l.${lcr_rate_field} as lcr_rate_field,
+       cg.prefix AS lcr_gw_prefix, cg.suffix AS lcr_gw_suffix,
+       l.lead_strip AS lcr_lead_strip, l.trail_strip AS lcr_trail_strip,
+       l.prefix AS lcr_prefix, l.suffix AS lcr_suffix
+  FROM lcr l
+    JOIN carriers c ON l.carrier_id=c.id
+    JOIN carrier_gateway cg ON c.id=cg.carrier_id
+  WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
+    AND digits_prefix @> %q
+    AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end
+  ORDER BY digits DESC, ${lcr_rate_field}, random();
+      "/>
+    </profile>
+
+    =============================================================
+    PostgreSQL with contrib prefix module which supports fast
+    prefix queries.  Ideal option.  Alternate syntax which requies
+    a session but allows variable substitution.
+    =============================================================
+    <profile name="pg_prefix2">
+      <param name="custom_sql" value="
+SELECT l.digits AS lcr_digits,
+       c.carrier_name AS lcr_carrier_name,
+       l.${lcr_rate_field} as lcr_rate_field,
+       cg.prefix AS lcr_gw_prefix, cg.suffix AS lcr_gw_suffix,
+       l.lead_strip AS lcr_lead_strip, l.trail_strip AS lcr_trail_strip,
+       l.prefix AS lcr_prefix, l.suffix AS lcr_suffix
+  FROM lcr l
+    JOIN carriers c ON l.carrier_id=c.id
+    JOIN carrier_gateway cg ON c.id=cg.carrier_id
+  WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
+    AND digits_prefix @> '${lcr_query_digits}'
+    AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end
+  ORDER BY digits DESC, ${lcr_rate_field}, random();
+      "/>
+    </profile>
+
+    =============================================================
+    Demonstrates use of computed inlist.
+    =============================================================
+    <profile name="inlist">
+      <param name="custom_sql" value="
+SELECT l.digits AS lcr_digits,
+       c.carrier_name AS lcr_carrier_name,
+       l.${lcr_rate_field} as lcr_rate_field,
+       cg.prefix AS lcr_gw_prefix, cg.suffix AS lcr_gw_suffix,
+       l.lead_strip AS lcr_lead_strip, l.trail_strip AS lcr_trail_strip,
+       l.prefix AS lcr_prefix, l.suffix AS lcr_suffix
+  FROM lcr l
+    JOIN carriers c ON l.carrier_id=c.id
+    JOIN carrier_gateway cg ON c.id=cg.carrier_id
+  WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
+    AND digits IN (${lcr_query_expanded_digits})
+    AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end
+  ORDER BY digits DESC, ${lcr_rate_field}, random();
+      "/>
+    </profile>
+-->
+  </profiles>
+</configuration>

src/mod/applications/mod_lcr/conf/dialplan/lcr.xml

@@ -0,0 +1,8 @@
+<include>	
+	<extension name="Outbound to PSTN 11 Digits">
+		<condition field="destination_number" expression="^(1[2-9][0-9]{2}[2-9]{7})$">
+			<action application="lcr" data="$1"/>
+			<action application="bridge" data="${lcr_auto_route}"/>
+		</condition>
+	</extension>
+</include>

src/mod/applications/mod_limit/conf/autoload_configs/limit.conf.xml

@@ -0,0 +1,5 @@
+<configuration name="limit.conf" description="LIMIT Configuration">
+  <settings>
+    <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
+  </settings>
+</configuration>

src/mod/applications/mod_limit/conf/dialplan/limit.xml

@@ -0,0 +1,14 @@
+<include>
+	<extension name="limit_exceeded">
+		<condition field="destination_number" expression="^limit_exceeded$">
+			<action application="playback" data="/sounds/overthelimit.wav"/>
+			<action application="hangup"/>
+		</condition>
+	</extension>
+
+	<extension name="limit" continue="true">
+		<condition>
+			<action application="limit" data="db $${domain} ${sip_auth_username} ${max_calls}"/>
+		</condition>
+	</extension> 
+</include>

src/mod/applications/mod_memcache/conf/autoload_configs/memcache.conf.xml

@@ -0,0 +1,6 @@
+<configuration name="memcache.conf" description="memcache Configuration">
+  <settings>
+    <!-- comma sep list of servers: eg:  localhost,otherhost:port,anotherone -->
+    <param name="memcache-servers" value="localhost"/>
+  </settings>
+</configuration>

src/mod/applications/mod_memcache/conf/dialplan/memcache.xml

@@ -0,0 +1,9 @@
+<include>
+	<extension name="memcache_examples">
+		<condition field="destination_number" expression="^memcache$">
+			<action application="set" data="foo=${memcache(get key)}"/>
+			<action application="set" data="ignore=${memcache(set key value)}"/>
+			<action application="set" data="ignore=${memcache(set key ${channel_var})}"/>
+		</condition>
+	</extension>
+</include>

src/mod/applications/mod_mongo/conf/autoload_configs/mongo.conf.xml

@@ -0,0 +1,20 @@
+<configuration name="mongo.conf">
+  <settings>
+    <!-- 
+      connection-string handles different ways to connect to mongo
+      samples:
+         server:port
+         foo/server:port,server:port   SET
+    -->
+    <param name="connection-string" value="127.0.0.1:27017"/>
+    <param name="min-connections" value="10"/>
+    <param name="max-connections" value="100"/>
+
+<!--
+    <param name="map" value="function() { emit(this.a, 1); }"/>
+    <param name="reduce" value="function(key, values) { return Array.sum(values); }"/>
+    <param name="finalize" value="function(key, value) { return value;}"/>
+-->
+
+  </settings>
+</configuration>

src/mod/applications/mod_nibblebill/conf/autoload_configs/nibblebill.conf.xml

@@ -0,0 +1,50 @@
+<configuration name="nibblebill.conf" description="Nibble Billing">
+  <settings>
+    <!-- See http://wiki.freeswitch.org/wiki/Mod_nibblebill for help with these options -->
+
+    <!-- Information for connecting to your database -->
+    <param name="db_username" value="bandwidth.com"/>
+    <param name="db_password" value="password"/>
+    <param name="db_dsn" value="bandwidth.com"/>
+
+    <!-- The database table where your CASH column is located -->
+    <param name="db_table" value="accounts"/>
+
+    <!-- The column name where we store the value of the account -->
+    <param name="db_column_cash" value="cash"/>
+
+    <!-- The column name for the unique ID identifying the account -->
+    <param name="db_column_account" value="id"/>
+
+    <!-- Custom SQL for loading current balance - overrides column names
+         channel vars are interpreted.
+         field nibble_balance is used for balance info
+
+    <param name="custom_sql_lookup" value="SELECT cash AS nibble_balance FROM accounts WHERE account_code='${nibble_account}'"/>
+    -->
+
+    <!-- Custom SQL for loading current balance - overrides column names
+         channel vars are interpreted.
+         nibble_increment is the amount to update
+
+    <param name="custom_sql_save" value="UPDATE accounts SET cash=cash-${nibble_increment} WHERE account_code='${nibble_account}'"/>
+    -->
+
+
+    <!-- Default heartbeat interval. Set to 'off' for no heartbeat (i.e. bill only at end of call) -->
+    <param name="global_heartbeat" value="60"/>
+
+    <!-- By default, warn a caller when their balance is at $5.00. You can set this to a negative number. -->
+    <param name="lowbal_amt" value="5"/>
+    <param name="lowbal_action" value="play ding"/>
+
+    <!-- By default, terminate a caller when their balance hits $0.00. You can set this to a negative number. -->
+    <param name="nobal_amt" value="0"/>
+    <param name="nobal_action" value="hangup"/>
+
+    <!-- If a call goes beyond a certain dollar amount, flag or terminate it -->
+    <param name="percall_max_amt" value="100"/>
+    <param name="percall_action" value="hangup"/>
+
+  </settings>
+</configuration>

src/mod/applications/mod_nibblebill/conf/dialplan/nibblebill.xml

@@ -0,0 +1,27 @@
+<include>
+
+	<extension name="tollfree800">
+		<condition field="destination_number" expression="^1{0,1}(800\d{7})$">
+			<action application="set" data="nibble_account=${accountcode}"/>
+			<action application="set" data="nibble_rate=0"/>
+			<action application="bridge" data="sofia/gateway/bandwidth.com/$1"/>
+		</condition>
+	</extension>
+
+	<extension name="special919rate">
+		<condition field="destination_number" expression="^1{0,1}(919\d{7})$">
+			<action application="set" data="nibble_account=${accountcode}"/>
+			<action application="set" data="nibble_rate=0.07"/>
+			<action application="bridge" data="sofia/gateway/bandwidth.com/$1"/>
+		</condition>
+	</extension>
+
+	<extension name="domestic">
+		<condition field="destination_number" expression="^(1{0,1}\d{10})$">
+			<action application="set" data="nibble_account=${accountcode}"/>
+			<action application="set" data="nibble_rate=0.05"/>
+			<action application="bridge" data="sofia/gateway/bandwidth.com/$1"/>
+		</condition>
+	</extension>
+
+</include>

src/mod/applications/mod_nibblebill/mod_nibblebill.c

@@ -52,19 +52,20 @@
 
 typedef struct {
 	switch_time_t lastts;		/* Last time we did any billing */
-	float total;				/* Total amount billed so far */
+	double total;				/* Total amount billed so far */
 
 	switch_time_t pausets;		/* Timestamp of when a pause action started. 0 if not paused */
-	float bill_adjustments;		/* Adjustments to make to the next billing, based on pause/resume events */
+	double bill_adjustments;	/* Adjustments to make to the next billing, based on pause/resume events */
 
+	int lowbal_action_executed;	/* Set to 1 once lowbal_action has been executed */
 } nibble_data_t;
 
 
 typedef struct nibblebill_results {
-	float balance;
+	double balance;
 
-	float percall_max;			/* Overrides global on a per-user level */
-	float lowbal_amt;			/*  ditto */
+	double percall_max;			/* Overrides global on a per-user level */
+	double lowbal_amt;			/*  ditto */
 } nibblebill_results_t;
 
 
@@ -80,11 +81,11 @@ static struct {
 	switch_mutex_t *mutex;
 
 	/* Global billing config options */
-	float percall_max_amt;		/* Per-call billing limit (safety check, for fraud) */
+	double percall_max_amt;		/* Per-call billing limit (safety check, for fraud) */
 	char *percall_action;		/* Exceeded length of per-call action */
-	float lowbal_amt;			/* When we warn them they are near depletion */
+	double lowbal_amt;			/* When we warn them they are near depletion */
 	char *lowbal_action;		/* Low balance action */
-	float nobal_amt;			/* Minimum amount that must remain in the account */
+	double nobal_amt;			/* Minimum amount that must remain in the account */
 	char *nobal_action;			/* Drop action */
 
 	/* Other options */
@@ -136,7 +137,7 @@ static int nibblebill_callback(void *pArg, int argc, char **argv, char **columnN
 
 	for (i = 0; i < argc; i++) {
 		if (!strcasecmp(columnNames[i], "nibble_balance")) {
-			cbt->balance = (float) atof(argv[0]);
+			cbt->balance = atof(argv[0]);
 		}
 	}
 	
@@ -179,15 +180,15 @@ static switch_status_t load_config(void)
 			} else if (!strcasecmp(var, "percall_action")) {
 				set_global_percall_action(val);
 			} else if (!strcasecmp(var, "percall_max_amt")) {
-				globals.percall_max_amt = (float) atof(val);
+				globals.percall_max_amt = atof(val);
 			} else if (!strcasecmp(var, "lowbal_action")) {
 				set_global_lowbal_action(val);
 			} else if (!strcasecmp(var, "lowbal_amt")) {
-				globals.lowbal_amt = (float) atof(val);
+				globals.lowbal_amt = atof(val);
 			} else if (!strcasecmp(var, "nobal_action")) {
 				set_global_nobal_action(val);
 			} else if (!strcasecmp(var, "nobal_amt")) {
-				globals.nobal_amt = (float) atof(val);
+				globals.nobal_amt = atof(val);
 			} else if (!strcasecmp(var, "global_heartbeat")) {
 				globals.global_heartbeat = atoi(val);
 			}
@@ -252,6 +253,23 @@ void debug_event_handler(switch_event_t *event)
 	}
 }
 
+static switch_status_t exec_app(switch_core_session_t *session, const char *app_string)
+{
+	switch_status_t status;
+	char *strings[2] = { 0 };
+	char *dup;
+
+	if (!app_string) {
+		return SWITCH_STATUS_FALSE;
+	}
+
+	dup = strdup(app_string);
+	switch_assert(dup);
+	switch_separate_string(dup, ' ', strings, sizeof(strings) / sizeof(strings[0]));
+	status = switch_core_session_execute_application(session, strings[0], strings[1]);
+	free(dup);
+	return status;
+}
 
 static void transfer_call(switch_core_session_t *session, char *destination)
 {
@@ -292,9 +310,8 @@ static void transfer_call(switch_core_session_t *session, char *destination)
 	free(mydup);
 }
 
-
 /* At this time, billing never succeeds if you don't have a database. */
-static switch_status_t bill_event(float billamount, const char *billaccount, switch_channel_t *channel)
+static switch_status_t bill_event(double billamount, const char *billaccount, switch_channel_t *channel)
 {
 	char *sql = NULL, *dsql = NULL;
 	switch_odbc_statement_handle_t stmt = NULL;
@@ -306,7 +323,7 @@ static switch_status_t bill_event(float billamount, const char *billaccount, swi
 
 	if (globals.custom_sql_save) {
 		if (switch_string_var_check_const(globals.custom_sql_save) || switch_string_has_escaped_data(globals.custom_sql_save)) {
-			switch_channel_set_variable_printf(channel, "nibble_increment", "%f", billamount, SWITCH_FALSE);
+			switch_channel_set_variable_printf(channel, "nibble_bill", "%f", billamount, SWITCH_FALSE);
 			sql = switch_channel_expand_variables(channel, globals.custom_sql_save);
 			if (sql != globals.custom_sql_save) dsql = sql;
 		} else {
@@ -339,14 +356,14 @@ static switch_status_t bill_event(float billamount, const char *billaccount, swi
 }
 
 
-static float get_balance(const char *billaccount, switch_channel_t *channel)
+static double get_balance(const char *billaccount, switch_channel_t *channel)
 {
 	char *dsql = NULL, *sql = NULL;
 	nibblebill_results_t pdata;
-	float balance = 0.00f;
+	double balance = 0.0;
 
 	if (!switch_odbc_available()) {
-		return -1.00f;
+		return -1.0;
 	}
 
 	memset(&pdata, 0, sizeof(pdata));
@@ -369,7 +386,7 @@ static float get_balance(const char *billaccount, switch_channel_t *channel)
 		switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error running this query: [%s]\n", sql);
 		/* Return -1 for safety */
 
-		balance = -1.00f;
+		balance = -1.0;
 	} else {
 		/* Successfully retrieved! */
 		balance = pdata.balance;
@@ -391,7 +408,7 @@ static switch_status_t do_billing(switch_core_session_t *session)
 	/* Local vars */
 	nibble_data_t *nibble_data;
 	switch_time_t ts = switch_micro_time_now();
-	float billamount;
+	double billamount;
 	char date[80] = "";
 	char *uuid;
 	switch_size_t retsize;
@@ -399,9 +416,9 @@ static switch_status_t do_billing(switch_core_session_t *session)
 	const char *billrate;
 	const char *billincrement;
 	const char *billaccount;
-	float nobal_amt = globals.nobal_amt;
-	//float lowbal_amt = globals.lowbal_amt;
-	float balance;
+	double nobal_amt = globals.nobal_amt;
+	double lowbal_amt = globals.lowbal_amt;
+	double balance;
 
 	if (!session) {
 		/* Why are we here? */
@@ -421,13 +438,13 @@ static switch_status_t do_billing(switch_core_session_t *session)
 	billaccount = switch_channel_get_variable(channel, "nibble_account");
 	
 	if (!zstr(switch_channel_get_variable(channel, "nobal_amt"))) {
-		nobal_amt = (float)atof(switch_channel_get_variable(channel, "nobal_amt"));
+		nobal_amt = atof(switch_channel_get_variable(channel, "nobal_amt"));
 	}
-	/*
+	
 	if (!zstr(switch_channel_get_variable(channel, "lowbal_amt"))) {
-		lowbal_amt = (float)atof(switch_channel_get_variable(channel, "lowbal_amt"));
+		lowbal_amt = atof(switch_channel_get_variable(channel, "lowbal_amt"));
 	}
-	*/
+	
 	/* Return if there's no billing information on this session */
 	if (!billrate || !billaccount) {
 		return SWITCH_STATUS_SUCCESS;
@@ -497,11 +514,15 @@ static switch_status_t do_billing(switch_core_session_t *session)
 	if ((ts - nibble_data->lastts) >= 0) {
 		/* If billincrement is set we bill by it and not by time elapsed */
 		if (!(switch_strlen_zero(billincrement))) {
-			float chargedunits = ((int)((ts - nibble_data->lastts) / 1000000) <= (int)atof(billincrement)) ? ((float)atof(billincrement) * 1000000) :  ceil((ts - nibble_data->lastts) / ((float)atof(billincrement) * 1000000)) * ((float)atof(billincrement) * 1000000);
-			billamount = ((float) atof(billrate) / 1000000 / 60) * chargedunits - nibble_data->bill_adjustments;
+			switch_time_t chargedunits = (ts - nibble_data->lastts) / 1000000 <= atol(billincrement) ? atol(billincrement) * 1000000 : (switch_time_t)(ceil((ts - nibble_data->lastts) / (atol(billincrement) * 1000000.0))) * atol(billincrement) * 1000000;
+			billamount = (atof(billrate) / 1000000 / 60) * chargedunits - nibble_data->bill_adjustments;
+			/* Account for the prepaid amount */
+			nibble_data->lastts += chargedunits;
 		} else {		
 			/* Convert billrate into microseconds and multiply by # of microseconds that have passed since last *successful* bill */
-			billamount = ((float) atof(billrate) / 1000000 / 60) * ((ts - nibble_data->lastts)) - nibble_data->bill_adjustments;
+			billamount = (atof(billrate) / 1000000 / 60) * ((ts - nibble_data->lastts)) - nibble_data->bill_adjustments;
+			/* Update the last time we billed */
+			nibble_data->lastts = ts;
 		}
 
 		switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "Billing $%f to %s (Call: %s / %f so far)\n", billamount, billaccount,
@@ -521,21 +542,31 @@ static switch_status_t do_billing(switch_core_session_t *session)
 			switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_CRIT, "Failed to log to database!\n");
 		}
 	} else {
-		switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_WARNING, "Just tried to bill %s negative minutes! That should be impossible.\n",
-						  uuid);
+		if (switch_strlen_zero(billincrement))
+			switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_WARNING, "Just tried to bill %s negative minutes! That should be impossible.\n", uuid);
 	}
 
-	/* Update the last time we billed */
-	nibble_data->lastts = ts;
-
 	/* Save this location */
 	if (channel) {
 		switch_channel_set_private(channel, "_nibble_data_", nibble_data);
 
 		/* don't verify balance and transfer to nobal if we're done with call */
 		if (switch_channel_get_state(channel) != CS_REPORTING && switch_channel_get_state(channel) != CS_HANGUP) {
-			/* See if this person has enough money left to continue the call */
+			
 			balance = get_balance(billaccount, channel);
+			
+			/* See if we've achieved low balance */
+			if (!nibble_data->lowbal_action_executed && balance <= lowbal_amt) {
+				switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "Balance of %f fell below low balance amount of %f! (Account %s)\n",
+								  balance, lowbal_amt, billaccount);
+
+				if (exec_app(session, globals.lowbal_action) != SWITCH_STATUS_SUCCESS)
+					switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "Low balance action didn't execute\n");
+				else
+					nibble_data->lowbal_action_executed = 1;
+			}
+
+			/* See if this person has enough money left to continue the call */
 			if (balance <= nobal_amt) {
 				/* Not enough money - reroute call to nobal location */
 				switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_CRIT, "Balance of %f fell below allowed amount of %f! (Account %s)\n",
@@ -665,7 +696,7 @@ static void nibblebill_resume(switch_core_session_t *session)
 	billrate = switch_channel_get_variable(channel, "nibble_rate");
 
 	/* Calculate how much was "lost" to billings during pause - we do this here because you never know when the billrate may change during a call */
-	nibble_data->bill_adjustments += ((float) atof(billrate) / 1000000 / 60) * ((ts - nibble_data->pausets));
+	nibble_data->bill_adjustments += (atof(billrate) / 1000000 / 60) * ((ts - nibble_data->pausets));
 	switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_INFO, "Resumed billing! Subtracted %f from this billing cycle.\n",
 					  (atof(billrate) / 1000000 / 60) * ((ts - nibble_data->pausets)));
 
@@ -711,11 +742,11 @@ static void nibblebill_reset(switch_core_session_t *session)
 	}
 }
 
-static float nibblebill_check(switch_core_session_t *session)
+static double nibblebill_check(switch_core_session_t *session)
 {
 	switch_channel_t *channel = switch_core_session_get_channel(session);
 	nibble_data_t *nibble_data;
-	float amount = 0;
+	double amount = 0;
 
 	if (!channel) {
 		return -99999;
@@ -744,7 +775,7 @@ static float nibblebill_check(switch_core_session_t *session)
 	return amount;
 }
 
-static void nibblebill_adjust(switch_core_session_t *session, float amount)
+static void nibblebill_adjust(switch_core_session_t *session, double amount)
 {
 	switch_channel_t *channel = switch_core_session_get_channel(session);
 	const char *billaccount;
@@ -780,7 +811,7 @@ SWITCH_STANDARD_APP(nibblebill_app_function)
 	if (!zstr(data) && (lbuf = strdup(data))
 		&& (argc = switch_separate_string(lbuf, ' ', argv, (sizeof(argv) / sizeof(argv[0]))))) {
 		if (!strcasecmp(argv[0], "adjust") && argc == 2) {
-			nibblebill_adjust(session, (float) atof(argv[1]));
+			nibblebill_adjust(session, atof(argv[1]));
 		} else if (!strcasecmp(argv[0], "flush")) {
 			do_billing(session);
 		} else if (!strcasecmp(argv[0], "pause")) {
@@ -812,7 +843,7 @@ SWITCH_STANDARD_API(nibblebill_api_function)
 			char *uuid = argv[0];
 			if ((psession = switch_core_session_locate(uuid))) {
 				if (!strcasecmp(argv[1], "adjust") && argc == 3) {
-					nibblebill_adjust(psession, (float) atof(argv[2]));
+					nibblebill_adjust(psession, atof(argv[2]));
 				} else if (!strcasecmp(argv[1], "flush")) {
 					do_billing(psession);
 				} else if (!strcasecmp(argv[1], "pause")) {

src/mod/applications/mod_osp/conf/osp.conf.xml

@@ -0,0 +1,53 @@
+<configuration name="osp.conf" description="OSP Module Configuration">
+	<settings>
+		<!-- Debug info flag -->
+		<param name="debug-info" value="disabled"/>
+		<!-- Log level for debug info -->
+		<param name="log-level" value="info"/>
+		<!-- Crypto hareware accelerate is disabled by default -->
+		<param name="crypto-hardware" value="disabled"/>
+		<!-- SIP settings -->
+		<param name="sip" module="sofia" profile="external"/>
+		<!-- H.323 settings -->
+		<!-- <param name="h323" module="h323" profile="external"/> -->
+		<!-- IAX settings -->
+		<!-- <param name="iax" module="iax" profile="external"/> -->
+		<!-- Skype settings -->
+		<!-- <param name="skype" module="skypopen" profile="external"/> -->
+		<!-- Default destination protocol -->
+		<param name="default-protocol" value="sip"/>
+	</settings>
+
+	<profiles>
+	<!-- Default OSP profile -->
+		<profile name="default">
+			<!-- Service point URLs, up to 8 allowed -->
+			<!-- <param name="service-point-url" value="http://osptestserver.transnexus.com:5045/osp"/> -->
+			<!-- <param name="service-point-url" value="https://127.0.0.1:1443/osp"/> -->
+			<param name="service-point-url" value="http://127.0.0.1:5045/osp"/>
+
+			<!-- FreeSWITCH IP address for OSP -->
+			<param name="device-ip" value="127.0.0.1:5080"/>
+
+			<!-- SSL lifetime in seconds -->
+			<param name="ssl-lifetime" value="300"/>
+			<!-- HTTP max connections, 1~1000 -->
+			<param name="http-max-connections" value="20"/>
+			<!-- HTTP persistence in seconds -->
+			<param name="http-persistence" value="60"/>
+			<!-- HTTP retry delay in seconds, 0~10 -->
+			<param name="http-retry-delay" value="0"/>
+			<!-- HTTP retry limit, 0~100 -->
+			<param name="http-retry-limit" value="2"/>
+			<!-- HTTP timeout in milliseconds, 200~60000 -->
+			<param name="http-timeout" value="10000"/>
+
+			<!-- OSP work mode, direct or indirect -->
+			<param name="work-mode" value="direct"/>
+			<!-- OSP service type, voice or npquery -->
+			<param name="service-type" value="voice"/>
+			<!-- Max number of destinations -->
+			<param name="max-destinations" value="12"/>
+		</profile>
+	</profiles>
+</configuration>

src/mod/applications/mod_redis/conf/autoload_configs/redis.conf.xml

@@ -0,0 +1,7 @@
+<configuration name="redis.conf" description="mod_redis Configuration">
+  <settings>
+    <param name="host" value="localhost"/>
+    <param name="port" value="6379"/>
+    <param name="timeout" value="10000"/>
+  </settings>
+</configuration>

src/mod/applications/mod_rss/conf/autoload_configs/rss.conf.xml

@@ -0,0 +1,7 @@
+<configuration name="rss.conf" description="RSS Parser">
+  <feeds>
+    <!-- Just download the files to wherever and refer to them here -->
+    <!-- <feed name="Slash Dot">/home/rss/rss.rss</feed> -->
+    <!-- <feed name="News Forge">/home/rss/newsforge.rss</feed> -->
+  </feeds>
+</configuration>

src/mod/applications/mod_rss/conf/dialplan/rss.xml

@@ -0,0 +1,7 @@
+<include>
+	<extension name="rss">
+		<condition field="destination_number" expression="^rss$">
+			<action application="rss"/>
+		</condition>
+	</extension>
+</include>

src/mod/applications/mod_rss/script/news.js

@@ -0,0 +1,12 @@
+if (session.ready()) {
+   session.answer();
+   session.speak("cepstral","David","Please wait while we refresh the RSS feeds.")
+
+   fetchURLFile("http://weather.yahooapis.com/forecastrss?p=60610","rss/weather.rss");
+   fetchURLFile("http://rss.news.yahoo.com/rss/topstories","rss/yahootop.rss");
+   fetchURLFile("http://rss.news.yahoo.com/rss/science","rss/yahoosci.rss");
+   fetchURLFile("http://rss.news.yahoo.com/rss/business","rss/yahoobus.rss");
+   fetchURLFile("http://rss.news.yahoo.com/rss/entertainment","rss/yahooent.rss");
+   fetchURLFile("http://rss.slashdot.org/Slashdot/slashdot","rss/slashdot.rss");
+   fetchURLFile("http://www.freeswitch.org/xml.php","rss/freeswitch.rss");
+}

src/mod/applications/mod_sms/mod_sms.2010.vcxproj

@@ -1,4 +1,4 @@
-<?xml version="1.0" encoding="utf-8"?>
+<?xml version="1.0" encoding="utf-8"?>
 <Project DefaultTargets="Build" ToolsVersion="4.0" xmlns="http://schemas.microsoft.com/developer/msbuild/2003">
   <ItemGroup Label="ProjectConfigurations">
     <ProjectConfiguration Include="Debug|Win32">
@@ -20,7 +20,7 @@
   </ItemGroup>
   <PropertyGroup Label="Globals">
     <ProjectName>mod_sms</ProjectName>
-    <ProjectGuid>{11C9BC3D-45E9-46E3-BE84-B8CEE4685E39}</ProjectGuid>
+    <ProjectGuid>{2469B306-B027-4FF2-8815-C9C1EA2CAE79}</ProjectGuid>
     <RootNamespace>mod_sms</RootNamespace>
     <Keyword>Win32Proj</Keyword>
   </PropertyGroup>
@@ -63,10 +63,6 @@
   <PropertyGroup Label="UserMacros" />
   <PropertyGroup>
     <_ProjectFileVersion>10.0.30319.1</_ProjectFileVersion>
-    <OutDir Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">$(Platform)\$(Configuration)\</OutDir>
-    <IntDir Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">$(Platform)\$(Configuration)\</IntDir>
-    <OutDir Condition="'$(Configuration)|$(Platform)'=='Release|x64'">$(Platform)\$(Configuration)\</OutDir>
-    <IntDir Condition="'$(Configuration)|$(Platform)'=='Release|x64'">$(Platform)\$(Configuration)\</IntDir>
   </PropertyGroup>
   <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">
     <ClCompile>
@@ -132,4 +128,4 @@
   <Import Project="$(VCTargetsPath)\Microsoft.Cpp.targets" />
   <ImportGroup Label="ExtensionTargets">
   </ImportGroup>
-</Project>
+</Project>

src/mod/applications/mod_snom/conf/dialplan/snom_demo.xml

@@ -0,0 +1,16 @@
+<include>		
+	<extension name="snom-demo-1">
+		<condition field="destination_number" expression="^9000$">
+			<!-- <key> <light> <label> <user> <host> <profile> <action_name> <action> -->
+			<action application="eval" data="${snom_bind_key(2 on DND ${sip_from_user} ${sip_from_host} ${sofia_profile_name} message api+uuid_transfer ${uuid} 9001)}"/>
+			<action application="playback" data="$${hold_music}"/>
+		</condition>
+	</extension>
+
+	<extension name="snom-demo-2">
+		<condition field="destination_number" expression="^9001$">
+			<action application="eval" data="${snom_bind_key(2 off DND ${sip_from_user} ${sip_from_host} ${sofia_profile_name} message notused)}"/>
+			<action application="transfer" data="3000"/>
+		</condition>
+	</extension>
+</include>

src/mod/applications/mod_spandsp/conf/autoload_configs/spandsp.conf.xml

@@ -0,0 +1,108 @@
+<configuration name="spandsp.conf" description="SpanDSP config">
+    <modem-settings>
+<!--
+    total-modems set to N will create that many soft-modems.  
+    If you use them with Hylafax you need the following for each one numbered 0..N:
+
+    1) A line like this in /etc/inittab:
+
+      f0:2345:respawn:/usr/lib/fax/faxgetty /dev/FS0
+
+    2) copy conf/config.FS0 to /var/spool/hylafax/etc (or wherver the appropriate dir is on your system)
+
+    Subsequent modem configs would incrment the 0 to 1 and so on.
+
+-->
+      <param name="total-modems" value="0"/>
+
+      <!-- Default context and dialplan to use on inbound calls from the modems -->
+      <param name="context" value="default"/>
+      <param name="dialplan" value="XML"/>
+
+      <!-- Extra tracing for debugging -->
+      <param name="verbose" value="false"/>
+    </modem-settings>
+
+    <fax-settings>
+	<param name="use-ecm"		value="true"/>
+	<param name="verbose"		value="false"/>
+	<param name="disable-v17"	value="false"/>
+	<param name="ident"		value="SpanDSP Fax Ident"/>
+	<param name="header"		value="SpanDSP Fax Header"/>
+
+	<param name="spool-dir"		value="/tmp"/>
+	<param name="file-prefix"	value="faxrx"/>
+    </fax-settings>
+
+    <descriptors>
+
+     <!-- These tones are defined in Annex to ITU Operational Bulletin No. 781 - 1.II.2003 -->
+     <!-- Various Tones Used in National Networks (According to ITU-T Recommendation E.180)(03/1998) -->
+
+     <!-- North America -->
+     <descriptor name="1">
+       <tone name="CED_TONE">
+         <element freq1="2100" freq2="0" min="500" max="0"/>
+       </tone>
+       <tone name="SIT">
+         <element freq1="950" freq2="0" min="256" max="400"/>
+         <element freq1="1400" freq2="0" min="256" max="400"/>
+         <element freq1="1800" freq2="0" min="256" max="400"/>
+       </tone>
+       <tone name="REORDER_TONE">
+         <element freq1="480" freq2="620" min="224" max="272"/>
+         <element freq1="0" freq2="0" min="224" max="272"/>
+       </tone>
+       <tone name="BUSY_TONE">
+         <element freq1="480" freq2="620" min="464" max="516"/>
+         <element freq1="0" freq2="0" min="464" max="516"/>
+       </tone>
+     </descriptor>
+
+     <!-- United Kingdom -->
+     <descriptor name="44">
+       <tone name="CED_TONE">
+         <element freq1="2100" freq2="0" min="500" max="0"/>
+       </tone>
+       <tone name="SIT">
+         <element freq1="950" freq2="0" min="256" max="400"/>
+         <element freq1="1400" freq2="0" min="256" max="400"/>
+         <element freq1="1800" freq2="0" min="256" max="400"/>
+       </tone>
+       <tone name="REORDER_TONE">
+         <element freq1="400" freq2="0" min="368" max="416"/>
+         <element freq1="0" freq2="0" min="336" max="368"/>
+         <element freq1="400" freq2="0" min="256" max="288"/>
+         <element freq1="0" freq2="0" min="512" max="544"/>
+       </tone>
+       <tone name="BUSY_TONE">
+         <element freq1="400" freq2="0" min="352" max="384"/>
+         <element freq1="0" freq2="0" min="352" max="384"/>
+         <element freq1="400" freq2="0" min="352" max="384"/>
+         <element freq1="0" freq2="0" min="352" max="384"/>
+       </tone>
+     </descriptor>
+
+     <!-- Germany -->
+     <descriptor name="49">
+       <tone name="CED_TONE">
+         <element freq1="2100" freq2="0" min="500" max="0"/>
+       </tone>
+       <tone name="SIT">
+         <element freq1="900" freq2="0" min="256" max="400"/>
+         <element freq1="1400" freq2="0" min="256" max="400"/>
+         <element freq1="1800" freq2="0" min="256" max="400"/>
+       </tone>
+       <tone name="REORDER_TONE">
+         <element freq1="425" freq2="0" min="224" max="272"/>
+         <element freq1="0" freq2="0" min="224" max="272"/>
+       </tone>
+       <tone name="BUSY_TONE">
+         <element freq1="425" freq2="0" min="464" max="516"/>
+         <element freq1="0" freq2="0" min="464" max="516"/>
+       </tone>
+     </descriptor>
+   </descriptors>
+
+</configuration>
+

src/mod/applications/mod_spandsp/conf/config.FS0

@@ -0,0 +1,78 @@
+CountryCode:		1
+AreaCode:		800
+FAXNumber:		+1.800.555.1212
+LongDistancePrefix:	1
+InternationalPrefix:	011
+DialStringRules:	etc/dialrules
+ServerTracing:		0xFFF
+SessionTracing:		0xFFF
+RecvFileMode:		0600
+LogFileMode:		0600
+DeviceMode:		0600
+RingsBeforeAnswer:	1
+SpeakerVolume:		off
+GettyArgs:		"-h %l dx_%s"
+LocalIdentifier:	"FS"
+TagLineFont:		etc/lutRS18.pcf
+TagLineFormat:		"From %%l|%c|Page %%P of %%T"
+MaxRecvPages:		200
+#
+#
+# Modem-related stuff: should reflect modem command interface
+# and hardware connection/cabling (e.g. flow control).
+#
+ModemType:		Class1		# use this to supply a hint
+
+#
+# Enabling this will use the hfaxd-protocol to set Caller*ID
+#
+#ModemSetOriginCmd:	AT+VSID="%s","%d"
+
+#
+# If "glare" during initialization becomes a problem then take
+# the modem off-hook during initialization, and then place it
+# back on-hook when done.
+#
+#ModemResetCmds:	"ATH1\nAT+VCID=1"	# enables CallID display
+#ModemReadyCmds:	ATH0
+
+Class1AdaptRecvCmd:	AT+FAR=1
+Class1TMConnectDelay:	400		# counteract quick CONNECT response
+
+#
+# If you have trouble with V.17 receiving or sending,
+# you may want to enable one of these, respectively.
+#
+#Class1RMQueryCmd:      "!24,48,72,96"  # enable this to disable V.17 receiving
+#Class1TMQueryCmd:      "!24,48,72,96"  # enable this to disable V.17 sending
+
+#
+# You'll likely want Caller*ID display (also displays DID) enabled.
+#
+ModemResetCmds:		AT+VCID=1	# enables CallID display
+
+#
+# The pty does not support changing parity.
+#
+PagerTTYParity:		none
+
+#
+# If you are "missing" Caller*ID data on some calls (but not all)
+# and if you do not have adequate glare protection you may want to
+# not answer based on RINGs, but rather enable the CallIDAnswerLength
+# for NDID, disable AT+VCID=1 and do this:
+#
+#RingsBeforeAnswer: 0
+#ModemRingResponse: AT+VRID=1
+
+# Uncomment DATE and TIME if you really want them, but you probably don't.
+#CallIDPattern:          "DATE="
+#CallIDPattern:          "TIME="
+CallIDPattern:          "NMBR="
+CallIDPattern:          "NAME="
+CallIDPattern:		"ANID="
+#CallIDPattern:          "USER="	# username provided by call
+#CallIDPattern:          "PASS="	# password provided by call
+#CallIDPattern:          "CDID="	# DID context in call
+CallIDPattern:          "NDID="
+#CallIDAnswerLength:	4

src/mod/applications/mod_spandsp/conf/dialplan/spandsp.xml

@@ -0,0 +1,26 @@
+    <extension name="fax_receive">
+      <condition field="destination_number" expression="^9178$">
+        <action application="answer" />
+        <action application="playback" data="silence_stream://2000"/>
+        <action application="rxfax" data="/tmp/rxfax.tif"/>
+        <action application="hangup"/>
+      </condition>
+    </extension>
+
+    <extension name="fax_transmit">
+      <condition field="destination_number" expression="^9179$">
+        <action application="txfax" data="/tmp/txfax.tif"/>
+        <action application="hangup"/>
+      </condition>
+    </extension>
+
+    <extension name="fax_receive">
+      <condition field="destination_number" expression="^9978$">
+        <action application="answer" />
+        <action application="playback" data="silence_stream://2000"/>
+        <action application="set" data="fax_enable_t38_request=true"/>
+        <action application="set" data="fax_enable_t38=true"/>
+        <action application="rxfax" data="/tmp/FAX-${uuid}.tif"/>
+        <action application="hangup"/>
+      </condition>
+    </extension>

src/mod/applications/mod_spandsp/mod_spandsp.c

@@ -337,7 +337,7 @@ switch_status_t load_configuration(switch_bool_t reload)
     spandsp_globals.timezone = "";
 
 	/* TODO make configuration param */
-	spandsp_globals.tonedebug = 1;
+	spandsp_globals.tonedebug = 0;
 
 	if ((xml = switch_xml_open_cfg("spandsp.conf", &cfg, NULL)) || (xml = switch_xml_open_cfg("fax.conf", &cfg, NULL))) {
         status = SWITCH_STATUS_SUCCESS;

src/mod/applications/mod_spandsp/mod_spandsp.h

@@ -137,6 +137,6 @@ switch_status_t spandsp_fax_detect_session(switch_core_session_t *session,
 														   int hits, const char *app, const char *data, switch_tone_detect_callback_t callback);
 
 switch_status_t spandsp_fax_stop_detect_session(switch_core_session_t *session);
-void spanfax_log_message(int level, const char *msg);
+void spanfax_log_message(void *user_data, int level, const char *msg);
 switch_status_t load_configuration(switch_bool_t reload);
 void mod_spandsp_indicate_data(switch_core_session_t *session, switch_bool_t self, switch_bool_t on);

src/mod/applications/mod_spandsp/mod_spandsp_fax.c

@@ -259,7 +259,7 @@ static void counter_increment(void)
 	switch_mutex_unlock(spandsp_globals.mutex);
 }
 
-void spanfax_log_message(int level, const char *msg)
+void spanfax_log_message(void *user_data, int level, const char *msg)
 {
 	int fs_log_level;
 
@@ -341,6 +341,7 @@ static int phase_b_handler(t30_state_t *s, void *user_data, int result)
 	/* Fire event */
 
 	if (switch_event_create_subclass(&event, SWITCH_EVENT_CUSTOM, pvt->app_mode == FUNCTION_TX ? SPANDSP_EVENT_TXFAXNEGOCIATERESULT : SPANDSP_EVENT_RXFAXNEGOCIATERESULT) == SWITCH_STATUS_SUCCESS) {
+		switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "uuid", switch_core_session_get_uuid(session));
 		switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "fax-transfer-rate", fax_transfer_rate);
 		switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "fax-ecm-used", (t30_stats.error_correcting_mode) ? "on" : "off");
 		switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "fax-local-station-id", local_ident);
@@ -437,6 +438,7 @@ static int phase_d_handler(t30_state_t *s, void *user_data, int msg)
 	switch_channel_execute_on(channel, "execute_on_fax_phase_d");
 
 	if (switch_event_create_subclass(&event, SWITCH_EVENT_CUSTOM, pvt->app_mode == FUNCTION_TX ? SPANDSP_EVENT_TXFAXPAGERESULT : SPANDSP_EVENT_RXFAXPAGERESULT) == SWITCH_STATUS_SUCCESS) {
+		switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "uuid", switch_core_session_get_uuid(session));
 		switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "fax-document-transferred-pages", fax_document_transferred_pages);
 		switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "fax-image-resolution", fax_image_resolution);
 		switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "fax-image-size", fax_image_size);
@@ -711,8 +713,8 @@ static switch_status_t spanfax_init(pvt_t *pvt, transport_mode_t trans_mode)
 
 		fax_set_transmit_on_idle(fax, TRUE);
 
-		span_log_set_message_handler(&fax->logging, spanfax_log_message);
-		span_log_set_message_handler(&t30->logging, spanfax_log_message);
+		span_log_set_message_handler(&fax->logging, spanfax_log_message, NULL);
+		span_log_set_message_handler(&t30->logging, spanfax_log_message, NULL);
 
 		if (pvt->verbose) {
 			span_log_set_level(&fax->logging, SPAN_LOG_SHOW_SEVERITY | SPAN_LOG_SHOW_PROTOCOL | SPAN_LOG_FLOW);
@@ -767,8 +769,8 @@ static switch_status_t spanfax_init(pvt_t *pvt, transport_mode_t trans_mode)
                 }
             }
         
-            span_log_set_message_handler(&t38->logging, spanfax_log_message);
-            span_log_set_message_handler(&t30->logging, spanfax_log_message);
+            span_log_set_message_handler(&t38->logging, spanfax_log_message, NULL);
+            span_log_set_message_handler(&t30->logging, spanfax_log_message, NULL);
 
             if (pvt->verbose) {
                 span_log_set_level(&t38->logging, SPAN_LOG_SHOW_SEVERITY | SPAN_LOG_SHOW_PROTOCOL | SPAN_LOG_FLOW);
@@ -822,8 +824,8 @@ static switch_status_t spanfax_init(pvt_t *pvt, transport_mode_t trans_mode)
 	 }
 	 
 
-     span_log_set_message_handler(&pvt->t38_gateway_state->logging, spanfax_log_message);
-     span_log_set_message_handler(&pvt->t38_core->logging, spanfax_log_message);
+     span_log_set_message_handler(&pvt->t38_gateway_state->logging, spanfax_log_message, NULL);
+     span_log_set_message_handler(&pvt->t38_core->logging, spanfax_log_message, NULL);
 
 	 if (pvt->verbose) {
 		 span_log_set_level(&pvt->t38_gateway_state->logging, SPAN_LOG_SHOW_SEVERITY | SPAN_LOG_SHOW_PROTOCOL | SPAN_LOG_FLOW);

src/mod/applications/mod_spandsp/mod_spandsp_modem.c

@@ -346,15 +346,15 @@ switch_status_t modem_init(modem_t *modem, modem_control_handler_t control_handl
 	}
 
 	if (spandsp_globals.modem_verbose) {
-		span_log_set_message_handler(&modem->t31_state->logging, spanfax_log_message);
-		span_log_set_message_handler(&modem->t31_state->audio.modems.v17_rx.logging, spanfax_log_message);
-		span_log_set_message_handler(&modem->t31_state->audio.modems.v29_rx.logging, spanfax_log_message);
-		span_log_set_message_handler(&modem->t31_state->audio.modems.v27ter_rx.logging, spanfax_log_message);
+		span_log_set_message_handler(&modem->t31_state->logging, spanfax_log_message, NULL);
+		span_log_set_message_handler(&modem->t31_state->audio.modems.fast_modems.v17_rx.logging, spanfax_log_message, NULL);
+		span_log_set_message_handler(&modem->t31_state->audio.modems.fast_modems.v29_rx.logging, spanfax_log_message, NULL);
+		span_log_set_message_handler(&modem->t31_state->audio.modems.fast_modems.v27ter_rx.logging, spanfax_log_message, NULL);
 
 		modem->t31_state->logging.level = SPAN_LOG_SHOW_SEVERITY | SPAN_LOG_SHOW_PROTOCOL | SPAN_LOG_FLOW;
-		modem->t31_state->audio.modems.v17_rx.logging.level = SPAN_LOG_SHOW_SEVERITY | SPAN_LOG_SHOW_PROTOCOL | SPAN_LOG_FLOW;
-		modem->t31_state->audio.modems.v29_rx.logging.level = SPAN_LOG_SHOW_SEVERITY | SPAN_LOG_SHOW_PROTOCOL | SPAN_LOG_FLOW;
-		modem->t31_state->audio.modems.v27ter_rx.logging.level = SPAN_LOG_SHOW_SEVERITY | SPAN_LOG_SHOW_PROTOCOL | SPAN_LOG_FLOW;
+		modem->t31_state->audio.modems.fast_modems.v17_rx.logging.level = SPAN_LOG_SHOW_SEVERITY | SPAN_LOG_SHOW_PROTOCOL | SPAN_LOG_FLOW;
+		modem->t31_state->audio.modems.fast_modems.v29_rx.logging.level = SPAN_LOG_SHOW_SEVERITY | SPAN_LOG_SHOW_PROTOCOL | SPAN_LOG_FLOW;
+		modem->t31_state->audio.modems.fast_modems.v27ter_rx.logging.level = SPAN_LOG_SHOW_SEVERITY | SPAN_LOG_SHOW_PROTOCOL | SPAN_LOG_FLOW;
 	}
 
 	modem->control_handler = control_handler;

src/mod/applications/mod_valet_parking/conf/dialplan/valet_parking.xml

@@ -0,0 +1,24 @@
+<include>
+    <!--
+         Valet park retrieval, works with valet_park extension below.
+         Retrieve a valet parked call by dialing 6000 + park number + #
+    -->
+    <extension name="valet_park">
+      <condition field="destination_number" expression="^(6000)$">
+        <action application="answer"/>
+        <action application="valet_park" data="valet_parking_lot ask 1 11 10000 ivr/ivr-enter_ext_pound.wav"/>
+      </condition>
+    </extension>
+
+    <!--
+         Valet park 6001-6099.  Blind x-fer to 6001, 6002, etc. to valet park the call.
+         Dial 6001, 6002, etc. to retrieve a call that is already valet parked.
+         After call is retrieved, park extension is free for another call.
+    -->
+    <extension name="valet_park">
+      <condition field="destination_number" expression="^(60\d[1-9])$">
+        <action application="answer"/>
+        <action application="valet_park" data="valet_parking_lot $1"/>
+      </condition>
+    </extension>
+</include>

src/mod/applications/mod_vmd/conf/dialplan/vmd.xml

@@ -0,0 +1,12 @@
+<include>
+	<extension name="vmdtest">
+		<condition field="destination_number" expression="^vmd(\d{10})$">
+			<action application="answer"/>
+			<action application="info"/>
+			<action application="vmd"/>
+			<action application="sleep" data="25000"/>
+			<action application="info"/> <!-- Look for chan var "vmd_detect" here -->
+			<action application="hangup"/>
+		</condition>
+	</extension>
+</include>

src/mod/applications/mod_vmd/scripts/vmd.js

@@ -0,0 +1,27 @@
+function onInput(s, type, obj, arg)
+{
+	try
+	{
+		if(type == "dtmf")
+		{
+			console_log("info", "DTMF digit: "+s.name+" ["+obj.digit+"] len ["+obj.duration+"]\n");
+		}
+		else if(type == "event" && session.getVariable("vmd_detect") == "TRUE")
+		{
+			console_log("info", "Voicemail Detected\n");
+		}
+		
+	}
+	catch(e)
+	{
+		console_log("err", e + "\n");
+	}
+	return true;
+}
+
+session.answer();
+session.execute("vmd", "start");
+while(session.ready())
+{
+	session.streamFile(argv[0], onInput);
+}

src/mod/applications/mod_vmd/scripts/vmd.lua

@@ -0,0 +1,17 @@
+local human_detected = false;
+local voicemail_detected = false;
+
+function onInput(session, type, obj)
+    if type == "dtmf" and obj['digit'] == '1' and human_detected == false then
+        human_detected = true;
+        return "break";
+    end
+
+    if type == "event" and voicemail_detected == false then
+        voicemail_detected = true;
+        return "break";
+    end
+end
+
+session:setInputCallback("onInput");
+session:execute("vmd","start");

src/mod/applications/mod_voicemail/conf/autoload_configs/voicemail.conf.xml

@@ -0,0 +1,70 @@
+<configuration name="voicemail.conf" description="Voicemail">
+  <settings>
+  </settings>
+  <profiles>
+    <profile name="default">
+      <param name="file-extension" value="wav"/>
+      <param name="terminator-key" value="#"/>
+      <param name="max-login-attempts" value="3"/>
+      <param name="digit-timeout" value="10000"/>
+      <param name="min-record-len" value="3"/>
+      <param name="max-record-len" value="300"/>
+      <param name="max-retries" value="3"/>
+      <param name="tone-spec" value="%(1000, 0, 640)"/>
+      <param name="callback-dialplan" value="XML"/>
+      <param name="callback-context" value="default"/>
+      <param name="play-new-messages-key" value="1"/>
+      <param name="play-saved-messages-key" value="2"/>
+      <param name="login-keys" value="0"/>
+      <param name="main-menu-key" value="0"/>
+      <param name="config-menu-key" value="5"/>
+      <param name="record-greeting-key" value="1"/>
+      <param name="choose-greeting-key" value="2"/>
+      <param name="change-pass-key" value="6"/>
+      <param name="record-name-key" value="3"/>
+      <param name="record-file-key" value="3"/>
+      <param name="listen-file-key" value="1"/>
+      <param name="save-file-key" value="2"/>
+      <param name="delete-file-key" value="7"/>
+      <param name="undelete-file-key" value="8"/>
+      <param name="email-key" value="4"/>
+      <param name="pause-key" value="0"/>
+      <param name="restart-key" value="1"/>
+      <param name="ff-key" value="6"/>
+      <param name="rew-key" value="4"/>
+      <param name="skip-greet-key" value="#"/>
+      <param name="previous-message-key" value="1"/>
+      <param name="next-message-key" value="3"/>
+      <param name="skip-info-key" value="*"/>
+      <param name="repeat-message-key" value="0"/>
+      <param name="record-silence-threshold" value="200"/>
+      <param name="record-silence-hits" value="2"/>
+      <param name="web-template-file" value="web-vm.tpl"/>
+      <param name="db-password-override" value="false"/>
+      <param name="allow-empty-password-auth" value="true"/>
+      <!-- if you need to change the sample rate of the recorded files e.g. gmail voicemail player -->
+      <!--<param name="record-sample-rate" value="11025"/>-->
+      <!-- the next two both must be set for this to be enabled
+           the extension is in the format of <dest> [<dialplan>] [<context>]
+       -->
+      <param name="operator-extension" value="operator XML default"/>
+      <param name="operator-key" value="9"/>
+      <param name="vmain-extension" value="vmain XML default"/>
+      <param name="vmain-key" value="*"/>
+      <!-- playback created files as soon as they were recorded by default -->
+      <!--<param name="auto-playback-recordings" value="true"/>-->
+      <email>
+	<param name="template-file" value="voicemail.tpl"/>
+	<param name="notify-template-file" value="notify-voicemail.tpl"/>
+	<!-- this is the format voicemail_time will have -->
+        <param name="date-fmt" value="%A, %B %d %Y, %I %M %p"/>
+        <param name="email-from" value="${voicemail_account}@${voicemail_domain}"/>
+      </email>
+      <!--<param name="storage-dir" value="/tmp"/>-->
+      <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
+      <!--<param name="record-comment" value="Your Comment"/>-->
+      <!--<param name="record-title" value="Your Title"/>-->
+      <!--<param name="record-copyright" value="Your Copyright"/>-->
+    </profile>
+  </profiles>
+</configuration> 

src/mod/applications/mod_voicemail/conf/dialplan/voicemail.xml

@@ -0,0 +1,20 @@
+<include>
+	<!-- This is a _VERY_ abbreviated demo for voicemail. A better demo would be useful -->
+
+	<!-- voicemail main extension -->
+	<extension name="vmain">
+		<condition field="destination_number" expression="^vmain$|^4000$|^\*98$">
+			<action application="answer"/>
+			<action application="sleep" data="1000"/>
+			<action application="voicemail" data="check default ${domain_name}"/>
+		</condition>
+	</extension>
+
+	<extension name="Local_Voicemail">
+		<condition field="destination_number" expression="^(10[01][0-9])$">
+			<action application="export" data="dialed_extension=$1"/>
+			<action application="export" data="domain=default"/>
+			<action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/>
+		</condition>
+	</extension>
+</include>

src/mod/applications/mod_voicemail/conf/notify-voicemail.tpl

@@ -0,0 +1,44 @@
+From: "${voicemail_caller_id_name}" <${voicemail_caller_id_number}@${voicemail_domain}>
+Date: ${RFC2822_DATE}
+To: <${voicemail_notify_email}>
+Subject: Voicemail from "${voicemail_caller_id_name}" <${voicemail_caller_id_number}> ${voicemail_message_len}
+X-Priority: ${voicemail_priority}
+X-Mailer: FreeSWITCH
+
+Content-Type: multipart/alternative; 
+	boundary="000XXX000"
+
+--000XXX000
+Content-Type: text/plain; charset=ISO-8859-1; Format=Flowed
+Content-Disposition: inline
+Content-Transfer-Encoding: 7bit
+
+Created: ${voicemail_time}
+From: "${voicemail_caller_id_name}" <${voicemail_caller_id_number}>
+Duration: ${voicemail_message_len}
+Account: ${voicemail_account}@${voicemail_domain}
+
+--000XXX000
+Content-Type: text/html; charset=ISO-8859-1
+Content-Disposition: inline
+Content-Transfer-Encoding: 7bit
+
+<html xmlns="http://www.w3.org/1999/xhtml">
+<head>
+<META http-equiv="Content-Type" content="text/html; charset=ISO-8859-1">
+<title>Voicemail from "${voicemail_caller_id_name}" <${voicemail_caller_id_number}> ${voicemail_message_len}</title>
+<meta content="text/html; charset=iso-8859-1" http-equiv="content-type"/>
+</head>
+<body>
+
+<font face=arial>
+<b>Message From "${voicemail_caller_id_name}" <A HREF="tel:${voicemail_caller_id_number}">${voicemail_caller_id_number}</A></b><br>
+<hr noshade size=1>
+Created: ${voicemail_time}<br>
+Duration: ${voicemail_message_len}<br>
+Account: ${voicemail_account}@${voicemail_domain}<br>
+</font>
+
+</body>
+</html>
+--000XXX000--

src/mod/applications/mod_voicemail/conf/voicemail.tpl

@@ -0,0 +1,44 @@
+From: "${voicemail_caller_id_name}" <${voicemail_caller_id_number}@${voicemail_domain}>
+Date: ${RFC2822_DATE}
+To: <${voicemail_email}>
+Subject: Voicemail from "${voicemail_caller_id_name}" <${voicemail_caller_id_number}> ${voicemail_message_len}
+X-Priority: ${voicemail_priority}
+X-Mailer: FreeSWITCH
+
+Content-Type: multipart/alternative; 
+	boundary="000XXX000"
+
+--000XXX000
+Content-Type: text/plain; charset=ISO-8859-1; Format=Flowed
+Content-Disposition: inline
+Content-Transfer-Encoding: 7bit
+
+Created: ${voicemail_time}
+From: "${voicemail_caller_id_name}" <${voicemail_caller_id_number}>
+Duration: ${voicemail_message_len}
+Account: ${voicemail_account}@${voicemail_domain}
+
+--000XXX000
+Content-Type: text/html; charset=ISO-8859-1
+Content-Disposition: inline
+Content-Transfer-Encoding: 7bit
+
+<html xmlns="http://www.w3.org/1999/xhtml">
+<head>
+<META http-equiv="Content-Type" content="text/html; charset=ISO-8859-1">
+<title>Voicemail from "${voicemail_caller_id_name}" <${voicemail_caller_id_number}> ${voicemail_message_len}</title>
+<meta content="text/html; charset=iso-8859-1" http-equiv="content-type"/>
+</head>
+<body>
+
+<font face=arial>
+<b>Message From "${voicemail_caller_id_name}" <A HREF="tel:${voicemail_caller_id_number}">${voicemail_caller_id_number}</A></b><br>
+<hr noshade size=1>
+Created: ${voicemail_time}<br>
+Duration: ${voicemail_message_len}<br>
+Account: ${voicemail_account}@${voicemail_domain}<br>
+</font>
+
+</body>
+</html>
+--000XXX000--

src/mod/applications/mod_voicemail/mod_voicemail.c

@@ -1544,8 +1544,6 @@ static switch_status_t listen_file(switch_core_session_t *session, vm_profile_t
 	char cid_buf[1024] = "";
 
 	if (switch_channel_ready(channel)) {
-		const char *vm_announce_cid = NULL;
-
 		switch_snprintf(cid_buf, sizeof(cid_buf), "%s|%s", cbt->cid_number, cbt->cid_name);
 
 		msg.from = __FILE__;
@@ -1556,10 +1554,8 @@ static switch_status_t listen_file(switch_core_session_t *session, vm_profile_t
 						  cid_buf, switch_channel_get_name(channel));
 		switch_core_session_receive_message(session, &msg);
 		
-		if (!zstr(cbt->cid_number) && (vm_announce_cid = switch_channel_get_variable(channel, "vm_announce_cid"))) {
-			switch_ivr_play_file(session, NULL, vm_announce_cid, NULL);
-			switch_ivr_sleep(session, 500, SWITCH_TRUE, NULL);
-			switch_ivr_say(session, cbt->cid_number, NULL, "name_spelled", "pronounced", NULL, NULL);
+		if (!zstr(cbt->cid_number) && (switch_true(switch_channel_get_variable(channel, "vm_announce_cid")))) {
+			TRY_CODE(switch_ivr_phrase_macro(session, VM_SAY_PHONE_NUMBER_MACRO, cbt->cid_number, NULL, NULL));
 		}
 		
 		args.input_callback = cancel_on_dtmf;
@@ -1865,6 +1861,7 @@ static void voicemail_check_main(switch_core_session_t *session, vm_profile_t *p
 	const char *caller_id_name = NULL;
 	const char *caller_id_number = NULL;
 	int auth_only = 0, authed = 0;
+	switch_event_t *event;
 
 	if (!(caller_id_name = switch_channel_get_variable(channel, "effective_caller_id_name"))) {
 		caller_id_name = caller_profile->caller_id_name;
@@ -2458,6 +2455,14 @@ static void voicemail_check_main(switch_core_session_t *session, vm_profile_t *p
 					}
 				}
 
+				switch_event_create_subclass(&event, SWITCH_EVENT_CUSTOM, VM_EVENT_MAINT);
+				switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "VM-Action", "authentication");
+				switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "VM-Auth-Result", auth ? "success" : "fail");
+				switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "VM-User", myid);
+				switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "VM-Domain", domain_name);
+				switch_channel_event_set_data(channel, event);
+				switch_event_fire(&event);
+
 				FREE_DOMAIN_ROOT();
 
 				if (auth) {
@@ -4378,7 +4383,7 @@ SWITCH_STANDARD_API(voicemail_inject_api_function)
 static int api_del_callback(void *pArg, int argc, char **argv, char **columnNames)
 {
 
-	unlink(argv[2]);
+	unlink(argv[3]);
 	
     return 0;
 }

src/mod/applications/mod_voicemail_ivr/conf/autoload_configs/voicemail_ivr.conf.xml

@@ -0,0 +1,208 @@
+<configuration name="voicemail_ivr.conf" description="Voicemail IVR">
+<profiles>
+	<profile name="default">
+		<settings>
+			<param name="IVR-Maximum-Attempts" value="3" />
+			<param name="IVR-Entry-Timeout" value="3000" />
+			<param name="Record-Format" value="wav" />
+			<!--<param name="Record-Sample-Rate" value="8000" />-->
+			<param name="Record-Silence-Hits" value="4" />
+			<param name="Record-Silence-Threshold" value="200" />
+			<param name="Record-Maximum-Length" value="30" />
+			<!--<param name="Record-Minimum-Length" value="3" />-->
+			<param name="Exit-Purge" value="true" />
+			<param name="Password-Mask" value="XXX." />
+			<param name="User-Mask" value="X." />
+
+		</settings>
+		<apis>
+			<api name="auth_login" value="vm_fsdb_auth_login" />
+			<api name="msg_list" value="vm_fsdb_msg_list" />
+			<api name="msg_count" value="vm_fsdb_msg_count" />
+			<api name="msg_delete" value="vm_fsdb_msg_delete" />
+			<api name="msg_undelete" value="vm_fsdb_msg_undelete" />
+			<api name="msg_save" value="vm_fsdb_msg_save" />
+			<api name="msg_purge" value="vm_fsdb_msg_purge" />
+			<api name="msg_get" value="vm_fsdb_msg_get" />
+			<api name="msg_forward" value="vm_fsdb_msg_forward" />
+			<api name="pref_greeting_set" value="vm_fsdb_pref_greeting_set" />
+			<api name="pref_recname_set" value="vm_fsdb_pref_recname_set" />
+			<api name="pref_password_set" value="vm_fsdb_pref_password_set" />
+		</apis>
+		<menus>
+			<menu name="std_authenticate">
+			<phrases>
+				<phrase name="fail_auth" value="fail_auth@voicemail_ivr" />
+			</phrases>
+			<keys>
+			</keys>
+			</menu>
+
+			<menu name="std_authenticate_ask_user">
+			<phrases>
+				<phrase name="instructions" value="enter_id@voicemail_ivr" />
+			</phrases>
+			<keys>
+				<key dtmf="#" action="ivrengine:terminate_entry" variable="VM-Key-Terminator" />
+			</keys>
+			</menu>
+
+			<menu name="std_authenticate_ask_password">
+			<phrases>
+				<phrase name="instructions" value="enter_pass@voicemail_ivr" />
+			</phrases>
+			<keys>
+				<key dtmf="#" action="ivrengine:terminate_entry" variable="VM-Key-Terminator" />
+			</keys>
+			</menu>
+
+			<menu name="std_main_menu">
+			<phrases>
+				<phrase name="msg_count" value="message_count@mtvoicemail" />
+				<phrase name="say_date" value="say_date_event@mtvoicemail" />
+				<phrase name="say_msg_number" value="say_message_number@mtvoicemail" />
+				<phrase name="menu_options" value="menu@mtvoicemail" />
+			</phrases>
+			<keys>
+				<key dtmf="1" action="new_msg:std_navigator" variable="VM-Key-Play-New-Messages" />
+				<key dtmf="2" action="saved_msg:std_navigator" variable="VM-Key-Play-Saved-Messages" />
+				<key dtmf="5" action="menu:std_preference" variable="VM-Key-Config-Menu"/>
+				<key dtmf="#" action="return" variable="VM-Key-Terminator" />
+			</keys>
+			</menu>
+
+			<menu name="std_navigator">
+			<settings>
+				<!--<param name="Nav-Action-On-Delete" value="next_msg" />-->
+			</settings>
+			<phrases>
+				<phrase name="msg_count" value="message_count@voicemail_ivr" />
+				<phrase name="say_date" value="say_date_event@voicemail_ivr" />
+				<phrase name="say_msg_number" value="say_message_number@voicemail_ivr" />
+				<phrase name="menu_options" value="listen_file_check@voicemail_ivr" />
+				<phrase name="ack" value="ack@voicemail_ivr" />
+				<phrase name="play_message" value="play_message@voicemail_ivr" />
+			</phrases>
+			<keys>
+				<key dtmf="1" action="skip_intro" variable="VM-Key-Main-Listen-File" />
+				<key dtmf="6" action="next_msg" variable="VM-Key-Main-Next-Msg" />
+				<key dtmf="4" action="prev_msg" />
+				<key dtmf="7" action="delete_msg" variable="VM-Key-Main-Delete-File" /> <!-- Same key for undelete if it already deleted -->
+				<key dtmf="8" action="menu:std_forward" variable="VM-Key-Main-Forward" />
+				<key dtmf="3" action="save_msg" variable="VM-Key-Main-Save-File" />
+				<key dtmf="2" action="callback" variable="VM-Key-Main-Callback" />
+				<key dtmf="5" action="menu:std_preference" />
+				<key dtmf="#" action="return" /> <!-- TODO Might Conflict with future fast-forward -->
+			</keys>
+			</menu>
+
+			<menu name="std_preference">
+			<phrases>
+				<phrase name="menu_options" value="config_menu@voicemail_ivr" />
+			</phrases>
+			<keys>
+				<key dtmf="1" action="menu:std_record_greeting_with_slot" variable="VM-Key-Record-Greeting" />
+				<key dtmf="2" action="menu:std_select_greeting_slot" variable="VM-Key-Choose-Greeting" />
+				<key dtmf="3" action="menu:std_record_name" variable="VM-Key-Record-Name" />
+				<key dtmf="6" action="menu:std_set_password" variable="VM-Key-Change-Password" />
+				<key dtmf="#" action="return" variable="VM-Key-Main-Menu" />
+			</keys>
+			</menu>
+
+			<menu name="std_record_greeting">
+			<phrases>
+				<phrase name="instructions" value="record_greeting@voicemail_ivr" />
+				<phrase name="play_recording" value="play_recording@voicemail_ivr" />
+				<phrase name="menu_options" value="record_file_check@voicemail_ivr" />
+			</phrases>
+			<keys>
+				<key dtmf="1" action="listen" variable="VM-Key-Listen-File" />
+				<key dtmf="3" action="save" variable="VM-Key-Save-File" />
+				<key dtmf="4" action="rerecord" variable="VM-Key-ReRecord-File" />
+				<key dtmf="#" action="skip_instruction" />
+			</keys>
+			</menu>
+
+
+			<menu name="std_record_name">
+			<phrases>
+				<phrase name="instructions" value="record_name@voicemail_ivr" />
+				<phrase name="play_recording" value="play_recording@voicemail_ivr" />
+				<phrase name="menu_options" value="record_file_check@voicemail_ivr" />
+			</phrases>
+			<keys>
+				<key dtmf="1" action="listen" variable="VM-Key-Listen-File" />
+				<key dtmf="3" action="save" variable="VM-Key-Save-File" />
+				<key dtmf="4" action="rerecord" variable="VM-Key-ReRecord-File" />
+				<key dtmf="#" action="skip_instruction" />
+			</keys>
+			</menu>
+
+			<menu name="std_record_message">
+			<phrases>
+				<phrase name="instructions" value="record_message@voicemail_ivr" />
+				<phrase name="play_recording" value="play_recording@voicemail_ivr" />
+				<phrase name="menu_options" value="record_file_check@voicemail_ivr" />
+			</phrases>
+			<keys>
+				<key dtmf="1" action="listen" variable="VM-Key-Listen-File" />
+				<key dtmf="3" action="save" variable="VM-Key-Save-File" />
+				<key dtmf="4" action="rerecord" variable="VM-Key-ReRecord-File" />
+				<key dtmf="#" action="skip_instruction" />
+			</keys>
+			</menu>			
+
+			<menu name="std_forward_ask_prepend">
+			<phrases>
+				<phrase name="menu_options" value="forward_ask_prepend@voicemail_ivr" />
+			</phrases>
+			<keys>
+				<key dtmf="1" action="prepend" variable="VM-Key-Prepend" />
+				<key dtmf="8" action="forward" variable="VM-Key-Forward" />
+				<key dtmf="#" action="return" variable="VM-Key-Return" />
+			</keys>
+			</menu>
+
+			<menu name="std_forward_ask_extension">
+			<phrases>
+				<phrase name="instructions" value="forward_ask_extension@voicemail_ivr" />
+				<phrase name="ack" value="ack@voicemail_ivr" />
+				<phrase name="invalid_extension" value="invalid_extension@voicemail_ivr" />
+			</phrases>
+			<keys>
+				<key dtmf="#" action="ivrengine:terminate_entry" variable="VM-Key-Terminator" />
+			</keys>
+			</menu>
+
+			<menu name="std_select_greeting_slot">
+			<phrases>
+				<phrase name="instructions" value="choose_greeting@voicemail_ivr" />
+				<phrase name="invalid_slot" value="choose_greeting_fail@voicemail_ivr" />
+				<phrase name="selected_slot" value="greeting_selected@voicemail_ivr" />
+			</phrases>
+			<keys>
+			</keys>
+			</menu>
+
+			<menu name="std_record_greeting_with_slot">
+			<phrases>
+				<phrase name="instructions" value="choose_greeting@voicemail_ivr" />
+			</phrases>
+			<keys>
+			</keys>
+			</menu>
+
+			<menu name="std_set_password">
+			<phrases>
+				<phrase name="instructions" value="enter_pass@voicemail_ivr" />
+			</phrases>
+			<keys>
+				<key dtmf="#" action="ivrengine:terminate_entry" variable="VM-Key-Terminator" />
+			</keys>
+			</menu>
+		</menus>
+	</profile>
+</profiles>
+</configuration>
+
+

src/mod/applications/mod_voicemail_ivr/config.c

@@ -93,6 +93,11 @@ void menu_init(vmivr_profile_t *profile, vmivr_menu_t *menu) {
 			}
 		}
 	}
+
+	if (!menu->phrase_params) {
+		switch_event_create(&menu->phrase_params, SWITCH_EVENT_REQUEST_PARAMS);
+	}
+
 end:
 	if (xml)
 		switch_xml_free(xml);

src/mod/asr_tts/mod_cepstral/conf/cepstral.conf.xml

@@ -0,0 +1,12 @@
+<configuration name="cepstral.conf" description="Cepstral TTS configuration">
+  <settings>
+    <!--
+        Possible encodings:
+        * utf-8
+        * us-ascii
+        * iso8859-1 (default)
+        * iso8859-15
+    -->
+    <param name="encoding" value="utf-8"/>
+  </settings>
+</configuration>

src/mod/asr_tts/mod_flite/Makefile

@@ -1,4 +1,4 @@
-FLITE=flite-1.5.1
+FLITE=flite-1.5.4
 
 FLITE_DIR=$(switch_srcdir)/libs/$(FLITE)-current
 FLITE_BUILDDIR=$(switch_builddir)/libs/$(FLITE)-current
@@ -6,6 +6,7 @@ FLITE_BUILDDIR=$(switch_builddir)/libs/$(FLITE)-current
 FLITE_LIBDIR=$(FLITE_BUILDDIR)/build/libs
 FLITE_A=$(FLITE_LIBDIR)/libflite_cmu_us_awb.a \
 	$(FLITE_LIBDIR)/libflite_cmu_us_kal.a \
+	$(FLITE_LIBDIR)/libflite_cmu_us_kal16.a \
 	$(FLITE_LIBDIR)/libflite_cmu_us_rms.a \
 	$(FLITE_LIBDIR)/libflite_cmu_us_slt.a \
 	$(FLITE_LIBDIR)/libflite_cmulex.a $(FLITE_LIBDIR)/libflite_usenglish.a $(FLITE_LIBDIR)/libflite.a -lm

src/mod/asr_tts/mod_flite/mod_flite.2008.vcproj

@@ -42,7 +42,7 @@
 			/>
 			<Tool
 				Name="VCCLCompilerTool"
-				AdditionalIncludeDirectories="../../../../libs/flite-1.3.99/include"
+				AdditionalIncludeDirectories="../../../../libs/flite-1.5.4/include"
 				PreprocessorDefinitions="APT_LIB_EXPORT"
 				UsePrecompiledHeader="0"
 			/>
@@ -108,7 +108,7 @@
 			/>
 			<Tool
 				Name="VCCLCompilerTool"
-				AdditionalIncludeDirectories="../../../../libs/flite-1.3.99/include"
+				AdditionalIncludeDirectories="../../../../libs/flite-1.5.4/include"
 				PreprocessorDefinitions="APT_LIB_EXPORT"
 				UsePrecompiledHeader="0"
 			/>
@@ -175,7 +175,7 @@
 			/>
 			<Tool
 				Name="VCCLCompilerTool"
-				AdditionalIncludeDirectories="../../../../libs/flite-1.3.99/include"
+				AdditionalIncludeDirectories="../../../../libs/flite-1.5.4/include"
 				PreprocessorDefinitions="APT_LIB_EXPORT"
 				UsePrecompiledHeader="0"
 			/>
@@ -241,7 +241,7 @@
 			/>
 			<Tool
 				Name="VCCLCompilerTool"
-				AdditionalIncludeDirectories="../../../../libs/flite-1.3.99/include"
+				AdditionalIncludeDirectories="../../../../libs/flite-1.5.4/include"
 				PreprocessorDefinitions="APT_LIB_EXPORT"
 				UsePrecompiledHeader="0"
 			/>
@@ -308,7 +308,7 @@
 			/>
 			<Tool
 				Name="VCCLCompilerTool"
-				AdditionalIncludeDirectories="../../../../libs/flite-1.3.99/include"
+				AdditionalIncludeDirectories="../../../../libs/flite-1.5.4/include"
 				UsePrecompiledHeader="0"
 			/>
 			<Tool
@@ -375,7 +375,7 @@
 			/>
 			<Tool
 				Name="VCCLCompilerTool"
-				AdditionalIncludeDirectories="../../../../libs/flite-1.3.99/include"
+				AdditionalIncludeDirectories="../../../../libs/flite-1.5.4/include"
 				UsePrecompiledHeader="0"
 			/>
 			<Tool
@@ -440,7 +440,7 @@
 			/>
 			<Tool
 				Name="VCCLCompilerTool"
-				AdditionalIncludeDirectories="../../../../libs/flite-1.3.99/include"
+				AdditionalIncludeDirectories="../../../../libs/flite-1.5.4/include"
 				UsePrecompiledHeader="0"
 			/>
 			<Tool
@@ -507,7 +507,7 @@
 			/>
 			<Tool
 				Name="VCCLCompilerTool"
-				AdditionalIncludeDirectories="../../../../libs/flite-1.3.99/include"
+				AdditionalIncludeDirectories="../../../../libs/flite-1.5.4/include"
 				UsePrecompiledHeader="0"
 			/>
 			<Tool

src/mod/asr_tts/mod_flite/mod_flite.2010.vcxproj

@@ -120,7 +120,7 @@
   </PropertyGroup>
   <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">
     <ClCompile>
-      <AdditionalIncludeDirectories>../../../../libs/flite-1.3.99/include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
+      <AdditionalIncludeDirectories>../../../../libs/flite-1.5.4/include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
       <PreprocessorDefinitions>APT_LIB_EXPORT;%(PreprocessorDefinitions)</PreprocessorDefinitions>
       <PrecompiledHeader>
       </PrecompiledHeader>
@@ -137,7 +137,7 @@
       <TargetEnvironment>X64</TargetEnvironment>
     </Midl>
     <ClCompile>
-      <AdditionalIncludeDirectories>../../../../libs/flite-1.3.99/include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
+      <AdditionalIncludeDirectories>../../../../libs/flite-1.5.4/include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
       <PreprocessorDefinitions>APT_LIB_EXPORT;%(PreprocessorDefinitions)</PreprocessorDefinitions>
       <PrecompiledHeader>
       </PrecompiledHeader>
@@ -152,7 +152,7 @@
   </ItemDefinitionGroup>
   <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">
     <ClCompile>
-      <AdditionalIncludeDirectories>../../../../libs/flite-1.3.99/include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
+      <AdditionalIncludeDirectories>../../../../libs/flite-1.5.4/include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
       <PreprocessorDefinitions>APT_LIB_EXPORT;%(PreprocessorDefinitions)</PreprocessorDefinitions>
       <PrecompiledHeader>
       </PrecompiledHeader>
@@ -170,7 +170,7 @@
       <TargetEnvironment>X64</TargetEnvironment>
     </Midl>
     <ClCompile>
-      <AdditionalIncludeDirectories>../../../../libs/flite-1.3.99/include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
+      <AdditionalIncludeDirectories>../../../../libs/flite-1.5.4/include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
       <PreprocessorDefinitions>APT_LIB_EXPORT;%(PreprocessorDefinitions)</PreprocessorDefinitions>
       <PrecompiledHeader>
       </PrecompiledHeader>
@@ -186,7 +186,7 @@
   </ItemDefinitionGroup>
   <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Debug Static|Win32'">
     <ClCompile>
-      <AdditionalIncludeDirectories>../../../../libs/flite-1.3.99/include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
+      <AdditionalIncludeDirectories>../../../../libs/flite-1.5.4/include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
       <PrecompiledHeader>
       </PrecompiledHeader>
     </ClCompile>
@@ -203,7 +203,7 @@
       <TargetEnvironment>X64</TargetEnvironment>
     </Midl>
     <ClCompile>
-      <AdditionalIncludeDirectories>../../../../libs/flite-1.3.99/include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
+      <AdditionalIncludeDirectories>../../../../libs/flite-1.5.4/include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
       <PrecompiledHeader>
       </PrecompiledHeader>
     </ClCompile>
@@ -218,7 +218,7 @@
   </ItemDefinitionGroup>
   <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Release Static|Win32'">
     <ClCompile>
-      <AdditionalIncludeDirectories>../../../../libs/flite-1.3.99/include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
+      <AdditionalIncludeDirectories>../../../../libs/flite-1.5.4/include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
       <PrecompiledHeader>
       </PrecompiledHeader>
     </ClCompile>
@@ -235,7 +235,7 @@
       <TargetEnvironment>X64</TargetEnvironment>
     </Midl>
     <ClCompile>
-      <AdditionalIncludeDirectories>../../../../libs/flite-1.3.99/include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
+      <AdditionalIncludeDirectories>../../../../libs/flite-1.5.4/include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
       <PrecompiledHeader>
       </PrecompiledHeader>
     </ClCompile>
@@ -258,7 +258,6 @@
     </ProjectReference>
     <ProjectReference Include="..\..\..\..\libs\win32\flite\flite.2010.vcxproj">
       <Project>{0ad1177e-1fd8-4643-9391-431467a11084}</Project>
-      <ReferenceOutputAssembly>false</ReferenceOutputAssembly>
     </ProjectReference>
     <ProjectReference Include="..\..\..\..\w32\Library\FreeSwitchCore.2010.vcxproj">
       <Project>{202d7a4e-760d-4d0e-afa1-d7459ced30ff}</Project>

src/mod/asr_tts/mod_flite/mod_flite.c

@@ -44,6 +44,10 @@ void unregister_cmu_us_rms(cst_voice * v);
 cst_voice *register_cmu_us_slt(void);
 void unregister_cmu_us_slt(cst_voice * v);
 
+cst_voice *register_cmu_us_kal16(void);
+void unregister_cmu_us_kal16(cst_voice * v);
+
+
 SWITCH_MODULE_LOAD_FUNCTION(mod_flite_load);
 SWITCH_MODULE_SHUTDOWN_FUNCTION(mod_flite_shutdown);
 SWITCH_MODULE_DEFINITION(mod_flite, mod_flite_load, mod_flite_shutdown, NULL);
@@ -53,6 +57,7 @@ static struct {
 	cst_voice *kal;
 	cst_voice *rms;
 	cst_voice *slt;
+	cst_voice *kal16;
 } globals;
 
 struct flite_data {
@@ -75,13 +80,18 @@ static switch_status_t flite_speech_open(switch_speech_handle_t *sh, const char
 	if (!strcasecmp(voice_name, "awb")) {
 		flite->v = globals.awb;
 	} else if (!strcasecmp(voice_name, "kal")) {
-		flite->v = globals.kal;
+/*  "kal" is 8kHz and the native rate is set to 16kHz
+ *  so kal talks a little bit too fast ...
+ *  for now: "symlink" kal to kal16
+ */		flite->v = globals.kal16;
 	} else if (!strcasecmp(voice_name, "rms")) {
 		flite->v = globals.rms;
 	} else if (!strcasecmp(voice_name, "slt")) {
 		flite->v = globals.slt;
+	} else if (!strcasecmp(voice_name, "kal16")) {
+		flite->v = globals.kal16;
 	} else {
-		switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Valid voice names are awb, kal, rms or slt.\n");
+		switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Valid voice names are awb, rms, slt or kal.\n");
 	}
 
 	if (flite->v) {
@@ -179,6 +189,7 @@ SWITCH_MODULE_LOAD_FUNCTION(mod_flite_load)
 	globals.kal = register_cmu_us_kal();
 	globals.rms = register_cmu_us_rms();
 	globals.slt = register_cmu_us_slt();
+	globals.kal16 = register_cmu_us_kal16();
 
 	/* connect my internal structure to the blank pointer passed to me */
 	*module_interface = switch_loadable_module_create_module_interface(pool, modname);
@@ -203,6 +214,7 @@ SWITCH_MODULE_SHUTDOWN_FUNCTION(mod_flite_shutdown)
 	unregister_cmu_us_kal(globals.kal);
 	unregister_cmu_us_rms(globals.rms);
 	unregister_cmu_us_slt(globals.slt);
+	unregister_cmu_us_kal16(globals.kal16);
 
 	return SWITCH_STATUS_UNLOAD;
 }

src/mod/codecs/mod_speex/mod_speex.c

@@ -1,6 +1,6 @@
 /* 
  * FreeSWITCH Modular Media Switching Software Library / Soft-Switch Application
- * Copyright (C) 2005-2011, Anthony Minessale II <anthm@freeswitch.org>
+ * Copyright (C) 2005-2012, Anthony Minessale II <anthm@freeswitch.org>
  *
  * Version: MPL 1.1
  *
@@ -24,6 +24,7 @@
  * Contributor(s):
  * 
  * Anthony Minessale II <anthm@freeswitch.org>
+ * Chris Rienzo <chris@rienzo.net>
  *
  *
  * mod_speex.c -- Speex Codec Module
@@ -279,10 +280,11 @@ static switch_status_t switch_speex_encode(switch_codec_t *codec,
 	if (is_speech) {
 		switch_clear_flag(context, SWITCH_CODEC_FLAG_SILENCE);
 		*flag |= SWITCH_CODEC_FLAG_SILENCE_STOP;
+		*flag &= ~SFF_CNG;
 	} else {
 		if (switch_test_flag(context, SWITCH_CODEC_FLAG_SILENCE)) {
 			*encoded_data_len = 0;
-			*flag |= SWITCH_CODEC_FLAG_SILENCE;
+			*flag |= SWITCH_CODEC_FLAG_SILENCE | SFF_CNG;
 			return SWITCH_STATUS_SUCCESS;
 		}
 
@@ -353,6 +355,126 @@ static switch_status_t switch_speex_destroy(switch_codec_t *codec)
 	return SWITCH_STATUS_SUCCESS;
 }
 
+/**
+ * read default settings from speex.conf
+ */
+static void load_configuration()
+{
+	switch_xml_t xml = NULL, cfg = NULL;
+
+	if ((xml = switch_xml_open_cfg("speex.conf", &cfg, NULL))) {
+		switch_xml_t x_lists;
+		if ((x_lists = switch_xml_child(cfg, "settings"))) {
+			const char *settings_name = switch_xml_attr(x_lists, "name");
+			switch_xml_t x_list;
+			if (zstr(settings_name)) {
+				settings_name = "";
+			}
+			for (x_list = switch_xml_child(x_lists, "param"); x_list; x_list = x_list->next) {
+				const char *name = switch_xml_attr(x_list, "name");
+				const char *value = switch_xml_attr(x_list, "value");
+				if (zstr(name)) {
+					continue;
+				}
+
+				if (zstr(value)) {
+					continue;
+				}
+
+				switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "%s %s = %s\n", settings_name, name, value);
+
+				if (!strcasecmp("quality", name)) {
+					/* compression quality, integer 0-10 */
+					int tmp = atoi(value);
+					if (switch_is_number(value) && tmp >= 0 && tmp <= 10) {
+						default_codec_settings.quality = tmp;
+					} else {
+						switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "ignoring invalid quality value: %s\n", value);
+					}
+				} else if (!strcasecmp("complexity", name)) {
+					/* compression complexity, integer 1-10 */
+					int tmp = atoi(value);
+					if (switch_is_number(value) && tmp >= 1 && tmp <= 10) {
+						default_codec_settings.complexity = tmp;
+					} else {
+						switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "ignoring invalid complexity value: %s\n", value);
+					}
+				} else if (!strcasecmp("enhancement", name)) {
+					/* enable perceptual enhancement, boolean */
+					default_codec_settings.enhancement = switch_true(value);
+				} else if (!strcasecmp("vad", name)) {
+					/* enable voice activity detection, boolean */
+					default_codec_settings.vad = switch_true(value);
+				} else if (!strcasecmp("vbr", name)) {
+					/* enable variable bit rate, boolean */
+					default_codec_settings.vbr = switch_true(value);
+				} else if (!strcasecmp("vbr-quality", name)) {
+					/* variable bit rate quality, float 0-10 */
+					float tmp = atof(value);
+					if (switch_is_number(value) && tmp >= 0 && tmp <= 10) {
+						default_codec_settings.vbr_quality = tmp;
+					} else {
+						switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "ignoring invalid vbr-quality value: %s\n", value);
+					}
+				} else if (!strcasecmp("abr", name)) {
+					/* average bit rate, integer bits per sec */
+					int tmp = atoi(value);
+					if (switch_is_number(value) && tmp >= 0) {
+						default_codec_settings.abr = tmp;
+					} else {
+						switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "ignoring invalid abr value: %s\n", value);
+					}
+				} else if (!strcasecmp("dtx", name)) {
+					/* discontinuous transmit, boolean */
+					default_codec_settings.dtx = switch_true(value);
+				} else if (!strcasecmp("preproc", name)) {
+					/* enable preprocessor, boolean */
+					default_codec_settings.preproc = switch_true(value);
+				} else if (!strcasecmp("pp-vad", name)) {
+					/* enable preprocessor VAD, boolean */
+					default_codec_settings.pp_vad = switch_true(value);
+				} else if (!strcasecmp("pp-agc", name)) {
+					/* enable preprocessor automatic gain control, boolean */
+					default_codec_settings.pp_agc = switch_true(value);
+				} else if (!strcasecmp("pp-agc-level", name)) {
+					/* agc level, float */
+					float tmp = atof(value);
+					if (switch_is_number(value) && tmp >= 0.0f) {
+						default_codec_settings.pp_agc_level = tmp;
+					} else {
+						switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "ignoring invalid pp-agc-level value: %s\n", value);
+					}
+				} else if (!strcasecmp("pp-denoise", name)) {
+					/* enable preprocessor denoiser, boolean */
+					default_codec_settings.pp_denoise = switch_true(value);
+				} else if (!strcasecmp("pp-dereverb", name)) {
+					/* enable preprocessor reverberation removal, boolean */
+					default_codec_settings.pp_dereverb = switch_true(value);
+				} else if (!strcasecmp("pp-dereverb-decay", name)) {
+					/* reverberation removal decay, float */
+					float tmp = atof(value);
+					if (switch_is_number(value) && tmp >= 0.0f) {
+						default_codec_settings.pp_dereverb_decay = tmp;
+					} else {
+						switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "ignoring invalid pp-dereverb-decay value: %s\n", value);
+					}
+				} else if (!strcasecmp("pp-dereverb-level", name)) {
+					/* reverberation removal level, float */
+					float tmp = atof(value);
+					if (switch_is_number(value) && tmp >= 0.0f) {
+						default_codec_settings.pp_dereverb_level = tmp;
+					} else {
+						switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "ignoring invalid pp-dereverb-level value: %s\n", value);
+					}
+				} else {
+					switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "ignoring invalid unknown param: %s = %s\n", name, value);
+				}
+			}
+		}
+		switch_xml_free(xml);
+	}
+}
+
 SWITCH_MODULE_LOAD_FUNCTION(mod_speex_load)
 {
 	switch_codec_interface_t *codec_interface;
@@ -361,6 +483,9 @@ SWITCH_MODULE_LOAD_FUNCTION(mod_speex_load)
 	int bps[4] = { 0, 24600, 42200, 44000 };
 	/* connect my internal structure to the blank pointer passed to me */
 	*module_interface = switch_loadable_module_create_module_interface(pool, modname);
+
+	load_configuration();
+
 	SWITCH_ADD_CODEC(codec_interface, "Speex");
 	codec_interface->parse_fmtp = switch_speex_fmtp_parse;
 	for (counta = 1; counta <= 3; counta++) {

src/mod/dialplans/mod_dialplan_asterisk/conf/extensions.conf

@@ -0,0 +1,21 @@
+[default]
+
+; Things you're used to....
+exten => music,n,Dial(SIP/1234@conference.freeswitch.org|120)
+
+exten => _1XXXXX,n,set(cool=${EXTEN})
+exten => _1XXXXX,n,set(myvar=true)
+exten => _1XXXXX,n,Goto(default|music)
+exten => 2137991400/1000,n,Goto(default|music)
+
+
+; Some new magic you can do....
+exten => ~^(18(0{2}|8{2}|7{2}|6{2})\d{7})$,n,enum($1)
+exten => ~^(18(0{2}|8{2}|7{2}|6{2})\d{7})$,n,bridge(${enum_auto_route})
+
+; instead of exten, put anything about the call you would rather match on.
+; either the names of a field in caller_profile or a string of variables to expand.
+caller_id_number => 2137991400,n,Goto(default|music) 
+${sip_from_user} => bill,n,Goto(default|music)
+
+

src/mod/dialplans/mod_dialplan_xml/mod_dialplan_xml.c

@@ -143,6 +143,7 @@ static int parse_exten(switch_core_session_t *session, switch_caller_profile_t *
 							  "Dialplan: %s Date/Time Match (PASS) [%s] break=%s\n",
 							  switch_channel_get_name(channel), exten_name, do_break_a ? do_break_a : "on-false");
 			anti_action = SWITCH_FALSE;
+			proceed = 1;
 		} else if (time_match == 0) {
 			switch_log_printf(SWITCH_CHANNEL_SESSION_LOG_CLEAN(session), SWITCH_LOG_DEBUG,
 							  "Dialplan: %s Date/TimeMatch (FAIL) [%s] break=%s\n",
@@ -189,7 +190,7 @@ static int parse_exten(switch_core_session_t *session, switch_caller_profile_t *
 				total++;
 				
 				field = (char *) switch_xml_attr(xregex, "field");
-
+				
 				if (field) {
 					if (strchr(field, '$')) {
 						if ((field_expanded = switch_channel_expand_variables(channel, field)) == field) {
@@ -222,9 +223,11 @@ static int parse_exten(switch_core_session_t *session, switch_caller_profile_t *
 					switch_log_printf(SWITCH_CHANNEL_SESSION_LOG_CLEAN(session), SWITCH_LOG_DEBUG,
 									  "Dialplan: %s Absolute Condition [%s] match=%s\n", switch_channel_get_name(channel), exten_name, all ? "all" : "any");
 					pass++;
+					proceed = 1;
 					if (!all && !xor) break;
 				} else if (time_match == 1) {
 					pass++;
+					proceed = 1;
 					if (!all && !xor) break;
 				}
 				
@@ -310,6 +313,7 @@ static int parse_exten(switch_core_session_t *session, switch_caller_profile_t *
 				switch_log_printf(SWITCH_CHANNEL_SESSION_LOG_CLEAN(session), SWITCH_LOG_DEBUG,
 								  "Dialplan: %s Absolute Condition [%s]\n", switch_channel_get_name(channel), exten_name);
 				anti_action = SWITCH_FALSE;
+				proceed = 1;
 			}
 
 		}

src/mod/endpoints/mod_alsa/conf/autoload_configs/alsa.conf.xml

@@ -0,0 +1,12 @@
+<configuration name="alsa.conf" description="Soundcard Endpoint">
+  <settings>
+    <!--Default dialplan and caller-id info -->
+    <param name="dialplan" value="XML"/>
+    <param name="cid-name" value="N800 Alsa"/>
+    <param name="cid-num" value="5555551212"/>
+
+    <!--audio sample rate and interval -->
+    <param name="sample-rate" value="8000"/>
+    <param name="codec-ms" value="20"/>
+  </settings>
+</configuration>

src/mod/endpoints/mod_gsmopen/.gitignore

@@ -1,4 +1,3 @@
 !/gsmlib/gsmlib-*/aclocal.m4
 !/gsmlib/gsmlib-*/configure
-!/gsmlib/gsmlib-1.10.tar.gz
-!/gsmlib/gsmlib_1.10-12ubuntu1.diff.gz
+!/gsmlib/gsmlib_1.10.orig.tar.gz

src/mod/endpoints/mod_h323/mod_h323.2010.vcxproj

@@ -68,7 +68,7 @@
   </PropertyGroup>
   <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">
     <ClCompile>
-      <AdditionalIncludeDirectories>%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
+      <AdditionalIncludeDirectories>..\..\..\..\..\ptlib\include;..\..\..\..\..\h323plus\include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
       <PreprocessorDefinitions>WIN32;_DEBUG;_WINDOWS;_USRDLL;MOD_EXPORTS;ALLOW_SMP_DANGERS;%(PreprocessorDefinitions)</PreprocessorDefinitions>
       <PrecompiledHeader>
       </PrecompiledHeader>
@@ -77,7 +77,7 @@
     <Link>
       <AdditionalOptions>/NODEFAULTLIB:LIMBCTD %(AdditionalOptions)</AdditionalOptions>
       <AdditionalDependencies>h323plusd.lib;ptlibsd.lib;%(AdditionalDependencies)</AdditionalDependencies>
-      <AdditionalLibraryDirectories>%(AdditionalLibraryDirectories)</AdditionalLibraryDirectories>
+      <AdditionalLibraryDirectories>..\..\..\..\..\ptlib\lib;..\..\..\..\..\h323plus\lib;%(AdditionalLibraryDirectories)</AdditionalLibraryDirectories>
       <RandomizedBaseAddress>false</RandomizedBaseAddress>
       <DataExecutionPrevention>
       </DataExecutionPrevention>
@@ -85,7 +85,7 @@
   </ItemDefinitionGroup>
   <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">
     <ClCompile>
-      <AdditionalIncludeDirectories>%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
+      <AdditionalIncludeDirectories>..\..\..\..\..\ptlib\include;..\..\..\..\..\h323plus\include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
       <PreprocessorDefinitions>WIN32;_DEBUG;_WINDOWS;_USRDLL;MOD_EXPORTS;ALLOW_SMP_DANGERS;%(PreprocessorDefinitions)</PreprocessorDefinitions>
       <PrecompiledHeader>
       </PrecompiledHeader>
@@ -94,7 +94,7 @@
     <Link>
       <AdditionalOptions>/NODEFAULTLIB:LIMBCTD %(AdditionalOptions)</AdditionalOptions>
       <AdditionalDependencies>h323plusd.lib;ptlibsd.lib;%(AdditionalDependencies)</AdditionalDependencies>
-      <AdditionalLibraryDirectories>%(AdditionalLibraryDirectories)</AdditionalLibraryDirectories>
+      <AdditionalLibraryDirectories>..\..\..\..\..\ptlib\lib\x64;..\..\..\..\..\h323plus\lib\x64;%(AdditionalLibraryDirectories)</AdditionalLibraryDirectories>
       <RandomizedBaseAddress>false</RandomizedBaseAddress>
       <DataExecutionPrevention>
       </DataExecutionPrevention>
@@ -102,7 +102,7 @@
   </ItemDefinitionGroup>
   <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">
     <ClCompile>
-      <AdditionalIncludeDirectories>%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
+      <AdditionalIncludeDirectories>..\..\..\..\..\ptlib\include;..\..\..\..\..\h323plus\include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
       <PreprocessorDefinitions>WIN32;_WINDOWS;_USRDLL;MOD_EXPORTS;%(PreprocessorDefinitions)</PreprocessorDefinitions>
       <PrecompiledHeader>
       </PrecompiledHeader>
@@ -110,7 +110,7 @@
     </ClCompile>
     <Link>
       <AdditionalDependencies>h323plus.lib;ptlibs.lib;%(AdditionalDependencies)</AdditionalDependencies>
-      <AdditionalLibraryDirectories>%(AdditionalLibraryDirectories)</AdditionalLibraryDirectories>
+      <AdditionalLibraryDirectories>..\..\..\..\..\ptlib\lib;..\..\..\..\..\h323plus\lib;%(AdditionalLibraryDirectories)</AdditionalLibraryDirectories>
       <GenerateDebugInformation>false</GenerateDebugInformation>
       <RandomizedBaseAddress>false</RandomizedBaseAddress>
       <DataExecutionPrevention>
@@ -119,7 +119,7 @@
   </ItemDefinitionGroup>
   <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Release|x64'">
     <ClCompile>
-      <AdditionalIncludeDirectories>%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
+      <AdditionalIncludeDirectories>..\..\..\..\..\ptlib\include;..\..\..\..\..\h323plus\include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
       <PreprocessorDefinitions>WIN32;_WINDOWS;_USRDLL;MOD_EXPORTS;%(PreprocessorDefinitions)</PreprocessorDefinitions>
       <PrecompiledHeader>
       </PrecompiledHeader>
@@ -127,11 +127,12 @@
     </ClCompile>
     <Link>
       <AdditionalDependencies>h323plus.lib;ptlibs.lib;%(AdditionalDependencies)</AdditionalDependencies>
-      <AdditionalLibraryDirectories>%(AdditionalLibraryDirectories)</AdditionalLibraryDirectories>
+      <AdditionalLibraryDirectories>..\..\..\..\..\ptlib\lib\x64;..\..\..\..\..\h323plus\lib\x64;%(AdditionalLibraryDirectories)</AdditionalLibraryDirectories>
       <GenerateDebugInformation>false</GenerateDebugInformation>
       <RandomizedBaseAddress>false</RandomizedBaseAddress>
       <DataExecutionPrevention>
       </DataExecutionPrevention>
+      <PerUserRedirection>false</PerUserRedirection>
     </Link>
   </ItemDefinitionGroup>
   <ItemGroup>

src/mod/endpoints/mod_portaudio/conf/autoload_configs/portaudio.conf.xml

@@ -0,0 +1,218 @@
+<configuration name="portaudio.conf" description="Soundcard Endpoint">
+  <settings>
+    <!-- indev, outdev, ringdev: 
+	 partial case sensitive string match on something in the name 
+	 or the device number prefixed with # eg "#1" (or blank for default) -->
+
+    <!-- device to use for input -->
+    <param name="indev" value=""/>
+    <!-- device to use for output -->
+    <param name="outdev" value=""/>
+
+    <!--device to use for inbound ring -->
+    <!--<param name="ringdev" value=""/>-->
+    <!--File to play as the ring sound -->
+    <!--<param name="ring-file" value="/sounds/ring.wav"/>-->
+    <!--Number of seconds to pause between rings -->
+    <!--<param name="ring-interval" value="5"/>-->
+    <!--Enable or Disable dual_streams-->
+    <!--<param name="dual-streams" value="true"/>-->
+
+    <!--file to play when calls are on hold-->
+    <param name="hold-file" value="$${hold_music}"/>
+    <!--Timer to use for hold music (i'd leave this one commented)-->
+    <!--<param name="timer-name" value="soft"/>-->
+
+    <!--Default dialplan and caller-id info -->
+    <param name="dialplan" value="XML"/>
+    <param name="cid-name" value="$${outbound_caller_name}"/>
+    <param name="cid-num" value="$${outbound_caller_id}"/>
+
+    <!--audio sample rate and interval -->
+    <param name="sample-rate" value="48000"/>
+    <param name="codec-ms" value="20"/>
+  </settings>
+
+  <!-- 
+	mod_portaudio "streams"
+
+	The portaudio streams were introduced to support multiple devices and multiple channels in mod_portaudio.
+	For example, if you have a sound card that supports multiple channels or have multiple sound cards and you
+	want to use them at the same time, you can do it configuring streams and endpoints here.
+
+	A "stream" is just a logical container for some settings required by portaudio in order to stream audio and
+	define a friendly name for that configuration. Streams in itself do not do anything else than contain configs. 
+	Once you have your streams defined you can proceed to define "endpoints". Go to the "<endpoints>" section 
+	for more information on endpoints.
+
+	You can use the command "pa shstreams" (portaudio shared streams) to show the configured streams.
+  -->
+  <streams>
+  	<!--
+	In this example we define 2 streams, one for a usb audio device and another for the usual Mac defaults
+	The name="" attribute in the <stream> tag must uniquely identify the stream configuration and can be
+	later used when creating endpoints in the "instream" and "outstream" parameters of the endpoint.
+	-->
+
+	<!-- This sample "usb1" configuration was tested with a USB Griffin iMic device -->
+  	<stream name="usb1">
+		<!-- 
+			Which device to use for input in this stream 
+			The value for this parameter must be either in the form '#devno', 
+			for example '#2' for device number 2, or 'device-name', like 'iMic USB audio system'
+			The output of command "pa devlist" will show you device names and numbers as enumerated
+			by portaudio.
+		-->
+		<param name="indev" value="#2" />
+
+		<!-- 
+			Same as the indev but for output. In this case the device is capable of input and output
+			Some devices are capable of input only or output only (see the default example)
+	       	-->
+		<param name="outdev" value="#2" />
+
+		<!-- The sample rate to use for this stream -->
+		<param name="sample-rate" value="48000" />
+
+		<!-- 
+			Size of the packets in milliseconds. The smaller the number the less latency you'll have
+			The minimum value is 10ms 
+		-->
+		<param name="codec-ms" value="10" />
+
+		<!-- 
+			How many channels to open for this stream.
+			If you're device is stereo, you can choose 2 here. However, bear in mind that then
+			your left and right channels will be separated and when creating endpoints you will have
+			to either choose the left or right channel. This may or may not be what you want. This separation
+			means that you can have 2 separate FreeSWITCH calls, listening to one of them in your left channel
+			and the other in the right chanel.
+		-->
+		<param name="channels" value="2" />
+  	</stream>
+
+	<!-- This default stream was tested using the default Macbook Pro input/output devices -->
+  	<stream name="default">
+		<!-- The default system input device -->
+		<param name="indev" value="#0" />
+		<!-- The default system output device -->
+		<param name="outdev" value="#1" />
+		<!-- CD quality sampling rate ftw -->
+		<param name="sample-rate" value="48000" />
+		<!-- Low latency -->
+		<param name="codec-ms" value="10" />
+		<!-- Choosing 1 channel allows to hear in both left-right channel when using a headset -->
+		<param name="channels" value="1" />
+  	</stream>
+  </streams>
+
+  <!-- 
+	mod_portaudio "endpoints"
+
+	Endpoints is a way to define the input and output that a given portaudio channel will use.
+	There is a lot of flexibility. You can create endpoints which are "send-only", which means
+	audio will be read from FreeSWITCH and sent down to the provided stream, but no audio will
+	be read from that stream and only silence provided back to FreeSWITCH.
+
+		send-only endpoint:
+			(FS CORE) ->-> audio ->-> sound-card-x
+
+	You can also create a read-only endpoint.
+
+		read-only-endpoint:
+			(FS CORE) <-<- audio <-<- sound-card-x
+
+	And of course you can create a bidirectional endpoint:
+		bidirectional-endpoint:
+			(FS CORE) <-> audio <-> sound-card-x
+
+	You can also define a stream which uses only the left or only the right channel of a given device stream.
+	This means you can have 2 SIP calls connected to the same device haring one call in your left ear and
+	the other call to your right ear :-)
+
+	The name="parameter" of the endpoint allows you to use it in the FreeSWITCH dial plan to dial, ie:
+
+	<action application="bridge" data="portaudio/endpoint/usb1out-left" />
+
+	You can use the command "pa endpoints" to show the configured endpoints.
+  -->
+  <endpoints>
+
+	<!-- 
+		An endpoint is a handle name to refer to a configuration that determines where to read media from 
+		and write media to. The endpoint can use any input/output stream combination for that purpose as 
+		long as the streams match the sampling rate and codec-ms (see <streams> XML tag).
+		You can also omit the instream or the outstream parameter (but obviously not both).
+	-->
+
+	<!-- 
+		Configuration for a "default" bidirectional endpoint that uses the default stream defined previously in
+		the <streams> section.
+	 -->
+  	<endpoint name="default">
+		<!-- 
+			The instream, outstream is the name of the stream and channel to use. The stream
+			name is the same you configured in the <streams> section. This parameters follow 
+			the syntax <stream-name>:<channel index>. You can omit either the outstream
+			or the instream, but not both! The channel index is zero-based and must be consistent 
+			with the number of channels available for that stream (as configured in the <stream> section).
+			You cannot use index 1 if you chose channels=1 in the stream configuration.
+		-->
+		<param name="instream" value="default:0" />
+		<param name="outstream" value="default:0" />
+	</endpoint>
+
+	<!-- 
+		This endpoint uses the USB stream defined previously in the <streams> section and
+		is 'send-only' or 'output-only' and uses the channel index 0 (left channel in a stereo device) 
+	-->
+  	<endpoint name="usb1out-left">
+		<param name="outstream" value="usb1:0" />
+	</endpoint>
+
+	<!-- 
+		This endpoint uses the USB stream defined previously in the <streams> section and
+		is 'send-only' or 'output-only' and uses the channel index 1 (right channel in a stereo device) 
+	-->
+  	<endpoint name="usb1out-right">
+		<param name="outstream" value="usb1:1" />
+	</endpoint>
+
+	<!-- 
+		This endpoint uses the USB stream defined previously in the <streams> section and
+		is 'receive-only' or 'input-only' and uses the channel index 0 (left channel in a stereo device) 
+	-->
+  	<endpoint name="usb1in-left">
+		<param name="instream" value="usb1:0" />
+	</endpoint>
+
+	<!-- 
+		This endpoint uses the USB stream defined previously in the <streams> section and
+		is 'receive-only' or 'input-only' and uses the channel index 1 (right channel in a stereo device) 
+	-->
+  	<endpoint name="usb1in-right">
+		<param name="instream" value="usb1:1" />
+	</endpoint>
+
+	<!-- 
+		This endpoint uses the USB stream defined previously in the <streams> section and
+		is 'bidirectional' or 'send-receive' and uses the channel index 0 (left channel in a stereo device) 
+	-->
+  	<endpoint name="usb1-left">
+		<param name="instream" value="usb1:0" />
+		<param name="outstream" value="usb1:0" />
+	</endpoint>
+
+	<!-- 
+		This endpoint uses the USB stream defined previously in the <streams> section and
+		is 'bidirectional' or 'send-receive' and uses the channel index 1 (right channel in a stereo device) 
+	-->
+  	<endpoint name="usb1-right">
+		<param name="instream" value="usb1:1" />
+		<param name="outstream" value="usb1:1" />
+	</endpoint>
+
+  </endpoints>
+
+</configuration>
+

src/mod/endpoints/mod_rtmp/conf/autoload_configs/rtmp.conf.xml

@@ -0,0 +1,17 @@
+<configuration name="rtmp.conf" description="RTMP Endpoint">
+  <profiles>
+	<profile name="default">
+		<settings>
+			<param name="bind-address" value="0.0.0.0:1935" />
+			<param name="context" value="public" />
+			<param name="dialplan" value="XML" />
+			<!-- If this is set to true, no unauthenticated inbound calls will be allowed -->
+			<param name="auth-calls" value="true" />
+			<!-- How much time should the clients buffer the media stream (miliseconds) -->
+			<param name="buffer-len" value="50" />
+			<!-- Sets the maximum size of outbound RTMP chunks -->
+			<param name="chunksize" value="512" />
+		</settings>
+	</profile>
+  </profiles>
+</configuration>

src/mod/endpoints/mod_rtmp/mod_rtmp.c

@@ -843,7 +843,7 @@ switch_call_cause_t rtmp_session_create_call(rtmp_session_t *rsession, switch_co
 	switch_channel_t *channel;
 	const char *dialplan, *context;
 	
-	if (!(*newsession = switch_core_session_request(rtmp_globals.rtmp_endpoint_interface, SOF_NONE, SWITCH_CALL_DIRECTION_INBOUND, NULL))) {
+	if (!(*newsession = switch_core_session_request(rtmp_globals.rtmp_endpoint_interface, SWITCH_CALL_DIRECTION_INBOUND, SOF_NONE, NULL))) {
 		return SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER;
 	}
 

src/mod/endpoints/mod_rtmp/rtmp.c

@@ -886,6 +886,11 @@ switch_status_t rtmp_handle_data(rtmp_session_t *rsession)
 							if (rsession->tech_pvt) {
 								uint16_t len = state->origlen;
 								
+								if (!rsession->tech_pvt->readbuf) {
+									return SWITCH_STATUS_FALSE;
+								}
+
+								
 								switch_mutex_lock(rsession->tech_pvt->readbuf_mutex);
 								if (rsession->tech_pvt->maxlen && switch_buffer_inuse(rsession->tech_pvt->readbuf) > rsession->tech_pvt->maxlen * 40) {
 									rsession->tech_pvt->over_size++;

src/mod/endpoints/mod_skinny/conf/autoload_configs/skinny.conf.xml

@@ -0,0 +1,6 @@
+<configuration name="skinny.conf" description="Skinny Endpoints">
+  <profiles>
+    <X-PRE-PROCESS cmd="include" data="../skinny_profiles/*.xml"/>
+  </profiles>
+</configuration>
+

src/mod/endpoints/mod_skinny/conf/dialplan/skinny-patterns.xml

@@ -0,0 +1,30 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!--
+    NOTICE:
+
+    This context is used for skinny to match dialed number
+
+    The special applications:
+    - skinny-process tells skinny to process the call (route, set call forwarding, ...)
+    - skinny-drop tells skinny to drop the call
+    - skinny-wait tells skinny to wait 'data' seconds for more numbers before drop
+-->
+<!-- http://wiki.freeswitch.org/wiki/Mod_skinny -->
+<include>
+  <context name="skinny-patterns">
+    <!--
+    Wait 10 seconds for another digit by default
+    -->
+    <extension name="Default">
+      <condition>
+        <action application="skinny-wait" data="10"/>
+      </condition>
+    </extension>
+
+    <!--
+	You can place files in the skinny-patterns directory to get included.
+    -->
+    <X-PRE-PROCESS cmd="include" data="skinny-patterns/*.xml"/>
+    
+  </context>
+</include>

src/mod/endpoints/mod_skinny/conf/dialplan/skinny-patterns/20-Demo.xml

@@ -0,0 +1,8 @@
+<include>
+  <extension name="Demo">
+    <condition field="destination_number" expression="^(9[0-9]{3})$">
+      <action application="skinny-process"/>
+    </condition>
+  </extension>
+</include>
+

src/mod/endpoints/mod_skinny/conf/dialplan/skinny-patterns/20-Local_extension.xml

@@ -0,0 +1,13 @@
+<include>
+  <extension name="Local_Extension">
+    <condition field="destination_number" expression="^(10[01][0-9])$">
+      <action application="skinny-process"/>
+    </condition>
+  </extension>
+  <extension name="Local_Extension_Skinny">
+    <condition field="destination_number" expression="^(11[01][0-9])$">
+      <action application="skinny-process"/>
+    </condition>
+  </extension>
+</include>
+

src/mod/endpoints/mod_skinny/conf/dialplan/skinny-patterns/90-External.xml

@@ -0,0 +1,8 @@
+<include>
+  <extension name="External">
+    <condition field="destination_number" expression="^(0[0-9]{10,})$">
+      <action application="skinny-process"/>
+    </condition>
+  </extension>
+</include>
+

src/mod/endpoints/mod_skinny/conf/dialplan/skinny-patterns/99-Default_Drop.xml

@@ -0,0 +1,9 @@
+<include>
+  <!-- drop long dialed number -->
+  <extension name="Default_Drop">
+    <condition field="destination_number" expression="^(.{14,})$">
+      <action application="skinny-drop"/>
+    </condition>
+  </extension>
+</include>
+

src/mod/endpoints/mod_skinny/conf/directory/default/skinny-example.xml

@@ -0,0 +1,35 @@
+<include>
+ <user id="SEP001120AABBCC">
+   <params>
+    <!-- for devices requesting firmware via SCCP, like ATA186
+    <param name="skinny-firmware-version" value="ATA030101SCCP04"
+    <param name="skinny-soft-key-set-set" value="default"
+    -->
+    <param name="foo" value="bar"/>
+   </params>
+   <skinny>
+    <buttons>
+      <!--
+      position: 1..
+      type: one of line, speed-dial
+      label: button label
+      -->
+      <!--
+      value is the directory number (or user)
+      caller-name is shown to the calling party during call
+      -->
+      <button position="1" type="Line" label="Line 1" value="1101" caller-name="Calling as 1101"/>
+      <button position="3" type="Line" label="Shared Line 10" value="1110" caller-name="Calling as 1110"/>
+      <!--
+      value is the directory number to call
+      -->
+      <button position="5" type="SpeedDial" label="Call 1001" value="1001"/>
+      <!--
+      value is the URL
+      -->
+      <button position="6" type="ServiceUrl" label="Some URL" value="http://phone-xml.berbee.com/menu.xml"/>
+    </buttons>
+   </skinny>
+  </user>
+</include>
+

src/mod/endpoints/mod_skinny/conf/skinny_profiles/internal.xml

@@ -0,0 +1,37 @@
+<profile name="internal">
+  <settings>
+    <param name="domain" value="$${domain}"/>
+    <param name="ip" value="$${local_ip_v4}"/>
+    <param name="port" value="2000"/>
+    <param name="patterns-dialplan" value="XML"/>
+    <param name="patterns-context" value="skinny-patterns"/>
+    <param name="dialplan" value="XML"/>
+    <param name="context" value="default"/>
+    <param name="keep-alive" value="60"/>
+    <param name="date-format" value="D/M/Y"/>
+    <param name="odbc-dsn" value=""/>
+    <param name="debug" value="4"/>
+    <param name="auto-restart" value="true"/>
+  </settings>
+  <soft-key-set-sets>
+    <soft-key-set-set name="default">
+      <soft-key-set name="KeySetOnHook" value="SoftkeyNewcall,SoftkeyRedial"/>
+      <soft-key-set name="KeySetConnected" value="SoftkeyEndcall,SoftkeyHold,SoftkeyNewcall,SoftkeyTransfer"/>
+      <soft-key-set name="KeySetOnHold" value="SoftkeyNewcall,SoftkeyResume,SoftkeyEndcall"/>
+      <soft-key-set name="KeySetRingIn" value="SoftkeyAnswer,SoftkeyEndcall,SoftkeyNewcall"/>
+      <soft-key-set name="KeySetOffHook" value=",SoftkeyRedial,SoftkeyEndcall"/>
+      <soft-key-set name="KeySetConnectedWithTransfer" value="SoftkeyEndcall,SoftkeyHold,SoftkeyNewcall,SoftkeyTransfer"/>
+      <soft-key-set name="KeySetDigitsAfterDialingFirstDigit" value="SoftkeyBackspace,,SoftkeyEndcall"/>
+      <!-- <soft-key-set name="KeySetConnectedWithConference" value=""/> -->
+      <soft-key-set name="KeySetRingOut" value=",,SoftkeyEndcall,SoftkeyTransfer"/>
+      <soft-key-set name="KeySetOffHookWithFeatures" value=",SoftkeyRedial,SoftkeyEndcall"/>
+      <soft-key-set name="KeySetInUseHint" value="SoftkeyNewcall,SoftkeyRedial"/>
+    </soft-key-set-set>
+  </soft-key-set-sets>
+  <device-types>
+    <device-type id="Cisco ATA 186">
+        <param name="firmware-version" value="ATA030101SCCP04"/>
+    </device-type>
+  </device-types>
+</profile>
+

src/mod/endpoints/mod_skypopen/README

@@ -6,4 +6,3 @@ tricks and tweaks, possible problems at:
 http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk
 
 < gmaruzz at gmail dot com >
-

src/mod/endpoints/mod_skypopen/mod_skypopen.2010.vcxproj

@@ -5,10 +5,18 @@
       <Configuration>Debug</Configuration>
       <Platform>Win32</Platform>
     </ProjectConfiguration>
+    <ProjectConfiguration Include="Debug|x64">
+      <Configuration>Debug</Configuration>
+      <Platform>x64</Platform>
+    </ProjectConfiguration>
     <ProjectConfiguration Include="Release|Win32">
       <Configuration>Release</Configuration>
       <Platform>Win32</Platform>
     </ProjectConfiguration>
+    <ProjectConfiguration Include="Release|x64">
+      <Configuration>Release</Configuration>
+      <Platform>x64</Platform>
+    </ProjectConfiguration>
   </ItemGroup>
   <PropertyGroup Label="Globals">
     <ProjectName>mod_skypopen</ProjectName>
@@ -21,10 +29,18 @@
     <ConfigurationType>DynamicLibrary</ConfigurationType>
     <CharacterSet>MultiByte</CharacterSet>
   </PropertyGroup>
+  <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Release|x64'" Label="Configuration">
+    <ConfigurationType>DynamicLibrary</ConfigurationType>
+    <CharacterSet>MultiByte</CharacterSet>
+  </PropertyGroup>
   <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'" Label="Configuration">
     <ConfigurationType>DynamicLibrary</ConfigurationType>
     <CharacterSet>MultiByte</CharacterSet>
   </PropertyGroup>
+  <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='Debug|x64'" Label="Configuration">
+    <ConfigurationType>DynamicLibrary</ConfigurationType>
+    <CharacterSet>MultiByte</CharacterSet>
+  </PropertyGroup>
   <Import Project="$(VCTargetsPath)\Microsoft.Cpp.props" />
   <ImportGroup Label="ExtensionSettings">
   </ImportGroup>
@@ -32,10 +48,18 @@
     <Import Project="$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props" Condition="exists('$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props')" Label="LocalAppDataPlatform" />
     <Import Project="..\..\..\..\w32\module_release.props" />
   </ImportGroup>
+  <ImportGroup Condition="'$(Configuration)|$(Platform)'=='Release|x64'" Label="PropertySheets">
+    <Import Project="$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props" Condition="exists('$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props')" Label="LocalAppDataPlatform" />
+    <Import Project="..\..\..\..\w32\module_release.props" />
+  </ImportGroup>
   <ImportGroup Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'" Label="PropertySheets">
     <Import Project="$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props" Condition="exists('$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props')" Label="LocalAppDataPlatform" />
     <Import Project="..\..\..\..\w32\module_debug.props" />
   </ImportGroup>
+  <ImportGroup Condition="'$(Configuration)|$(Platform)'=='Debug|x64'" Label="PropertySheets">
+    <Import Project="$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props" Condition="exists('$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props')" Label="LocalAppDataPlatform" />
+    <Import Project="..\..\..\..\w32\module_debug.props" />
+  </ImportGroup>
   <PropertyGroup Label="UserMacros" />
   <PropertyGroup>
     <_ProjectFileVersion>10.0.30319.1</_ProjectFileVersion>
@@ -55,6 +79,21 @@
       </DataExecutionPrevention>
     </Link>
   </ItemDefinitionGroup>
+  <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">
+    <ClCompile>
+      <AdditionalIncludeDirectories>%(RootDir)%(Directory)..\..\..\..\libs\spandsp\src\msvc;%(RootDir)%(Directory)..\..\..\..\libs\spandsp\src;%(RootDir)%(Directory)..\..\..\..\libs\tiff-3.8.2\libtiff;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
+      <PrecompiledHeader>
+      </PrecompiledHeader>
+      <WarningLevel>Level4</WarningLevel>
+      <TreatWarningAsError>false</TreatWarningAsError>
+    </ClCompile>
+    <Link>
+      <AdditionalOptions>rpcrt4.lib  "..\..\..\..\libs\spandsp\src\x64\Debug\libtiff.lib" "..\..\..\..\x64\Debug\libspandsp.lib" %(AdditionalOptions)</AdditionalOptions>
+      <RandomizedBaseAddress>false</RandomizedBaseAddress>
+      <DataExecutionPrevention>
+      </DataExecutionPrevention>
+    </Link>
+  </ItemDefinitionGroup>
   <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">
     <ClCompile>
       <AdditionalIncludeDirectories>%(RootDir)%(Directory)..\..\..\..\libs\spandsp\src\msvc;%(RootDir)%(Directory)..\..\..\..\libs\spandsp\src;%(RootDir)%(Directory)..\..\..\..\libs\tiff-3.8.2\libtiff;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
@@ -68,6 +107,19 @@
       </DataExecutionPrevention>
     </Link>
   </ItemDefinitionGroup>
+  <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Release|x64'">
+    <ClCompile>
+      <AdditionalIncludeDirectories>%(RootDir)%(Directory)..\..\..\..\libs\spandsp\src\msvc;%(RootDir)%(Directory)..\..\..\..\libs\spandsp\src;%(RootDir)%(Directory)..\..\..\..\libs\tiff-3.8.2\libtiff;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
+      <PrecompiledHeader>
+      </PrecompiledHeader>
+    </ClCompile>
+    <Link>
+      <AdditionalOptions>rpcrt4.lib "..\..\..\..\libs\spandsp\src\x64\Release\libtiff.lib" "..\..\..\..\x64\Release\libspandsp.lib" %(AdditionalOptions)</AdditionalOptions>
+      <RandomizedBaseAddress>false</RandomizedBaseAddress>
+      <DataExecutionPrevention>
+      </DataExecutionPrevention>
+    </Link>
+  </ItemDefinitionGroup>
   <ItemGroup>
     <ClCompile Include="mod_skypopen.c" />
     <ClCompile Include="skypopen_protocol.c" />

src/mod/endpoints/mod_skypopen/skypopen_protocol.c

@@ -1329,7 +1329,7 @@ LRESULT APIENTRY skypopen_present(HWND hWindow, UINT uiMessage, WPARAM uiParam,
 
 	lReturnCode = 0;
 	fIssueDefProc = 0;
-	tech_pvt = (private_t *) GetWindowLong(hWindow, GWL_USERDATA);
+	tech_pvt = (private_t *) GetWindowLong(hWindow, GWLP_USERDATA);
 
 	if (!running) {
 		DEBUGA_SKYPE("let's DIE!\n", SKYPOPEN_P_LOG);
@@ -1340,7 +1340,7 @@ LRESULT APIENTRY skypopen_present(HWND hWindow, UINT uiMessage, WPARAM uiParam,
 	switch (uiMessage) {
 	case WM_CREATE:
 		tech_pvt = (private_t *) ((LPCREATESTRUCT) ulParam)->lpCreateParams;
-		SetWindowLong(hWindow, GWL_USERDATA, (LONG) tech_pvt);
+		SetWindowLong(hWindow, GWLP_USERDATA, (LONG) tech_pvt);
 		DEBUGA_SKYPE("got CREATE\n", SKYPOPEN_P_LOG);
 		break;
 	case WM_DESTROY:

src/mod/endpoints/mod_sofia/conf/sip_profiles/external.xml

@@ -0,0 +1,95 @@
+<profile name="external">
+  <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --> 
+  <!-- This profile is only for outbound registrations to providers -->
+  <gateways>
+    <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
+  </gateways>
+
+  <aliases>
+    <!-- 
+    <alias name="outbound"/>
+    <alias name="nat"/>
+    -->
+  </aliases>
+
+  <domains>
+    <domain name="all" alias="false" parse="true"/>
+  </domains>
+
+  <settings>
+    <param name="debug" value="0"/>
+	<!-- If you want FreeSWITCH to shutdown if this profile fails to load, uncomment the next line. -->
+	<!-- <param name="shutdown-on-fail" value="true"/> -->
+    <param name="sip-trace" value="no"/>
+    <param name="sip-capture" value="no"/>        
+    <param name="rfc2833-pt" value="101"/>
+    <!-- RFC 5626 : Send reg-id and sip.instance -->
+    <!--<param name="enable-rfc-5626" value="true"/> -->
+    <param name="sip-port" value="$${external_sip_port}"/>
+    <param name="dialplan" value="XML"/>
+    <param name="context" value="public"/>
+    <param name="dtmf-duration" value="2000"/>
+    <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
+    <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
+    <param name="hold-music" value="$${hold_music}"/>
+    <param name="rtp-timer-name" value="soft"/>
+    <!--<param name="enable-100rel" value="true"/>-->
+    <!--<param name="disable-srv503" value="true"/>-->                
+    <!-- This could be set to "passive" -->
+    <param name="local-network-acl" value="localnet.auto"/>
+    <param name="manage-presence" value="false"/>
+
+    <!-- used to share presence info across sofia profiles 
+	 manage-presence needs to be set to passive on this profile
+	 if you want it to behave as if it were the internal profile 
+	 for presence.
+    -->
+    <!-- Name of the db to use for this profile -->
+    <!--<param name="dbname" value="share_presence"/>-->
+    <!--<param name="presence-hosts" value="$${domain}"/>-->
+    <!--<param name="force-register-domain" value="$${domain}"/>-->
+    <!--all inbound reg will stored in the db using this domain -->
+    <!--<param name="force-register-db-domain" value="$${domain}"/>-->
+    <!-- ************************************************* -->
+
+    <!--<param name="aggressive-nat-detection" value="true"/>-->
+    <param name="inbound-codec-negotiation" value="generous"/>
+    <param name="nonce-ttl" value="60"/>
+    <param name="auth-calls" value="false"/>
+    <!--
+	DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS!
+    -->
+    <param name="rtp-ip" value="$${local_ip_v4}"/>
+    <param name="sip-ip" value="$${local_ip_v4}"/>
+    <param name="ext-rtp-ip" value="auto-nat"/>
+    <param name="ext-sip-ip" value="auto-nat"/>
+    <param name="rtp-timeout-sec" value="300"/>
+    <param name="rtp-hold-timeout-sec" value="1800"/>
+    <!--<param name="enable-3pcc" value="true"/>-->
+
+    <!-- TLS: disabled by default, set to "true" to enable -->
+    <param name="tls" value="$${external_ssl_enable}"/>
+    <!-- Set to true to not bind on the normal sip-port but only on the TLS port -->
+    <param name="tls-only" value="false"/>
+    <!-- additional bind parameters for TLS -->
+    <param name="tls-bind-params" value="transport=tls"/>
+    <!-- Port to listen on for TLS requests. (5081 will be used if unspecified) -->
+    <param name="tls-sip-port" value="$${external_tls_port}"/>
+    <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
+    <param name="tls-cert-dir" value="$${external_ssl_dir}"/>
+    <!-- Optionally set the passphrase password used by openSSL to encrypt/decrypt TLS private key files -->
+    <param name="tls-passphrase" value=""/>
+    <!-- Verify the date on TLS certificates -->
+    <param name="tls-verify-date" value="true"/>
+    <!-- TLS verify policy, when registering/inviting gateways with other servers (outbound) or handling inbound registration/invite requests how should we verify their certificate -->
+    <!-- set to 'in' to only verify incoming connections, 'out' to only verify outgoing connections, 'all' to verify all connections, also 'in_subjects', 'out_subjects' and 'all_subjects' for subject validation. Multiple policies can be split with a '|' pipe -->
+    <param name="tls-verify-policy" value="none"/>
+    <!-- Certificate max verify depth to use for validating peer TLS certificates when the verify policy is not none -->
+    <param name="tls-verify-depth" value="2"/>
+    <!-- If the tls-verify-policy is set to subjects_all or subjects_in this sets which subjects are allowed, multiple subjects can be split with a '|' pipe -->
+    <param name="tls-verify-in-subjects" value=""/>
+    <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
+    <param name="tls-version" value="$${sip_tls_version}"/>
+
+  </settings>
+</profile>

src/mod/endpoints/mod_sofia/conf/sip_profiles/external/example.xml

@@ -0,0 +1,38 @@
+<include>
+  <!--<gateway name="asterlink.com">-->
+  <!--/// account username *required* ///-->
+  <!--<param name="username" value="cluecon"/>-->
+  <!--/// auth realm: *optional* same as gateway name, if blank ///-->
+  <!--<param name="realm" value="asterlink.com"/>-->
+  <!--/// username to use in from: *optional* same as  username, if blank ///-->
+  <!--<param name="from-user" value="cluecon"/>-->
+  <!--/// domain to use in from: *optional* same as  realm, if blank ///-->
+  <!--<param name="from-domain" value="asterlink.com"/>-->
+  <!--/// account password *required* ///-->
+  <!--<param name="password" value="2007"/>--> 
+  <!--/// extension for inbound calls: *optional* same as username, if blank ///-->
+  <!--<param name="extension" value="cluecon"/>-->
+  <!--/// proxy host: *optional* same as realm, if blank ///-->
+  <!--<param name="proxy" value="asterlink.com"/>-->
+  <!--/// send register to this proxy: *optional* same as proxy, if blank ///-->
+  <!--<param name="register-proxy" value="mysbc.com"/>-->
+  <!--/// expire in seconds: *optional* 3600, if blank ///-->
+  <!--<param name="expire-seconds" value="60"/>-->
+  <!--/// do not register ///-->
+  <!--<param name="register" value="false"/>-->
+  <!-- which transport to use for register -->
+  <!--<param name="register-transport" value="udp"/>-->
+  <!--How many seconds before a retry when a failure or timeout occurs -->
+  <!--<param name="retry-seconds" value="30"/>-->
+  <!--Use the callerid of an inbound call in the from field on outbound calls via this gateway -->
+  <!--<param name="caller-id-in-from" value="false"/>-->
+  <!--extra sip params to send in the contact-->
+  <!--<param name="contact-params" value="tport=tcp"/>-->
+  <!--send an options ping every x seconds, failure will unregister and/or mark it down-->
+  <!--<param name="ping" value="25"/>-->
+  <!--</gateway>-->
+  <!--rfc5626 : Abilitazione rfc5626 ///-->
+  <!--<param name="rfc-5626" value="true"/>-->
+  <!--rfc5626 : extra sip params to send in the contact-->
+  <!--<param name="reg-id" value="1"/>-->
+</include>

src/mod/endpoints/mod_sofia/conf/sip_profiles/internal-ipv6.xml

@@ -0,0 +1,130 @@
+<profile name="internal-ipv6">
+  <!--
+      This is an example of a sofia profile setup to listen on IPv6.
+  -->
+  <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
+  <!--aliases are other names that will work as a valid profile name for this profile-->
+  <settings>
+    <!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
+    <param name="debug" value="0"/>
+    <param name="sip-trace" value="no"/>
+    <param name="context" value="public"/>
+    <param name="rfc2833-pt" value="101"/>
+    <!-- port to bind to for sip traffic -->
+    <param name="sip-port" value="$${internal_sip_port}"/>
+    <param name="dialplan" value="XML"/>
+    <param name="dtmf-duration" value="2000"/>
+    <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
+    <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>
+    <param name="use-rtp-timer" value="true"/>
+    <param name="rtp-timer-name" value="soft"/>
+    <!-- ip address to use for rtp -->
+    <param name="rtp-ip" value="$${local_ip_v6}"/>
+    <!-- ip address to bind to -->
+    <param name="sip-ip" value="$${local_ip_v6}"/>
+    <param name="hold-music" value="$${hold_music}"/>
+    <!--<param name="enable-100rel" value="false"/>-->
+    <!--<param name="disable-srv503" value="true"/>-->
+    <param name="apply-inbound-acl" value="domains"/>
+    <!--<param name="apply-register-acl" value="domains"/>-->
+    <!--<param name="dtmf-type" value="info"/>-->
+    <param name="record-template" value="$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
+    <!--enable to use presence and mwi -->
+    <param name="manage-presence" value="true"/>
+    <!-- This setting is for AAL2 bitpacking on G726 -->
+    <!-- <param name="bitpacking" value="aal2"/> -->
+    <!--max number of open dialogs in proceeding -->
+    <!--<param name="max-proceeding" value="1000"/>-->
+    <!--session timers for all call to expire after the specified seconds -->
+    <!--<param name="session-timeout" value="1800"/>-->
+    <!--<param name="multiple-registrations" value="true"/>-->
+    <!--set to 'greedy' if you want your codec list to take precedence -->
+    <param name="inbound-codec-negotiation" value="generous"/>
+    <!-- if you want to send any special bind params of your own -->
+    <!--<param name="bind-params" value="transport=udp"/>-->
+    <!--<param name="unregister-on-options-fail" value="true"/>-->
+
+    <!-- TLS: disabled by default, set to "true" to enable -->
+    <param name="tls" value="$${internal_ssl_enable}"/>
+    <!-- additional bind parameters for TLS -->
+    <param name="tls-bind-params" value="transport=tls"/>
+    <!-- Port to listen on for TLS requests. (5061 will be used if unspecified) -->
+    <param name="tls-sip-port" value="$${internal_tls_port}"/>
+    <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
+    <param name="tls-cert-dir" value="$${internal_ssl_dir}"/>
+    <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
+    <param name="tls-version" value="$${sip_tls_version}"/>
+    
+    <!--If you don't want to pass through timestampes from 1 RTP call to another (on a per call basis with rtp_rewrite_timestamps chanvar)-->
+    <!--<param name="rtp-rewrite-timestamps" value="true"/>-->
+    <!--<param name="pass-rfc2833" value="true"/>-->
+    <!--If you have ODBC support and a working dsn you can use it instead of SQLite-->
+    <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
+    
+    <!--Uncomment to set all inbound calls to no media mode-->
+    <!--<param name="inbound-bypass-media" value="true"/>-->
+
+    <!--Uncomment to set all inbound calls to proxy media mode-->
+    <!--<param name="inbound-proxy-media" value="true"/>-->
+    
+    <!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok-->
+    <!--<param name="inbound-late-negotiation" value="true"/>-->
+    
+    <!-- this lets anything register -->
+    <!--  comment the next line and uncomment one or both of the other 2 lines for call authentication -->
+    <!-- <param name="accept-blind-reg" value="true"/> -->
+
+    <!-- accept any authentication without actually checking (not a good feature for most people) -->
+    <!-- <param name="accept-blind-auth" value="true"/> -->
+    
+    <!-- suppress CNG on this profile or per call with the 'suppress_cng' variable -->
+    <!-- <param name="suppress-cng" value="true"/> -->
+    
+    <!--TTL for nonce in sip auth-->
+    <param name="nonce-ttl" value="60"/>
+    <!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec 
+	that the originator is using-->
+    <!--<param name="disable-transcoding" value="true"/>-->
+    <!-- Used for when phones respond to a challenged ACK with method INVITE in the hash -->
+    <!--<param name="NDLB-broken-auth-hash" value="true"/>-->
+    <!-- add a ;received="<ip>:<port>" to the contact when replying to register for nat handling -->
+    <!--<param name="NDLB-received-in-nat-reg-contact" value="true"/>-->
+    <param name="auth-calls" value="$${internal_auth_calls}"/>
+    <!-- on authed calls, authenticate *all* the packets not just invite -->
+    <param name="auth-all-packets" value="false"/>
+    <!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> -->
+    <!-- <param name="ext-sip-ip" value="$${external_sip_ip}"/> -->
+    <!-- rtp inactivity timeout -->
+    <param name="rtp-timeout-sec" value="300"/>
+    <param name="rtp-hold-timeout-sec" value="1800"/>
+    <!-- VAD choose one (out is a good choice); -->
+    <!-- <param name="vad" value="in"/> -->
+    <!-- <param name="vad" value="out"/> -->
+    <!-- <param name="vad" value="both"/> -->
+    <!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
+    <!--
+	These are enabled to make the default config work better out of the box.
+	If you need more than ONE domain you'll need to not use these options.
+
+    -->
+    <!--all inbound reg will look in this domain for the users -->
+    <param name="force-register-domain" value="$${domain}"/>
+    <!--all inbound reg will stored in the db using this domain -->
+    <param name="force-register-db-domain" value="$${domain}"/>
+    <!-- disable register and transfer which may be undesirable in a public switch -->
+    <!--<param name="disable-transfer" value="true"/>-->
+    <!--<param name="disable-register" value="true"/>-->
+    <!--<param name="enable-3pcc" value="true"/>-->
+    <!-- use stun when specified (default is true) -->
+    <!--<param name="stun-enabled" value="true"/>-->
+    <!-- use stun when specified (default is true) -->
+    <!-- set to true to have the profile determine stun is not useful and turn it off globally-->
+    <!--<param name="stun-auto-disable" value="true"/>-->
+
+	 <!-- the following can be used as workaround with bogus SRV/NAPTR records -->
+	 <!--<param name="disable-srv" value="false" />-->
+	 <!--<param name="disable-naptr" value="false" />-->
+
+  </settings>
+</profile>
+

src/mod/endpoints/mod_sofia/conf/sip_profiles/internal.xml

@@ -0,0 +1,385 @@
+<profile name="internal">
+  <!--
+      This is a sofia sip profile/user agent.  This will service exactly one ip and port.
+      In FreeSWITCH you can run multiple sip user agents on their own ip and port.
+      
+      When you hear someone say "sofia profile" this is what they are talking about.
+  -->
+  
+  <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
+  <!--aliases are other names that will work as a valid profile name for this profile-->
+  <aliases>
+    <!--
+    <alias name="default"/>
+    -->
+  </aliases>
+  <!-- Outbound Registrations -->
+  <gateways>
+    <X-PRE-PROCESS cmd="include" data="internal/*.xml"/>
+  </gateways>
+  
+  <domains>
+    <!-- indicator to parse the directory for domains with parse="true" to get gateways-->
+    <!--<domain name="$${domain}" parse="true"/>-->
+    <!-- indicator to parse the directory for domains with parse="true" to get gateways and alias every domain to this profile -->
+    <!--<domain name="all" alias="true" parse="true"/>-->
+    <domain name="all" alias="true" parse="false"/> 
+  </domains>
+  
+  <settings>
+    <!--
+	When calls are in no media this will bring them back to media
+	when you press the hold button.
+    -->
+    <!--<param name="media-option" value="resume-media-on-hold"/> -->
+    <!--
+	This will allow a call after an attended transfer go back to 
+	 bypass media after an attended transfer.
+    -->
+    <!--<param name="media-option" value="bypass-media-after-att-xfer"/>-->
+    <!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
+    <param name="debug" value="0"/>
+	<!-- If you want FreeSWITCH to shutdown if this profile fails to load, uncomment the next line. -->
+	<!-- <param name="shutdown-on-fail" value="true"/> -->
+    <param name="sip-trace" value="no"/>
+    <param name="sip-capture" value="no"/>
+
+    <!-- Use presence_map.conf.xml to convert extension regex to presence protos for routing -->
+    <!-- <param name="presence-proto-lookup" value="true"/> -->
+    
+
+    <!-- Don't be picky about negotiated DTMF just always offer 2833 and accept both 2833 and INFO -->
+    <!--<param name="liberal-dtmf" value="true"/>-->
+
+
+    <!-- 
+        Sometimes, in extremely rare edge cases, the Sofia SIP stack may stop
+        responding. These options allow you to enable and control a watchdog
+        on the Sofia SIP stack so that if it stops responding for the
+        specified number of milliseconds, it will cause FreeSWITCH to crash
+        immediately. This is useful if you run in an HA environment and
+        need to ensure automated recovery from such a condition. Note that if
+        your server is idle a lot, the watchdog may fire due to not receiving
+        any SIP messages. Thus, if you expect your system to be idle, you
+        should leave the watchdog disabled. It can be toggled on and off
+        through the FreeSWITCH CLI either on an individual profile basis or
+        globally for all profiles. So, if you run in an HA environment with a
+        master and slave, you should use the CLI to make sure the watchdog is
+        only enabled on the master. 
+        If such crash occurs, FreeSWITCH will dump core if allowed. The
+        stacktrace will include function watchdog_triggered_abort().
+    -->
+    <param name="watchdog-enabled" value="no"/>
+    <param name="watchdog-step-timeout" value="30000"/>
+    <param name="watchdog-event-timeout" value="30000"/>
+
+    <param name="log-auth-failures" value="false"/>
+    <param name="forward-unsolicited-mwi-notify" value="false"/>
+
+    <param name="context" value="public"/>
+    <param name="rfc2833-pt" value="101"/>
+    <!-- port to bind to for sip traffic -->
+    <param name="sip-port" value="$${internal_sip_port}"/>
+    <param name="dialplan" value="XML"/>
+    <param name="dtmf-duration" value="2000"/>
+    <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
+    <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>
+    <param name="rtp-timer-name" value="soft"/>
+    <!-- ip address to use for rtp, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
+    <param name="rtp-ip" value="$${local_ip_v4}"/>
+    <!-- ip address to bind to, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
+    <param name="sip-ip" value="$${local_ip_v4}"/>
+    <param name="hold-music" value="$${hold_music}"/>
+    <param name="apply-nat-acl" value="nat.auto"/>
+
+
+    <!-- (default true) set to false if you do not wish to have called party info in 1XX responses -->
+    <!-- <param name="cid-in-1xx" value="false"/> -->
+
+    <!-- extended info parsing -->
+    <!-- <param name="extended-info-parsing" value="true"/> -->
+
+    <!--<param name="aggressive-nat-detection" value="true"/>-->
+    <!--
+	There are known issues (asserts and segfaults) when 100rel is enabled.
+	It is not recommended to enable 100rel at this time.
+    -->
+    <!--<param name="enable-100rel" value="true"/>-->
+
+    <!-- uncomment if you don't wish to try a next SRV destination on 503 response -->
+    <!-- RFC3263 Section 4.3 -->	    
+    <!--<param name="disable-srv503" value="true"/>-->    
+    
+    <!-- Enable Compact SIP headers. -->
+    <!--<param name="enable-compact-headers" value="true"/>-->
+    <!--
+	enable/disable session timers 
+    -->
+    <!--<param name="enable-timer" value="false"/>-->
+    <!--<param name="minimum-session-expires" value="120"/>-->
+    <param name="apply-inbound-acl" value="domains"/>
+    <!--
+	This defines your local network, by default we detect your local network
+	and create this localnet.auto ACL for this.
+    -->
+    <param name="local-network-acl" value="localnet.auto"/>
+    <!--<param name="apply-register-acl" value="domains"/>-->
+    <!--<param name="dtmf-type" value="info"/>-->
+
+
+    <!-- 'true' means every time 'first-only' means on the first register -->
+    <!--<param name="send-message-query-on-register" value="true"/>-->
+    
+    <!-- 'true' means every time 'first-only' means on the first register -->
+    <!--<param name="send-presence-on-register" value="first-only"/> -->
+      
+
+    <!-- Caller-ID type (choose one, can be overridden by inbound call type and/or sip_cid_type channel variable -->
+    <!-- Remote-Party-ID header -->
+    <!--<param name="caller-id-type" value="rpid"/>-->
+
+    <!-- P-*-Identity family of headers -->
+    <!--<param name="caller-id-type" value="pid"/>-->
+
+    <!-- neither one -->
+    <!--<param name="caller-id-type" value="none"/>-->
+
+
+
+    <param name="record-path" value="$${recordings_dir}"/>
+    <param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
+    <!--enable to use presence -->
+    <param name="manage-presence" value="true"/>
+    <!-- send a presence probe on each register to query devices to send presence instead of sending presence with less info -->
+    <!--<param name="presence-probe-on-register" value="true"/>-->
+    <!--<param name="manage-shared-appearance" value="true"/>-->
+    <!-- used to share presence info across sofia profiles -->
+    <!-- Name of the db to use for this profile -->
+    <!--<param name="dbname" value="share_presence"/>-->
+    <param name="presence-hosts" value="$${domain},$${local_ip_v4}"/>
+    <param name="presence-privacy" value="$${presence_privacy}"/>
+    <!-- ************************************************* -->
+    
+    <!-- This setting is for AAL2 bitpacking on G726 -->
+    <!-- <param name="bitpacking" value="aal2"/> -->
+    <!--max number of open dialogs in proceeding -->
+    <!--<param name="max-proceeding" value="1000"/>-->
+    <!--session timers for all call to expire after the specified seconds -->
+    <!--<param name="session-timeout" value="1800"/>-->
+    <!-- Can be 'true' or 'contact' -->
+    <!--<param name="multiple-registrations" value="contact"/>-->
+    <!--set to 'greedy' if you want your codec list to take precedence -->
+    <param name="inbound-codec-negotiation" value="generous"/>
+    <!-- if you want to send any special bind params of your own -->
+    <!--<param name="bind-params" value="transport=udp"/>-->
+    <!--<param name="unregister-on-options-fail" value="true"/>-->
+
+    <!-- TLS: disabled by default, set to "true" to enable -->
+    <param name="tls" value="$${internal_ssl_enable}"/>
+    <!-- Set to true to not bind on the normal sip-port but only on the TLS port -->
+    <param name="tls-only" value="false"/>
+    <!-- additional bind parameters for TLS -->
+    <param name="tls-bind-params" value="transport=tls"/>
+    <!-- Port to listen on for TLS requests. (5061 will be used if unspecified) -->
+    <param name="tls-sip-port" value="$${internal_tls_port}"/>
+    <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
+    <param name="tls-cert-dir" value="$${internal_ssl_dir}"/>
+    <!-- Optionally set the passphrase password used by openSSL to encrypt/decrypt TLS private key files -->
+    <param name="tls-passphrase" value=""/>
+    <!-- Verify the date on TLS certificates -->
+    <param name="tls-verify-date" value="true"/>
+    <!-- TLS verify policy, when registering/inviting gateways with other servers (outbound) or handling inbound registration/invite requests how should we verify their certificate -->
+    <!-- set to 'in' to only verify incoming connections, 'out' to only verify outgoing connections, 'all' to verify all connections, also 'in_subjects', 'out_subjects' and 'all_subjects' for subject validation. Multiple policies can be split with a '|' pipe -->
+    <param name="tls-verify-policy" value="none"/>
+    <!-- Certificate max verify depth to use for validating peer TLS certificates when the verify policy is not none -->
+    <param name="tls-verify-depth" value="2"/>
+    <!-- If the tls-verify-policy is set to subjects_all or subjects_in this sets which subjects are allowed, multiple subjects can be split with a '|' pipe -->
+    <param name="tls-verify-in-subjects" value=""/>
+    <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
+    <param name="tls-version" value="$${sip_tls_version}"/>
+
+    <!-- turn on auto-flush during bridge (skip timer sleep when the socket already has data) 
+	 (reduces delay on latent connections default true, must be disabled explicitly)-->
+    <!--<param name="rtp-autoflush-during-bridge" value="false"/>-->
+    
+    <!--If you don't want to pass through timestamps from 1 RTP call to another (on a per call basis with rtp_rewrite_timestamps chanvar)-->
+    <!--<param name="rtp-rewrite-timestamps" value="true"/>-->
+    <!--<param name="pass-rfc2833" value="true"/>-->
+    <!--If you have ODBC support and a working dsn you can use it instead of SQLite-->
+    <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
+    
+    <!--Uncomment to set all inbound calls to no media mode-->
+    <!--<param name="inbound-bypass-media" value="true"/>-->
+
+    <!--Uncomment to set all inbound calls to proxy media mode-->
+    <!--<param name="inbound-proxy-media" value="true"/>-->
+    
+    <!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok-->
+    <!--<param name="inbound-late-negotiation" value="true"/>-->
+    
+    <!-- this lets anything register -->
+    <!--  comment the next line and uncomment one or both of the other 2 lines for call authentication -->
+    <!-- <param name="accept-blind-reg" value="true"/> -->
+
+    <!-- accept any authentication without actually checking (not a good feature for most people) -->
+    <!-- <param name="accept-blind-auth" value="true"/> -->
+    
+    <!-- suppress CNG on this profile or per call with the 'suppress_cng' variable -->
+    <!-- <param name="suppress-cng" value="true"/> -->
+    
+    <!--TTL for nonce in sip auth-->
+    <param name="nonce-ttl" value="60"/>
+    <!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec 
+	that the originator is using-->
+    <!--<param name="disable-transcoding" value="true"/>-->
+    <!-- Handle 302 Redirect in the dialplan -->
+    <!--<param name="manual-redirect" value="true"/> -->
+    <!-- Disable Transfer -->
+    <!--<param name="disable-transfer" value="true"/> -->
+    <!-- Disable Register -->
+    <!--<param name="disable-register" value="true"/> -->
+    <!-- Used for when phones respond to a challenged ACK with method INVITE in the hash -->
+    <!--<param name="NDLB-broken-auth-hash" value="true"/>-->
+    <!-- add a ;received="<ip>:<port>" to the contact when replying to register for nat handling -->
+    <!--<param name="NDLB-received-in-nat-reg-contact" value="true"/>-->
+    <param name="auth-calls" value="$${internal_auth_calls}"/>
+    <!-- Force the user and auth-user to match. -->
+    <param name="inbound-reg-force-matching-username" value="true"/>
+    <!-- on authed calls, authenticate *all* the packets not just invite -->
+    <param name="auth-all-packets" value="false"/>
+    
+    <!-- external_sip_ip
+      Used as the public IP address for SDP.
+      Can be an one of:
+           ip address            - "12.34.56.78"
+           a stun server lookup  - "stun:stun.server.com"
+           a DNS name            - "host:host.server.com"
+           auto                  - Use guessed ip.
+           auto-nat              - Use ip learned from NAT-PMP or UPNP
+       -->
+    <param name="ext-rtp-ip" value="auto-nat"/>
+    <param name="ext-sip-ip" value="auto-nat"/>
+
+    <!-- rtp inactivity timeout -->
+    <param name="rtp-timeout-sec" value="300"/>
+    <param name="rtp-hold-timeout-sec" value="1800"/>
+    <!-- VAD choose one (out is a good choice); -->
+    <!-- <param name="vad" value="in"/> -->
+    <!-- <param name="vad" value="out"/> -->
+    <!-- <param name="vad" value="both"/> -->
+    <!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
+    <!--
+	These are enabled to make the default config work better out of the box.
+	If you need more than ONE domain you'll need to not use these options.
+
+    -->
+    <!--all inbound reg will look in this domain for the users -->
+    <param name="force-register-domain" value="$${domain}"/>
+    <!--force the domain in subscriptions to this value -->
+    <param name="force-subscription-domain" value="$${domain}"/>
+    <!--all inbound reg will stored in the db using this domain -->
+    <param name="force-register-db-domain" value="$${domain}"/>
+
+    <!--<param name="delete-subs-on-register" value="false"/>-->
+
+    <!-- enable rtcp on every channel also can be done per leg basis with rtcp_audio_interval_msec variable set to passthru to pass it across a call-->
+    <!--<param name="rtcp-audio-interval-msec" value="5000"/>-->
+    <!--<param name="rtcp-video-interval-msec" value="5000"/>-->
+
+    <!--force suscription expires to a lower value than requested-->
+    <!--<param name="force-subscription-expires" value="60"/>-->
+    <!-- disable register and transfer which may be undesirable in a public switch -->
+    <!--<param name="disable-transfer" value="true"/>-->
+    <!--<param name="disable-register" value="true"/>-->
+
+    <!-- 
+	 enable-3pcc can be set to either 'true' or 'proxy', true accepts the call
+	 right away, proxy waits until the call has been answered then sends accepts
+    -->
+    <!--<param name="enable-3pcc" value="true"/>-->
+	
+    <!-- use at your own risk or if you know what this does.-->
+    <!--<param name="NDLB-force-rport" value="true"/>-->
+    <!--
+	Choose the realm challenge key. Default is auto_to if not set. 
+	
+	auto_from  - uses the from field as the value for the sip realm.
+	auto_to    - uses the to field as the value for the sip realm.
+	<anyvalue> - you can input any value to use for the sip realm.
+
+	If you want URL dialing to work you'll want to set this to auto_from.
+	
+	If you use any other value besides auto_to or auto_from you'll loose
+	the ability to do multiple domains.
+	
+	Note: comment out to restore the behavior before 2008-09-29
+
+    -->
+    <param name="challenge-realm" value="auto_from"/>
+    <!--<param name="disable-rtp-auto-adjust" value="true"/>-->
+    <!-- on inbound calls make the uuid of the session equal to the sip call id of that call -->
+    <!--<param name="inbound-use-callid-as-uuid" value="true"/>-->
+    <!-- on outbound calls set the callid to match the uuid of the session -->
+    <!--<param name="outbound-use-uuid-as-callid" value="true"/>-->
+    <!-- set to false disable this feature -->
+    <!--<param name="rtp-autofix-timing" value="false"/>-->
+
+    <!-- set this param to false if your gateway for some reason hates X- headers that it is supposed to ignore-->
+    <!--<param name="pass-callee-id" value="false"/>-->
+
+    <!-- clear clears them all or supply the name to add or the name prefixed with ~ to remove
+	 valid values:
+
+	 clear
+	 CISCO_SKIP_MARK_BIT_2833
+	 SONUS_SEND_INVALID_TIMESTAMP_2833
+
+    -->
+    <!--<param name="auto-rtp-bugs" data="clear"/>-->
+
+	 <!-- the following can be used as workaround with bogus SRV/NAPTR records -->
+	 <!--<param name="disable-srv" value="false" />-->
+	 <!--<param name="disable-naptr" value="false" />-->
+
+	<!-- The following can be used to fine-tune timers within sofia's transport layer 
+		 Those settings are for advanced users and can safely be left as-is -->
+		
+	<!-- Initial retransmission interval (in milliseconds).
+		Set the T1 retransmission interval used by the SIP transaction engine. 
+		The T1 is the initial duration used by request retransmission timers A and E (UDP) as well as response retransmission timer G. 	 -->
+	<!-- <param name="timer-T1" value="500" /> -->
+	
+	<!--  Transaction timeout (defaults to T1 * 64).
+		Set the T1x64 timeout value used by the SIP transaction engine.
+		The T1x64 is duration used for timers B, F, H, and J (UDP) by the SIP transaction engine. 
+		The timeout value T1x64 can be adjusted separately from the initial retransmission interval T1. -->
+	<!-- <param name="timer-T1X64" value="32000" /> -->
+	
+	
+	<!-- Maximum retransmission interval (in milliseconds).
+		Set the maximum retransmission interval used by the SIP transaction engine. 
+		The T2 is the maximum duration used for the timers E (UDP) and G by the SIP transaction engine. 
+		Note that the timer A is not capped by T2. Retransmission interval of INVITE requests grows exponentially 
+		until the timer B fires.  -->
+	<!-- <param name="timer-T2" value="4000" /> -->
+		
+	<!--
+		Transaction lifetime (in milliseconds).
+		Set the lifetime for completed transactions used by the SIP transaction engine. 
+		A completed transaction is kept around for the duration of T4 in order to catch late responses. 
+		The T4 is the maximum duration for the messages to stay in the network and the duration of SIP timer K. -->
+	<!-- <param name="timer-T4" value="4000" /> -->
+
+	<!-- Turn on a jitterbuffer for every call -->
+	<!-- <param name="auto-jitterbuffer-msec" value="60"/> -->
+
+
+	<!-- By default mod_sofia will ignore the codecs in the sdp for hold/unhold operations 
+	     Set this to true if you want to actually parse the sdp and re-negotiate the codec during hold/unhold.
+	     It's probably not what you want so stick with the default unless you really need to change this.
+	-->
+	<!--<param name="renegotiate-codec-on-hold" value="true"/>-->
+    
+  </settings>
+</profile>
+