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Merge branch 'fsorig'

Browse Source 
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
This commit is contained in:
David Yat Sin
2012-04-03 18:08:49 -04:00
parent 1ad0bc250c 66d90d5dfc
commit 27e4cdb4dd
940 changed files with 118344 additions and 11720 deletions
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src/mod/applications/mod_abstraction/conf/autoload_configs/abstraction.conf.xml
+5
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@@ -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
+11
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@@ -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
+38
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@@ -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
+44 -10
View File
@@ -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
+33
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@@ -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
+36 -11
View File
@@ -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
+209
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@@ -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
+31
View File
@@ -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
+19 -10
View File
@@ -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
+6
View File
@@ -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
+5
View File
@@ -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
+10
View File
@@ -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
+8
View File
@@ -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
+21 -5
View File
@@ -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
+28
View File
@@ -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
+27
View File
@@ -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
+3 -3
View File
@@ -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
+13
View File
@@ -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}/$1"/>-->
</routes>
</configuration>
src/mod/applications/mod_enum/conf/dialplan/99999_enum.xml
+8
View File
@@ -0,0 +1,8 @@
<include>
<extension name="enum">
<condition field="${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
+10
View File
@@ -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
+19
View File
@@ -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
+6
View File
@@ -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
+7
View File
@@ -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/httapi.conf.xml → src/mod/applications/mod_httapi/conf/autoload_configs/httapi.conf.xml
View File
src/mod/applications/mod_httapi/mod_httapi_doc.txt → src/mod/applications/mod_httapi/docs/mod_httapi_doc.txt
+15
View File
@@ -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
+48 -2
View File
@@ -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
+8
View File
@@ -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
+7
View File
@@ -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
+77
View File
@@ -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
+4
View File
@@ -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
+89
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@@ -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, ${lcr_rate_field}, random();
"/>
</profile>
-->
</profiles>
</configuration>
src/mod/applications/mod_lcr/conf/dialplan/lcr.xml
+8
View File
@@ -0,0 +1,8 @@
<include>
<extension name="Outbound to PSTN 11 Digits">
<condition field="destination_number" expression="^(1[2-9][0-9]{2}[2-9]{7})$">
<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
+5
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@@ -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
+14
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@@ -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
+6
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@@ -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
+9
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@@ -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
+20
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@@ -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
+50
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@@ -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
+27
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@@ -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
+73 -42
View File
@@ -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
+53
View File
@@ -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
+7
View File
@@ -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
+7
View File
@@ -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
+7
View File
@@ -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
+12
View File
@@ -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
+3 -7
View File
@@ -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
+16
View File
@@ -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
+108
View File
@@ -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
+78
View File
@@ -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
+26
View File
@@ -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
+1 -1
View File
@@ -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
+1 -1
View File
@@ -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
+9 -7
View File
@@ -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
+7 -7
View File
@@ -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
+24
View File
@@ -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 $1"/>
</condition>
</extension>
</include>
src/mod/applications/mod_vmd/conf/dialplan/vmd.xml
+12
View File
@@ -0,0 +1,12 @@
<include>
<extension name="vmdtest">
<condition field="destination_number" expression="^vmd(\d{10})$">
<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
+27
View File
@@ -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
+17
View File
@@ -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
+70
View File
@@ -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
+20
View File
@@ -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} ${dialed_extension}"/>
</condition>
</extension>
</include>
src/mod/applications/mod_voicemail/conf/notify-voicemail.tpl
+44
View File
@@ -0,0 +1,44 @@
From: "${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
+44
View File
@@ -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
+12 -7
View File
@@ -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
+208
View File
@@ -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
+5
View File
@@ -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
+12
View File
@@ -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
+2 -1
View File
@@ -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
+8 -8
View File
@@ -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
+8 -9
View File
@@ -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
+14 -2
View File
@@ -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
+127 -2
View File
@@ -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
+21
View File
@@ -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)
${sip_from_user} => bill,n,Goto(default|music)
src/mod/dialplans/mod_dialplan_xml/mod_dialplan_xml.c
+5 -1
View File
@@ -143,6 +143,7 @@ static int parse_exten(switch_core_session_t *session, switch_caller_profile_t *
"Dialplan: %s Date/Time Match (PASS) [%s] break=%s\n",
switch_channel_get_name(channel), exten_name, do_break_a ? do_break_a : "on-false");
anti_action = SWITCH_FALSE;
proceed = 1;
} else if (time_match == 0) {
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG_CLEAN(session), SWITCH_LOG_DEBUG,
"Dialplan: %s Date/TimeMatch (FAIL) [%s] break=%s\n",
@@ -189,7 +190,7 @@ static int parse_exten(switch_core_session_t *session, switch_caller_profile_t *
total++;
field = (char *) switch_xml_attr(xregex, "field");
if (field) {
if (strchr(field, '$')) {
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
+12
View File
@@ -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 -2
View File
@@ -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
+9 -8
View File
@@ -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
+218
View File
@@ -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
+17
View File
@@ -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
+1 -1
View File
@@ -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
+5
View File
@@ -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
+6
View File
@@ -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
+30
View File
@@ -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
+8
View File
@@ -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
+13
View File
@@ -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
+8
View File
@@ -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
+9
View File
@@ -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
+35
View File
@@ -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
+37
View File
@@ -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
-1
View File
@@ -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
+52
View File
@@ -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
+2 -2
View File
@@ -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
+95
View File
@@ -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
Vendored
+38
View File
@@ -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
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<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
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<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>
src/mod/endpoints/mod_sofia/conf/sip_profiles/internal/example.xml
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<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"/>-->
<!-- Put the extension in the contact -->
<!--<param name="extension-in-contact" value="true"/>-->
<!--send an options ping every x seconds, failure will unregister and/or mark it down-->
<!--<param name="ping" value="25"/>-->
<!--<param name="cid-type" value="rpid"/>-->
<!--</gateway>-->
</include>
src/mod/endpoints/mod_sofia/mod_sofia.c
+72 -40
View File
@@ -495,12 +495,16 @@ switch_status_t sofia_on_hangup(switch_core_session_t *session)
val = switch_channel_get_variable(tech_pvt->channel, "disable_q850_reason");
if (!val || switch_false(val)) {
if (switch_channel_test_flag(channel, CF_INTERCEPT) || cause == SWITCH_CAUSE_PICKED_OFF || cause == SWITCH_CAUSE_LOSE_RACE) {
switch_snprintf(reason, sizeof(reason), "SIP;cause=200;text=\"Call completed elsewhere\"");
} else if (cause > 0 && cause < 128) {
switch_snprintf(reason, sizeof(reason), "Q.850;cause=%d;text=\"%s\"", cause, switch_channel_cause2str(cause));
if ((val = switch_channel_get_variable(tech_pvt->channel, "sip_reason_str"))) {
switch_snprintf(reason, sizeof(reason), "%s", val);
} else {
switch_snprintf(reason, sizeof(reason), "%s;cause=%d;text=\"%s\"", tech_pvt->profile->username, cause, switch_channel_cause2str(cause));
if (switch_channel_test_flag(channel, CF_INTERCEPT) || cause == SWITCH_CAUSE_PICKED_OFF || cause == SWITCH_CAUSE_LOSE_RACE) {
switch_snprintf(reason, sizeof(reason), "SIP;cause=200;text=\"Call completed elsewhere\"");
} else if (cause > 0 && cause < 128) {
switch_snprintf(reason, sizeof(reason), "Q.850;cause=%d;text=\"%s\"", cause, switch_channel_cause2str(cause));
} else {
switch_snprintf(reason, sizeof(reason), "%s;cause=%d;text=\"%s\"", tech_pvt->profile->username, cause, switch_channel_cause2str(cause));
}
}
}
@@ -1728,7 +1732,8 @@ static switch_status_t sofia_receive_message(switch_core_session_t *session, swi
}
if ((var = switch_channel_get_variable(channel, SOFIA_SECURE_MEDIA_VARIABLE)) && switch_true(var)) {
if ((var = switch_channel_get_variable(channel, SOFIA_SECURE_MEDIA_VARIABLE)) &&
(switch_true(var) || !strcasecmp(var, SWITCH_RTP_CRYPTO_KEY_32) || !strcasecmp(var, SWITCH_RTP_CRYPTO_KEY_80))) {
sofia_set_flag_locked(tech_pvt, TFLAG_SECURE);
}
@@ -1917,13 +1922,18 @@ static switch_status_t sofia_receive_message(switch_core_session_t *session, swi
{
switch_t38_options_t *t38_options = switch_channel_get_private(tech_pvt->channel, "t38_options");
sofia_glue_set_image_sdp(tech_pvt, t38_options, msg->numeric_arg);
if (t38_options) {
sofia_glue_set_image_sdp(tech_pvt, t38_options, msg->numeric_arg);
if (!switch_channel_test_flag(channel, CF_PROXY_MEDIA)) {
switch_channel_set_flag(channel, CF_REQ_MEDIA);
if (!switch_channel_test_flag(channel, CF_PROXY_MEDIA)) {
switch_channel_set_flag(channel, CF_REQ_MEDIA);
}
sofia_set_flag_locked(tech_pvt, TFLAG_SENT_UPDATE);
sofia_glue_do_invite(session);
} else {
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_WARNING, "%s Request to send IMAGE on channel with not t38 options.\n",
switch_channel_get_name(channel));
}
sofia_set_flag_locked(tech_pvt, TFLAG_SENT_UPDATE);
sofia_glue_do_invite(session);
}
break;
@@ -2033,7 +2043,7 @@ static switch_status_t sofia_receive_message(switch_core_session_t *session, swi
const char *uuid;
const char *call_id = NULL, *to_user = NULL, *to_host = NULL, *to_tag = NULL, *from_tag = NULL, *from_user = NULL, *from_host = NULL;
if ((uuid = switch_channel_get_variable(channel, SWITCH_SIGNAL_BOND_VARIABLE))) {
if ((uuid = switch_channel_get_variable(channel, SWITCH_SIGNAL_BOND_VARIABLE)) || (uuid = switch_channel_get_variable(channel, "originate_signal_bond"))) {
switch_core_session_t *rsession;
if ((rsession = switch_core_session_locate(uuid))) {
switch_channel_t *rchannel = switch_core_session_get_channel(rsession);
@@ -2103,12 +2113,7 @@ static switch_status_t sofia_receive_message(switch_core_session_t *session, swi
number = tech_pvt->caller_profile->destination_number;
}
if (sofia_test_flag(tech_pvt, TFLAG_UPDATING_DISPLAY) || !switch_channel_test_flag(channel, CF_ANSWERED)) {
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(tech_pvt->session), SWITCH_LOG_WARNING,
"Cannot send display update to %s "
"Did not receive reply to last update or channel has not been answered yet.\n",
switch_channel_get_name(tech_pvt->channel));
} else {
if (!sofia_test_flag(tech_pvt, TFLAG_UPDATING_DISPLAY) && switch_channel_test_flag(channel, CF_ANSWERED)) {
if (zstr(tech_pvt->last_sent_callee_id_name) || strcmp(tech_pvt->last_sent_callee_id_name, name) ||
zstr(tech_pvt->last_sent_callee_id_number) || strcmp(tech_pvt->last_sent_callee_id_number, number)) {
@@ -2159,6 +2164,7 @@ static switch_status_t sofia_receive_message(switch_core_session_t *session, swi
nua_update(tech_pvt->nh,
NUTAG_SESSION_TIMER(tech_pvt->session_timeout),
NUTAG_SESSION_REFRESHER(tech_pvt->session_refresher),
TAG_IF(!zstr(tech_pvt->route_uri), NUTAG_PROXY(tech_pvt->route_uri)),
TAG_IF(!zstr_buf(message), SIPTAG_HEADER_STR(message)),
TAG_IF(!zstr(tech_pvt->user_via), SIPTAG_VIA_STR(tech_pvt->user_via)), TAG_END());
} else if ((ua && (switch_stristr("Yealink", ua)))) {
@@ -2168,6 +2174,7 @@ static switch_status_t sofia_receive_message(switch_core_session_t *session, swi
nua_update(tech_pvt->nh,
NUTAG_SESSION_TIMER(tech_pvt->session_timeout),
NUTAG_SESSION_REFRESHER(tech_pvt->session_refresher),
TAG_IF(!zstr(tech_pvt->route_uri), NUTAG_PROXY(tech_pvt->route_uri)),
TAG_IF(!zstr_buf(message), SIPTAG_HEADER_STR(message)),
TAG_IF(!zstr(tech_pvt->user_via), SIPTAG_VIA_STR(tech_pvt->user_via)), TAG_END());
}
@@ -2566,6 +2573,8 @@ static switch_status_t sofia_receive_message(switch_core_session_t *session, swi
}
}
switch_channel_check_zrtp(tech_pvt->channel);
if ((status = sofia_glue_tech_choose_port(tech_pvt, 0)) != SWITCH_STATUS_SUCCESS) {
switch_channel_hangup(channel, SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER);
goto end_lock;
@@ -2803,8 +2812,29 @@ static int show_reg_callback_xml(void *pArg, int argc, char **argv, char **colum
return 0;
}
static const char *status_names[] = { "DOWN", "UP", NULL };
static int sql2str_callback(void *pArg, int argc, char **argv, char **columnNames)
{
struct cb_helper_sql2str *cbt = (struct cb_helper_sql2str *) pArg;
switch_copy_string(cbt->buf, argv[0], cbt->len);
cbt->matches++;
return 0;
}
static uint32_t sofia_profile_reg_count(sofia_profile_t *profile)
{
struct cb_helper_sql2str cb;
char reg_count[80] = "";
char *sql;
cb.buf = reg_count;
cb.len = sizeof(reg_count);
sql = switch_mprintf("select count(*) from sip_registrations where profile_name = '%q'", profile->name);
sofia_glue_execute_sql_callback(profile, profile->ireg_mutex, sql, sql2str_callback, &cb);
free(sql);
return strtoul(reg_count, NULL, 10);
}
static const char *status_names[] = { "DOWN", "UP", NULL };
static switch_status_t cmd_status(char **argv, int argc, switch_stream_handle_t *stream)
{
@@ -2978,6 +3008,7 @@ static switch_status_t cmd_status(char **argv, int argc, switch_stream_handle_t
stream->write_function(stream, "FAILED-CALLS-IN \t%u\n", profile->ib_failed_calls);
stream->write_function(stream, "CALLS-OUT \t%u\n", profile->ob_calls);
stream->write_function(stream, "FAILED-CALLS-OUT \t%u\n", profile->ob_failed_calls);
stream->write_function(stream, "REGISTRATIONS \t%lu\n", sofia_profile_reg_count(profile));
}
cb.profile = profile;
@@ -3254,9 +3285,9 @@ static switch_status_t cmd_xml_status(char **argv, int argc, switch_stream_handl
stream->write_function(stream, " <calls-out>%u</calls-out>\n", profile->ob_calls);
stream->write_function(stream, " <failed-calls-in>%u</failed-calls-in>\n", profile->ib_failed_calls);
stream->write_function(stream, " <failed-calls-out>%u</failed-calls-out>\n", profile->ob_failed_calls);
stream->write_function(stream, " <registrations>%lu</registrations>\n", sofia_profile_reg_count(profile));
stream->write_function(stream, " </profile-info>\n");
}
stream->write_function(stream, " <registrations>\n");
cb.profile = profile;
cb.stream = stream;
@@ -3275,6 +3306,13 @@ static switch_status_t cmd_xml_status(char **argv, int argc, switch_stream_handl
"network_ip,network_port,sip_username,sip_realm,mwi_user,mwi_host"
" from sip_registrations where profile_name='%q' and contact like '%%%q%%'", profile->name, argv[3]);
}
if (!sql && argv[2] && !strcasecmp(argv[2], "reg")) {
sql = switch_mprintf("select call_id,sip_user,sip_host,contact,status,"
"rpid,expires,user_agent,server_user,server_host,profile_name,hostname,"
"network_ip,network_port,sip_username,sip_realm,mwi_user,mwi_host"
" from sip_registrations where profile_name='%q'", profile->name);
}
if (!sql && argv[2] && !strcasecmp(argv[2], "user") && argv[3]) {
char *dup = strdup(argv[3]);
char *host = NULL, *user = NULL;
@@ -3305,17 +3343,15 @@ static switch_status_t cmd_xml_status(char **argv, int argc, switch_stream_handl
switch_safe_free(sqlextra);
}
if (!sql) {
sql = switch_mprintf("select call_id,sip_user,sip_host,contact,status,"
"rpid,expires,user_agent,server_user,server_host,profile_name,hostname,"
"network_ip,network_port,sip_username,sip_realm,mwi_user,mwi_host"
" from sip_registrations where profile_name='%q'", profile->name);
if (sql) {
stream->write_function(stream, " <registrations>\n");
sofia_glue_execute_sql_callback(profile, profile->ireg_mutex, sql, show_reg_callback_xml, &cb);
switch_safe_free(sql);
stream->write_function(stream, " </registrations>\n");
}
sofia_glue_execute_sql_callback(profile, profile->ireg_mutex, sql, show_reg_callback_xml, &cb);
switch_safe_free(sql);
stream->write_function(stream, " </registrations>\n");
stream->write_function(stream, "</profile>\n");
sofia_glue_release_profile(profile);
@@ -3684,15 +3720,6 @@ static int contact_callback(void *pArg, int argc, char **argv, char **columnName
return 0;
}
static int sql2str_callback(void *pArg, int argc, char **argv, char **columnNames)
{
struct cb_helper_sql2str *cbt = (struct cb_helper_sql2str *) pArg;
switch_copy_string(cbt->buf, argv[0], cbt->len);
cbt->matches++;
return 0;
}
SWITCH_STANDARD_API(sofia_count_reg_function)
{
char *data;
@@ -4650,8 +4677,8 @@ static switch_call_cause_t sofia_outgoing_channel(switch_core_session_t *session
}
now = switch_epoch_time_now(NULL);
sql = switch_mprintf("insert into sip_dialogs (uuid,presence_id,presence_data,profile_name,hostname,rcd) "
"values ('%q', '%q', '%q', '%q', '%q', %ld)", switch_core_session_get_uuid(nsession),
sql = switch_mprintf("insert into sip_dialogs (uuid,presence_id,presence_data,profile_name,hostname,rcd,call_info_state) "
"values ('%q', '%q', '%q', '%q', '%q', %ld, '')", switch_core_session_get_uuid(nsession),
switch_str_nil(presence_id), switch_str_nil(presence_data), profile->name, mod_sofia_globals.hostname, (long) now);
sofia_glue_actually_execute_sql(profile, sql, profile->ireg_mutex);
switch_safe_free(sql);
@@ -5337,6 +5364,7 @@ SWITCH_MODULE_LOAD_FUNCTION(mod_sofia_load)
switch_chat_interface_t *chat_interface;
switch_api_interface_t *api_interface;
switch_management_interface_t *management_interface;
uint32_t cpus = switch_core_cpu_count();
struct in_addr in;
memset(&mod_sofia_globals, 0, sizeof(mod_sofia_globals));
@@ -5374,6 +5402,10 @@ SWITCH_MODULE_LOAD_FUNCTION(mod_sofia_load)
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Waiting for profiles to start\n");
switch_yield(1500000);
/* start one message thread per cpu */
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "Starting %u message threads.\n", cpus);
sofia_msg_thread_start(cpus);
if (switch_event_bind_removable(modname, SWITCH_EVENT_CUSTOM, MULTICAST_EVENT, event_handler, NULL,
&mod_sofia_globals.custom_node) != SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Couldn't bind!\n");
src/mod/endpoints/mod_sofia/mod_sofia.h
+3
View File
@@ -262,6 +262,7 @@ typedef enum {
PFLAG_SHUTDOWN,
PFLAG_PRESENCE_MAP,
PFLAG_OPTIONS_RESPOND_503_ON_BUSY,
PFLAG_PRESENCE_DISABLE_EARLY,
/* No new flags below this line */
PFLAG_MAX
} PFLAGS;
@@ -1116,6 +1117,7 @@ switch_status_t sofia_glue_send_notify(sofia_profile_t *profile, const char *use
const char *body, const char *o_contact, const char *network_ip);
char *sofia_glue_get_extra_headers(switch_channel_t *channel, const char *prefix);
void sofia_glue_set_extra_headers(switch_core_session_t *session, sip_t const *sip, const char *prefix);
char *sofia_glue_get_extra_headers_from_event(switch_event_t *event, const char *prefix);
void sofia_info_send_sipfrag(switch_core_session_t *aleg, switch_core_session_t *bleg);
void sofia_update_callee_id(switch_core_session_t *session, sofia_profile_t *profile, sip_t const *sip, switch_bool_t send);
void sofia_send_callee_id(switch_core_session_t *session, const char *name, const char *number);
@@ -1149,3 +1151,4 @@ void sofia_glue_pause_jitterbuffer(switch_core_session_t *session, switch_bool_t
void sofia_process_dispatch_event(sofia_dispatch_event_t **dep);
char *sofia_glue_get_host(const char *str, switch_memory_pool_t *pool);
void sofia_presence_check_subscriptions(sofia_profile_t *profile, time_t now);
void sofia_msg_thread_start(int idx);
src/mod/endpoints/mod_sofia/sofia.c
+220 -107
View File
@@ -88,9 +88,9 @@ void sofia_handle_sip_r_notify(switch_core_session_t *session, int status,
sofia_dispatch_event_t *de, tagi_t tags[])
{
if (status >= 300 && sip && sip->sip_call_id && (!sofia_private || !sofia_private->is_call)) {
if (status == 481 && sip && !sip->sip_retry_after && sip->sip_call_id && (!sofia_private || !sofia_private->is_call)) {
char *sql;
sql = switch_mprintf("delete from sip_subscriptions where call_id='%q'", sip->sip_call_id->i_id);
switch_assert(sql != NULL);
sofia_glue_execute_sql(profile, &sql, SWITCH_TRUE);
@@ -239,21 +239,25 @@ static void extract_header_vars(sofia_profile_t *profile, sip_t const *sip,
switch_stream_handle_t stream = { 0 };
int x = 0;
if (switch_channel_direction(channel) == SWITCH_CALL_DIRECTION_OUTBOUND) {
SWITCH_STANDARD_STREAM(stream);
SWITCH_STANDARD_STREAM(stream);
for(vp = sip->sip_via; vp; vp = vp->v_next) {
char *v = sip_header_as_string(nh->nh_home, (void *) vp);
stream.write_function(&stream, x == 0 ? "%s" : ",%s", v);
su_free(nh->nh_home, v);
x++;
}
for(vp = sip->sip_via; vp; vp = vp->v_next) {
char *v = sip_header_as_string(nh->nh_home, (void *) vp);
switch_channel_set_variable(channel, "sip_recover_via", (char *)stream.data);
free(stream.data);
stream.write_function(&stream, x == 0 ? "%s" : ",%s", v);
su_free(nh->nh_home, v);
x++;
}
switch_channel_set_variable(channel, "sip_full_via", (char *)stream.data);
if (switch_channel_direction(channel) == SWITCH_CALL_DIRECTION_OUTBOUND) {
switch_channel_set_variable(channel, "sip_recover_via", (char *)stream.data);
}
free(stream.data);
}
if (sip->sip_from) {
@@ -577,6 +581,14 @@ void sofia_handle_sip_i_bye(switch_core_session_t *session, int status,
sofia_set_flag_locked(tech_pvt, TFLAG_BYE);
call_info = switch_channel_get_variable(channel, "presence_call_info_full");
if (sip->sip_reason) {
char *reason_header = sip_header_as_string(nh->nh_home, (void *) sip->sip_reason);
if (!zstr(reason_header)) {
switch_channel_set_variable_partner(channel, "sip_reason", reason_header);
}
}
if (sip->sip_reason && sip->sip_reason->re_protocol && (!strcasecmp(sip->sip_reason->re_protocol, "Q.850")
|| !strcasecmp(sip->sip_reason->re_protocol, "FreeSWITCH")
|| !strcasecmp(sip->sip_reason->re_protocol, profile->username)) && sip->sip_reason->re_cause) {
@@ -977,7 +989,6 @@ static void our_sofia_event_callback(nua_event_t event,
case nua_r_unregister:
case nua_r_unsubscribe:
case nua_r_publish:
case nua_i_cancel:
case nua_i_error:
case nua_i_active:
case nua_i_terminated:
@@ -985,6 +996,19 @@ static void our_sofia_event_callback(nua_event_t event,
case nua_i_prack:
case nua_r_prack:
break;
case nua_i_cancel:
if (sip && channel && sip->sip_reason) {
char *reason_header = sip_header_as_string(nh->nh_home, (void *) sip->sip_reason);
if (!zstr(reason_header)) {
switch_channel_set_variable_partner(channel, "sip_reason", reason_header);
}
}
break;
case nua_r_cancel:
{
if (status > 299 && nh) {
@@ -1228,7 +1252,7 @@ void *SWITCH_THREAD_FUNC sofia_msg_thread_run(switch_thread_t *thread, void *obj
static int IDX = 0;
static void sofia_msg_thread_start(int idx)
void sofia_msg_thread_start(int idx)
{
if (idx >= SOFIA_MAX_MSG_QUEUE || (idx < mod_sofia_globals.msg_queue_len && mod_sofia_globals.msg_queue_thread[idx])) {
@@ -2618,8 +2642,9 @@ static void parse_gateways(sofia_profile_t *profile, switch_xml_t gateways_tag)
gateway->ping = switch_epoch_time_now(NULL) + ping_freq;
gateway->options_to_uri = switch_core_sprintf(gateway->pool, "<sip:%s;transport=%s>",
!zstr(from_domain) ? from_domain : proxy, register_transport);
gateway->options_from_uri = switch_core_sprintf(gateway->pool, "<sip:%s;transport=%s>",
profile->extrtpip ? profile->extrtpip : profile->sipip, register_transport);
//gateway->options_from_uri = switch_core_sprintf(gateway->pool, "<sip:%s;transport=%s>",
// profile->extrtpip ? profile->extrtpip : profile->sipip, register_transport);
gateway->options_from_uri = gateway->options_to_uri;
} else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "ERROR: invalid ping!\n");
}
@@ -3031,6 +3056,12 @@ switch_status_t reconfig_sofia(sofia_profile_t *profile)
} else {
sofia_clear_pflag(profile, PFLAG_T38_PASSTHRU);
}
} else if (!strcasecmp(var, "presence-disable-early")) {
if (switch_true(val)) {
sofia_set_pflag(profile, PFLAG_PRESENCE_DISABLE_EARLY);
} else {
sofia_clear_pflag(profile, PFLAG_PRESENCE_DISABLE_EARLY);
}
} else if (!strcasecmp(var, "ignore-183nosdp")) {
if (switch_true(val)) {
sofia_set_pflag(profile, PFLAG_IGNORE_183NOSDP);
@@ -3356,7 +3387,7 @@ switch_status_t reconfig_sofia(sofia_profile_t *profile)
} else if (!strcasecmp(var, "context")) {
profile->context = switch_core_strdup(profile->pool, val);
} else if (!strcasecmp(var, "local-network-acl")) {
if (!strcasecmp(var, "none")) {
if (!strcasecmp(val, "none")) {
profile->local_network = NULL;
} else {
profile->local_network = switch_core_strdup(profile->pool, val);
@@ -3605,6 +3636,18 @@ switch_status_t config_sofia(int reload, char *profile_name)
mod_sofia_globals.debug_sla = atoi(val);
} else if (!strcasecmp(var, "auto-restart")) {
mod_sofia_globals.auto_restart = switch_true(val);
} else if (!strcasecmp(var, "message-threads")) {
int num = atoi(val);
if (num < 1 || num > SOFIA_MAX_MSG_QUEUE - 1) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "message-threads must be between 1 and %d", SOFIA_MAX_MSG_QUEUE -1);
}
if (num < 1) num = 1;
if (num > SOFIA_MAX_MSG_QUEUE - 1) num = SOFIA_MAX_MSG_QUEUE -1;
sofia_msg_thread_start(num);
} else if (!strcasecmp(var, "reg-deny-binding-fetch-and-no-lookup")) { /* backwards compatibility */
mod_sofia_globals.reg_deny_binding_fetch_and_no_lookup = switch_true(val); /* remove when noone complains about the extra lookup */
if (switch_true(val)) {
@@ -3805,6 +3848,12 @@ switch_status_t config_sofia(int reload, char *profile_name)
} else {
sofia_clear_pflag(profile, PFLAG_T38_PASSTHRU);
}
} else if (!strcasecmp(var, "presence-disable-early")) {
if (switch_true(val)) {
sofia_set_pflag(profile, PFLAG_PRESENCE_DISABLE_EARLY);
} else {
sofia_clear_pflag(profile, PFLAG_PRESENCE_DISABLE_EARLY);
}
} else if (!strcasecmp(var, "ignore-183nosdp")) {
if (switch_true(val)) {
sofia_set_pflag(profile, PFLAG_IGNORE_183NOSDP);
@@ -3845,19 +3894,7 @@ switch_status_t config_sofia(int reload, char *profile_name)
} else {
sofia_clear_pflag(profile, PFLAG_CID_IN_1XX);
}
} else if (!strcasecmp(var, "message-threads")) {
int num = atoi(val);
if (num < 1 || num > SOFIA_MAX_MSG_QUEUE - 1) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "message-threads must be between 1 and %d", SOFIA_MAX_MSG_QUEUE -1);
}
if (num < 1) num = 1;
if (num > SOFIA_MAX_MSG_QUEUE - 1) num = SOFIA_MAX_MSG_QUEUE -1;
sofia_msg_thread_start(num);
} else if (!strcasecmp(var, "disable-hold")) {
if (switch_true(val)) {
sofia_set_pflag(profile, PFLAG_DISABLE_HOLD);
@@ -4719,7 +4756,7 @@ static void sofia_handle_sip_r_options(switch_core_session_t *session, int statu
gateway->ping = switch_epoch_time_now(NULL) + gateway->ping_freq;
sofia_reg_release_gateway(gateway);
gateway->pinging = 0;
} else if (sofia_test_pflag(profile, PFLAG_UNREG_OPTIONS_FAIL) && status != 200 && sip && sip->sip_to) {
} else if (sofia_test_pflag(profile, PFLAG_UNREG_OPTIONS_FAIL) && (status != 200 && status != 486) && sip && sip->sip_to) {
char *sql;
time_t now = switch_epoch_time_now(NULL);
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_WARNING, "Expire registration '%s@%s' due to options failure\n",
@@ -4757,6 +4794,13 @@ static void sofia_handle_sip_r_invite(switch_core_session_t *session, int status
return;
}
if (status >= 500 && sip->sip_reason && sip->sip_reason->re_protocol && (!strcasecmp(sip->sip_reason->re_protocol, "Q.850")
|| !strcasecmp(sip->sip_reason->re_protocol, "FreeSWITCH")
|| !strcasecmp(sip->sip_reason->re_protocol, profile->username)) && sip->sip_reason->re_cause) {
tech_pvt->q850_cause = atoi(sip->sip_reason->re_cause);
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "Remote Reason: %d\n", tech_pvt->q850_cause);
}
sofia_glue_get_addr(de->data->e_msg, network_ip, sizeof(network_ip), &network_port);
switch_channel_set_variable_printf(channel, "sip_local_network_addr", "%s", profile->extsipip ? profile->extsipip : profile->sipip);
@@ -4789,6 +4833,7 @@ static void sofia_handle_sip_r_invite(switch_core_session_t *session, int status
if (channel && sip->sip_call_info) {
char *p;
call_info = sip_header_as_string(nua_handle_home(nh), (void *) sip->sip_call_info);
if (switch_stristr("appearance", call_info)) {
switch_channel_set_variable(channel, "presence_call_info_full", call_info);
if ((p = strchr(call_info, ';'))) {
@@ -4825,10 +4870,15 @@ static void sofia_handle_sip_r_invite(switch_core_session_t *session, int status
if (mod_sofia_globals.debug_sla > 1) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "QUERY SQL %s\n", sql);
}
sofia_glue_execute_sql_now(profile, &sql, SWITCH_TRUE);
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_WARNING, "Auto-Fixing Broken SLA [<sip:%s>;%s]\n",
sip->sip_from->a_url->url_host, buf);
switch_channel_set_variable_printf(channel, "presence_call_info_full", "<sip:%s>;%s", sip->sip_from->a_url->url_host, buf);
switch_channel_set_variable(channel, "presence_call_info", buf);
}
}
}
}
}
@@ -5003,14 +5053,16 @@ static void sofia_handle_sip_r_invite(switch_core_session_t *session, int status
}
if (sofia_test_pflag(profile, PFLAG_MANUAL_REDIRECT)) {
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "Redirect: Transfering to %s %s %s\n",
p_contact->m_url->url_user, sip_redirect_dialplan, sip_redirect_context);
if (!(v = switch_channel_get_variable(channel, "outbound_redirect_info"))) {
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "Redirect: Transfering to %s %s %s\n",
p_contact->m_url->url_user, sip_redirect_dialplan, sip_redirect_context);
if (switch_true(switch_channel_get_variable(channel, "recording_follow_transfer"))) {
switch_core_media_bug_transfer_recordings(session, a_session);
if (switch_true(switch_channel_get_variable(channel, "recording_follow_transfer"))) {
switch_core_media_bug_transfer_recordings(session, a_session);
}
switch_ivr_session_transfer(a_session, p_contact->m_url->url_user, sip_redirect_dialplan, sip_redirect_context);
}
switch_ivr_session_transfer(a_session, p_contact->m_url->url_user, sip_redirect_dialplan, sip_redirect_context);
switch_channel_hangup(channel, SWITCH_CAUSE_REDIRECTION_TO_NEW_DESTINATION);
} else if ((!strcmp(profile->sipip, p_contact->m_url->url_host))
|| (profile->extsipip && !strcmp(profile->extsipip, p_contact->m_url->url_host))
@@ -5248,8 +5300,9 @@ static void sofia_handle_sip_r_invite(switch_core_session_t *session, int status
sql = switch_mprintf("insert into sip_dialogs "
"(call_id,uuid,sip_to_user,sip_to_host,sip_to_tag,sip_from_user,sip_from_host,sip_from_tag,contact_user,"
"contact_host,state,direction,user_agent,profile_name,hostname,contact,presence_id,presence_data,call_info,rcd) "
"values('%q','%q','%q','%q','%q','%q','%q','%q','%q','%q','%q','%q','%q','%q','%q','%q','%q','%q','%q',%ld)",
"contact_host,state,direction,user_agent,profile_name,hostname,contact,presence_id,presence_data,"
"call_info,rcd,call_info_state) "
"values('%q','%q','%q','%q','%q','%q','%q','%q','%q','%q','%q','%q','%q','%q','%q','%q','%q','%q','%q',%ld,'')",
call_id,
switch_core_session_get_uuid(session),
to_user, to_host, to_tag, from_user, from_host, from_tag, contact_user,
@@ -5672,11 +5725,13 @@ static void sofia_handle_sip_i_state(switch_core_session_t *session, int status,
const char *invite_route_uri = switch_channel_get_variable(tech_pvt->channel, "sip_invite_route_uri");
nua_ack(nh,
TAG_IF(invite_full_from, SIPTAG_FROM_STR(invite_full_from)),
TAG_IF(invite_full_to, SIPTAG_TO_STR(invite_full_to)),
TAG_IF(!zstr(invite_full_via), SIPTAG_VIA_STR(invite_full_via)),
TAG_IF(!zstr(invite_route_uri), SIPTAG_ROUTE_STR(invite_route_uri)),
TAG_END());
TAG_IF(invite_full_from, SIPTAG_FROM_STR(invite_full_from)),
TAG_IF(invite_full_to, SIPTAG_TO_STR(invite_full_to)),
TAG_IF(!zstr(tech_pvt->user_via), SIPTAG_VIA_STR(tech_pvt->user_via)),
TAG_IF((zstr(tech_pvt->user_via) && !zstr(invite_full_via)), SIPTAG_VIA_STR(invite_full_via)),
TAG_IF(!zstr(invite_route_uri), SIPTAG_ROUTE_STR(invite_route_uri)),
TAG_END());
} else {
nua_ack(nh,
@@ -5788,11 +5843,11 @@ static void sofia_handle_sip_i_state(switch_core_session_t *session, int status,
}
} else {
if (sofia_test_pflag(profile, PFLAG_3PCC)) {
if (switch_channel_test_flag(channel, CF_PROXY_MODE) || switch_channel_test_flag(channel, CF_PROXY_MEDIA)) {
if (switch_channel_test_flag(channel, CF_PROXY_MODE) || switch_channel_test_flag(channel, CF_PROXY_MEDIA)) {
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_INFO, "No SDP in INVITE and 3pcc=yes cannot work with bypass or proxy media, hanging up.\n");
switch_channel_set_variable(channel, SWITCH_ENDPOINT_DISPOSITION_VARIABLE, "3PCC DISABLED");
switch_channel_hangup(channel, SWITCH_CAUSE_MANDATORY_IE_MISSING);
} else {
} else {
switch_channel_set_variable(channel, SWITCH_ENDPOINT_DISPOSITION_VARIABLE, "RECEIVED_NOSDP");
sofia_glue_tech_choose_port(tech_pvt, 0);
sofia_glue_set_local_sdp(tech_pvt, NULL, 0, NULL, 0);
@@ -5812,23 +5867,23 @@ static void sofia_handle_sip_i_state(switch_core_session_t *session, int status,
SIPTAG_CONTENT_TYPE_STR("application/sdp"), SIPTAG_PAYLOAD_STR(tech_pvt->local_sdp_str), TAG_END());
}
}
} else if (sofia_test_pflag(profile, PFLAG_3PCC_PROXY)) {
//3PCC proxy mode delays the 200 OK until the call is answered
} else if (sofia_test_pflag(profile, PFLAG_3PCC_PROXY)) {
//3PCC proxy mode delays the 200 OK until the call is answered
// so can be made to work with bypass media as we have time to find out what the other end thinks codec offer should be...
switch_channel_set_variable(channel, SWITCH_ENDPOINT_DISPOSITION_VARIABLE, "RECEIVED_NOSDP");
sofia_set_flag_locked(tech_pvt, TFLAG_3PCC);
//sofia_glue_tech_choose_port(tech_pvt, 0);
//sofia_glue_set_local_sdp(tech_pvt, NULL, 0, NULL, 0);
sofia_set_flag(tech_pvt, TFLAG_LATE_NEGOTIATION);
//Moves into CS_INIT so call moves forward into the dialplan
switch_channel_set_state(channel, CS_INIT);
} else {
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_INFO, "No SDP in INVITE and 3pcc not enabled, hanging up.\n");
switch_channel_set_variable(channel, SWITCH_ENDPOINT_DISPOSITION_VARIABLE, "3PCC DISABLED");
switch_channel_hangup(channel, SWITCH_CAUSE_MANDATORY_IE_MISSING);
}
goto done;
switch_channel_set_variable(channel, SWITCH_ENDPOINT_DISPOSITION_VARIABLE, "RECEIVED_NOSDP");
sofia_set_flag_locked(tech_pvt, TFLAG_3PCC);
//sofia_glue_tech_choose_port(tech_pvt, 0);
//sofia_glue_set_local_sdp(tech_pvt, NULL, 0, NULL, 0);
sofia_set_flag(tech_pvt, TFLAG_LATE_NEGOTIATION);
//Moves into CS_INIT so call moves forward into the dialplan
switch_channel_set_state(channel, CS_INIT);
} else {
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_INFO, "No SDP in INVITE and 3pcc not enabled, hanging up.\n");
switch_channel_set_variable(channel, SWITCH_ENDPOINT_DISPOSITION_VARIABLE, "3PCC DISABLED");
switch_channel_hangup(channel, SWITCH_CAUSE_MANDATORY_IE_MISSING);
}
goto done;
}
} else if (tech_pvt && sofia_test_flag(tech_pvt, TFLAG_SDP) && !r_sdp) {
sofia_set_flag_locked(tech_pvt, TFLAG_NOSDP_REINVITE);
@@ -5883,53 +5938,56 @@ static void sofia_handle_sip_i_state(switch_core_session_t *session, int status,
switch_core_session_message_t *msg;
if (switch_channel_test_flag(channel, CF_PROXY_MODE) && !is_t38 && profile->media_options & MEDIA_OPT_MEDIA_ON_HOLD) {
tech_pvt->hold_laps = 1;
switch_channel_set_variable(channel, SWITCH_R_SDP_VARIABLE, r_sdp);
switch_channel_clear_flag(channel, CF_PROXY_MODE);
sofia_glue_tech_set_local_sdp(tech_pvt, NULL, SWITCH_FALSE);
if (switch_stristr("sendonly", r_sdp) || switch_stristr("0.0.0.0", r_sdp)) {
tech_pvt->hold_laps = 1;
switch_channel_set_variable(channel, SWITCH_R_SDP_VARIABLE, r_sdp);
switch_channel_clear_flag(channel, CF_PROXY_MODE);
sofia_glue_tech_set_local_sdp(tech_pvt, NULL, SWITCH_FALSE);
if (!switch_channel_media_ready(channel)) {
if (switch_channel_direction(tech_pvt->channel) == SWITCH_CALL_DIRECTION_INBOUND) {
//const char *r_sdp = switch_channel_get_variable(channel, SWITCH_R_SDP_VARIABLE);
if (!switch_channel_media_ready(channel)) {
if (switch_channel_direction(tech_pvt->channel) == SWITCH_CALL_DIRECTION_INBOUND) {
//const char *r_sdp = switch_channel_get_variable(channel, SWITCH_R_SDP_VARIABLE);
tech_pvt->num_codecs = 0;
tech_pvt->num_codecs = 0;
sofia_glue_tech_prepare_codecs(tech_pvt);
if (sofia_glue_tech_media(tech_pvt, r_sdp) != SWITCH_STATUS_SUCCESS) {
switch_channel_set_variable(channel, SWITCH_ENDPOINT_DISPOSITION_VARIABLE, "CODEC NEGOTIATION ERROR");
status = SWITCH_STATUS_FALSE;
switch_core_session_rwunlock(other_session);
goto done;
}
}
}
if (!switch_rtp_ready(tech_pvt->rtp_session)) {
sofia_glue_tech_prepare_codecs(tech_pvt);
if (sofia_glue_tech_media(tech_pvt, r_sdp) != SWITCH_STATUS_SUCCESS) {
switch_channel_set_variable(channel, SWITCH_ENDPOINT_DISPOSITION_VARIABLE, "CODEC NEGOTIATION ERROR");
status = SWITCH_STATUS_FALSE;
if ((status = sofia_glue_tech_choose_port(tech_pvt, 0)) != SWITCH_STATUS_SUCCESS) {
switch_channel_hangup(channel, SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER);
switch_core_session_rwunlock(other_session);
goto done;
}
}
}
sofia_glue_set_local_sdp(tech_pvt, NULL, 0, NULL, 1);
if (!switch_rtp_ready(tech_pvt->rtp_session)) {
sofia_glue_tech_prepare_codecs(tech_pvt);
if ((status = sofia_glue_tech_choose_port(tech_pvt, 0)) != SWITCH_STATUS_SUCCESS) {
switch_channel_hangup(channel, SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER);
switch_core_session_rwunlock(other_session);
goto done;
if (sofia_use_soa(tech_pvt)) {
nua_respond(tech_pvt->nh, SIP_200_OK,
SIPTAG_CONTACT_STR(tech_pvt->reply_contact),
SOATAG_USER_SDP_STR(tech_pvt->local_sdp_str),
SOATAG_REUSE_REJECTED(1),
SOATAG_ORDERED_USER(1), SOATAG_AUDIO_AUX("cn telephone-event"),
TAG_IF(sofia_test_pflag(profile, PFLAG_DISABLE_100REL), NUTAG_INCLUDE_EXTRA_SDP(1)), TAG_END());
} else {
nua_respond(tech_pvt->nh, SIP_200_OK,
NUTAG_MEDIA_ENABLE(0),
SIPTAG_CONTACT_STR(tech_pvt->reply_contact),
SIPTAG_CONTENT_TYPE_STR("application/sdp"), SIPTAG_PAYLOAD_STR(tech_pvt->local_sdp_str), TAG_END());
}
}
sofia_glue_set_local_sdp(tech_pvt, NULL, 0, NULL, 1);
launch_media_on_hold(session);
if (sofia_use_soa(tech_pvt)) {
nua_respond(tech_pvt->nh, SIP_200_OK,
SIPTAG_CONTACT_STR(tech_pvt->reply_contact),
SOATAG_USER_SDP_STR(tech_pvt->local_sdp_str),
SOATAG_REUSE_REJECTED(1),
SOATAG_ORDERED_USER(1), SOATAG_AUDIO_AUX("cn telephone-event"),
TAG_IF(sofia_test_pflag(profile, PFLAG_DISABLE_100REL), NUTAG_INCLUDE_EXTRA_SDP(1)), TAG_END());
} else {
nua_respond(tech_pvt->nh, SIP_200_OK,
NUTAG_MEDIA_ENABLE(0),
SIPTAG_CONTACT_STR(tech_pvt->reply_contact),
SIPTAG_CONTENT_TYPE_STR("application/sdp"), SIPTAG_PAYLOAD_STR(tech_pvt->local_sdp_str), TAG_END());
switch_core_session_rwunlock(other_session);
goto done;
}
launch_media_on_hold(session);
switch_core_session_rwunlock(other_session);
goto done;
}
if (switch_channel_test_flag(channel, CF_PROXY_MEDIA)) {
@@ -6212,6 +6270,8 @@ static void sofia_handle_sip_i_state(switch_core_session_t *session, int status,
sofia_set_flag_locked(tech_pvt, TFLAG_ANS);
if (match) {
switch_channel_check_zrtp(channel);
if (sofia_glue_tech_choose_port(tech_pvt, 0) == SWITCH_STATUS_SUCCESS) {
if (sofia_glue_activate_rtp(tech_pvt, 0) == SWITCH_STATUS_SUCCESS) {
switch_channel_mark_answered(channel);
@@ -8000,12 +8060,62 @@ void sofia_handle_sip_i_invite(nua_t *nua, sofia_profile_t *profile, nua_handle_
if ((call_info = sip_call_info(sip))) {
call_info_str = sip_header_as_string(nh->nh_home, (void *) call_info);
if (sofia_test_pflag(profile, PFLAG_MANAGE_SHARED_APPEARANCE) && switch_stristr("appearance", call_info_str)) {
char *p;
switch_channel_set_variable(channel, "presence_call_info_full", call_info_str);
if ((p = strchr(call_info_str, ';'))) {
p++;
switch_channel_set_variable(channel, "presence_call_info", p);
}
}
if (call_info->ci_params && (msg_params_find(call_info->ci_params, "answer-after=0"))) {
switch_channel_set_variable(channel, "sip_auto_answer_detected", "true");
}
switch_channel_set_variable(channel, "sip_call_info", call_info_str);
} else if (sofia_test_pflag(profile, PFLAG_MANAGE_SHARED_APPEARANCE)) {
char buf[128] = "";
char *sql;
char *state = "progressing";
if (sip &&
sip->sip_from && sip->sip_from->a_url && sip->sip_from->a_url->url_user && sip->sip_from->a_url->url_host &&
sip->sip_to && sip->sip_to->a_url && sip->sip_to->a_url->url_user && sip->sip_to->a_url->url_host) {
sql =
switch_mprintf("select 'appearance-index=1' from sip_subscriptions where expires > -1 and hostname='%q' and event='call-info' and "
"sub_to_user='%q' and sub_to_host='%q'", mod_sofia_globals.hostname, sip->sip_to->a_url->url_user,
sip->sip_from->a_url->url_host);
sofia_glue_execute_sql2str(profile, profile->ireg_mutex, sql, buf, sizeof(buf));
if (mod_sofia_globals.debug_sla > 1) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "QUERY SQL %s [%s]\n", sql, buf);
}
free(sql);
if (!zstr(buf)) {
sql = switch_mprintf("update sip_dialogs set call_info='%q',call_info_state='%q' "
"where uuid='%q'", buf, state, switch_core_session_get_uuid(session));
if (mod_sofia_globals.debug_sla > 1) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "QUERY SQL %s\n", sql);
}
sofia_glue_execute_sql_now(profile, &sql, SWITCH_TRUE);
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_WARNING, "Auto-Fixing Broken SLA [<sip:%s>;%s]\n",
sip->sip_from->a_url->url_host, buf);
switch_channel_set_variable_printf(channel, "presence_call_info_full", "<sip:%s>;%s", sip->sip_from->a_url->url_host, buf);
switch_channel_set_variable(channel, "presence_call_info", buf);
call_info_str = switch_core_session_sprintf(session, "<sip:%s>;%s", sip->sip_from->a_url->url_host, buf);
}
}
}
if (profile->pres_type) {
const char *presence_id = switch_channel_get_variable(channel, "presence_id");
if (zstr(presence_id)) {
@@ -8124,7 +8234,8 @@ void sofia_handle_sip_i_invite(nua_t *nua, sofia_profile_t *profile, nua_handle_
free(sql);
}
}
}
check_decode(displayname, session);
tech_pvt->caller_profile = switch_caller_profile_new(switch_core_session_get_pool(session),
@@ -8176,7 +8287,7 @@ void sofia_handle_sip_i_invite(nua_t *nua, sofia_profile_t *profile, nua_handle_
if (app && data && !strcasecmp(app, "conference")) {
tech_pvt->caller_profile->destination_number = switch_core_sprintf(tech_pvt->caller_profile->pool, "answer,conference:%s", data);
tech_pvt->caller_profile->destination_number = switch_core_sprintf(tech_pvt->caller_profile->pool, "answer,conference:%s+flags{dist-dtmf}", data);
tech_pvt->caller_profile->dialplan = "inline";
} else {
if (switch_core_session_check_interface(b_session, sofia_endpoint_interface)) {
@@ -8212,7 +8323,7 @@ void sofia_handle_sip_i_invite(nua_t *nua, sofia_profile_t *profile, nua_handle_
if (!one_leg &&
(!b_tech_pvt || !sofia_test_flag(b_tech_pvt, TFLAG_SIP_HOLD)) &&
(!c_tech_pvt || !sofia_test_flag(c_tech_pvt, TFLAG_SIP_HOLD))) {
char *ext = switch_core_session_sprintf(session, "answer,conference:%s@sla+flags{mintwo}", uuid);
char *ext = switch_core_session_sprintf(session, "answer,conference:%s@sla+flags{mintwo|dist-dtmf}", uuid);
switch_channel_set_flag(c_channel, CF_REDIRECT);
switch_ivr_session_transfer(b_session, ext, "inline", NULL);
@@ -8225,7 +8336,7 @@ void sofia_handle_sip_i_invite(nua_t *nua, sofia_profile_t *profile, nua_handle_
if (do_conf) {
tech_pvt->caller_profile->destination_number = switch_core_sprintf(tech_pvt->caller_profile->pool,
"answer,conference:%s@sla+flags{mintwo}", uuid);
"answer,conference:%s@sla+flags{mintwo|dist-dtmf}", uuid);
} else {
if (one_leg && c_app) {
if (c_data) {
@@ -8426,8 +8537,9 @@ void sofia_handle_sip_i_invite(nua_t *nua, sofia_profile_t *profile, nua_handle_
sql = switch_mprintf("insert into sip_dialogs "
"(call_id,uuid,sip_to_user,sip_to_host,sip_to_tag,sip_from_user,sip_from_host,sip_from_tag,contact_user,"
"contact_host,state,direction,user_agent,profile_name,hostname,contact,presence_id,presence_data,call_info,rcd) "
"values('%q','%q','%q','%q','%q','%q','%q','%q','%q','%q','%q','%q','%q','%q','%q','%q','%q','%q','%q',%ld)",
"contact_host,state,direction,user_agent,profile_name,hostname,contact,presence_id,presence_data,"
"call_info,rcd,call_info_state) "
"values('%q','%q','%q','%q','%q','%q','%q','%q','%q','%q','%q','%q','%q','%q','%q','%q','%q','%q','%q',%ld,'')",
call_id,
tech_pvt->sofia_private->uuid,
to_user, to_host, to_tag, dialog_from_user, dialog_from_host, from_tag,
@@ -8493,7 +8605,8 @@ void sofia_handle_sip_i_options(int status,
(sess_count >= sess_max || !sofia_test_pflag(profile, PFLAG_RUNNING) || !switch_core_ready_inbound())) {
nua_respond(nh, 503, "Maximum Calls In Progress", NUTAG_WITH_THIS_MSG(de->data->e_msg), SIPTAG_RETRY_AFTER_STR("300"), TAG_END());
} else {
nua_respond(nh, SIP_200_OK, NUTAG_WITH_THIS_MSG(de->data->e_msg), TAG_END());
nua_respond(nh, SIP_200_OK, NUTAG_WITH_THIS_MSG(de->data->e_msg),
TAG_IF(sip->sip_record_route, SIPTAG_RECORD_ROUTE(sip->sip_record_route)), TAG_END());
}
}
src/mod/endpoints/mod_sofia/sofia_glue.c
+154 -43
View File
@@ -44,10 +44,10 @@ void sofia_glue_set_image_sdp(private_object_t *tech_pvt, switch_t38_options_t *
char buf[2048] = "";
char max_buf[128] = "";
char max_data[128] = "";
const char *ip = t38_options->local_ip;
uint32_t port = t38_options->local_port;
const char *ip;
uint32_t port;
const char *family = "IP4";
const char *username = tech_pvt->profile->username;
const char *username;
const char *bit_removal_on = "a=T38FaxFillBitRemoval\n";
const char *bit_removal_off = "";
@@ -59,6 +59,13 @@ void sofia_glue_set_image_sdp(private_object_t *tech_pvt, switch_t38_options_t *
const char *var;
int broken_boolean;
switch_assert(tech_pvt);
switch_assert(t38_options);
ip = t38_options->local_ip;
port = t38_options->local_port;
username = tech_pvt->profile->username;
//sofia_clear_flag(tech_pvt, TFLAG_ENABLE_SOA);
var = switch_channel_get_variable(tech_pvt->channel, "t38_broken_boolean");
@@ -177,7 +184,8 @@ static void generate_m(private_object_t *tech_pvt, char *buf, size_t buflen,
int rate;
int already_did[128] = { 0 };
int ptime = 0, noptime = 0;
const char *zrtp;
switch_snprintf(buf + strlen(buf), buflen - strlen(buf), "m=audio %d RTP/%sAVP",
port, secure ? "S" : "");
@@ -336,6 +344,10 @@ static void generate_m(private_object_t *tech_pvt, char *buf, size_t buflen,
switch_snprintf(buf + strlen(buf), buflen - strlen(buf), "a=ptime:%d\n", cur_ptime);
}
if ((zrtp = switch_channel_get_variable(tech_pvt->channel, "sdp_zrtp_hash_string"))) {
switch_snprintf(buf + strlen(buf), buflen - strlen(buf), "a=zrtp-hash:%s\n", zrtp);
}
if (sr) {
switch_snprintf(buf + strlen(buf), buflen - strlen(buf), "a=%s\n", sr);
}
@@ -379,6 +391,7 @@ void sofia_glue_set_local_sdp(private_object_t *tech_pvt, const char *ip, switch
switch_event_t *map = NULL, *ptmap = NULL;
const char *b_sdp = NULL;
int verbose_sdp = 0;
const char *zrtp;
sofia_glue_check_dtmf_type(tech_pvt);
@@ -532,6 +545,10 @@ void sofia_glue_set_local_sdp(private_object_t *tech_pvt, const char *ip, switch
switch_snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "a=ptime:%d\n", ptime);
}
if ((zrtp = switch_channel_get_variable(tech_pvt->channel, "sdp_zrtp_hash_string"))) {
switch_snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "a=zrtp-hash:%s\n", zrtp);
}
if (sr) {
switch_snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "a=%s\n", sr);
}
@@ -558,13 +575,22 @@ void sofia_glue_set_local_sdp(private_object_t *tech_pvt, const char *ip, switch
if (mult && switch_false(mult)) {
char *bp = buf;
int both = 1;
if ((!zstr(tech_pvt->local_crypto_key) && sofia_test_flag(tech_pvt, TFLAG_SECURE))) {
generate_m(tech_pvt, buf, sizeof(buf), port, 0, append_audio, sr, use_cng, cng_type, map, verbose_sdp, 1);
bp = (buf + strlen(buf));
/* asterisk can't handle AVP and SAVP in sep streams, way to blow off the spec....*/
if (switch_true(switch_channel_get_variable(tech_pvt->channel, "sdp_secure_savp_only"))) {
both = 0;
}
}
generate_m(tech_pvt, bp, sizeof(buf) - strlen(buf), port, 0, append_audio, sr, use_cng, cng_type, map, verbose_sdp, 0);
if (both) {
generate_m(tech_pvt, bp, sizeof(buf) - strlen(buf), port, 0, append_audio, sr, use_cng, cng_type, map, verbose_sdp, 0);
}
} else {
@@ -583,14 +609,23 @@ void sofia_glue_set_local_sdp(private_object_t *tech_pvt, const char *ip, switch
if (cur_ptime != this_ptime) {
char *bp = buf;
cur_ptime = this_ptime;
int both = 1;
cur_ptime = this_ptime;
if ((!zstr(tech_pvt->local_crypto_key) && sofia_test_flag(tech_pvt, TFLAG_SECURE))) {
generate_m(tech_pvt, buf, sizeof(buf), port, cur_ptime, append_audio, sr, use_cng, cng_type, map, verbose_sdp, 1);
bp = (buf + strlen(buf));
/* asterisk can't handle AVP and SAVP in sep streams, way to blow off the spec....*/
if (switch_true(switch_channel_get_variable(tech_pvt->channel, "sdp_secure_savp_only"))) {
both = 0;
}
}
if (both) {
generate_m(tech_pvt, bp, sizeof(buf) - strlen(buf), port, cur_ptime, append_audio, sr, use_cng, cng_type, map, verbose_sdp, 0);
}
generate_m(tech_pvt, bp, sizeof(buf) - strlen(buf), port, cur_ptime, append_audio, sr, use_cng, cng_type, map, verbose_sdp, 0);
}
}
@@ -762,6 +797,11 @@ void sofia_glue_tech_prepare_codecs(private_object_t *tech_pvt)
switch_assert(tech_pvt->session != NULL);
if ((abs = switch_channel_get_variable(tech_pvt->channel, "absolute_codec_string"))) {
/* inherit_codec == true will implicitly clear the absolute_codec_string
variable if used since it was the reason it was set in the first place and is no longer needed */
if (switch_true(switch_channel_get_variable(tech_pvt->channel, "inherit_codec"))) {
switch_channel_set_variable(tech_pvt->channel, "absolute_codec_string", NULL);
}
codec_string = abs;
goto ready;
}
@@ -1968,6 +2008,30 @@ void sofia_glue_set_extra_headers(switch_core_session_t *session, sip_t const *s
switch_channel_api_on(channel, "api_on_sip_extra_headers");
}
char *sofia_glue_get_extra_headers_from_event(switch_event_t *event, const char *prefix)
{
char *extra_headers = NULL;
switch_stream_handle_t stream = { 0 };
switch_event_header_t *hp;
SWITCH_STANDARD_STREAM(stream);
for (hp = event->headers; hp; hp = hp->next) {
if (!zstr(hp->name) && !zstr(hp->value) && !strncasecmp(hp->name, prefix, strlen(prefix))) {
char *name = strdup(hp->name);
const char *hname = name + strlen(prefix);
stream.write_function(&stream, "%s: %s\r\n", hname, (char *)hp->value);
free(name);
}
}
if (!zstr((char *) stream.data)) {
extra_headers = stream.data;
} else {
switch_safe_free(stream.data);
}
return extra_headers;
}
switch_status_t sofia_glue_do_invite(switch_core_session_t *session)
{
@@ -2291,6 +2355,29 @@ switch_status_t sofia_glue_do_invite(switch_core_session_t *session)
switch_channel_set_variable(channel, "sip_to_host", sofia_glue_get_host(to_str, switch_core_session_get_pool(session)));
switch_channel_set_variable(channel, "sip_from_host", sofia_glue_get_host(from_str, switch_core_session_get_pool(session)));
if (!switch_channel_get_variable(channel, "presence_id")) {
char *from = switch_core_session_strdup(session, from_str);
const char *s;
if ((s = switch_stristr("<", from))) {
from = (char *)s + 1;
}
if (!strncasecmp(from, "sip:", 4)) {
from += 4;
}
if (!strncasecmp(from, "sips:", 5)) {
from += 5;
}
if ((p = strchr(from, ':')) || (p = strchr(from, ';')) || (p = strchr(from, '>'))) {
*p++ = '\0';
}
switch_channel_set_variable(channel, "presence_id", from);
}
if (!(tech_pvt->nh = nua_handle(tech_pvt->profile->nua, NULL,
NUTAG_URL(url_str),
TAG_IF(call_id, SIPTAG_CALL_ID_STR(call_id)),
@@ -2352,11 +2439,22 @@ switch_status_t sofia_glue_do_invite(switch_core_session_t *session)
switch (cid_type) {
case CID_TYPE_PID:
if (switch_test_flag(caller_profile, SWITCH_CPF_SCREEN)) {
tech_pvt->asserted_id = switch_core_session_sprintf(tech_pvt->session, "\"%s\"<sip:%s@%s>", use_name, use_number, rpid_domain);
if (zstr(tech_pvt->caller_profile->caller_id_name) || !strcasecmp(tech_pvt->caller_profile->caller_id_name, "_undef_")) {
tech_pvt->asserted_id = switch_core_session_sprintf(tech_pvt->session, "<sip:%s@%s>",
use_number, rpid_domain);
} else {
tech_pvt->asserted_id = switch_core_session_sprintf(tech_pvt->session, "\"%s\"<sip:%s@%s>",
use_name, use_number, rpid_domain);
}
} else {
tech_pvt->preferred_id = switch_core_session_sprintf(tech_pvt->session, "\"%s\"<sip:%s@%s>",
tech_pvt->caller_profile->caller_id_name,
tech_pvt->caller_profile->caller_id_number, rpid_domain);
if (zstr(tech_pvt->caller_profile->caller_id_name) || !strcasecmp(tech_pvt->caller_profile->caller_id_name, "_undef_")) {
tech_pvt->preferred_id = switch_core_session_sprintf(tech_pvt->session, "<sip:%s@%s>",
tech_pvt->caller_profile->caller_id_number, rpid_domain);
} else {
tech_pvt->preferred_id = switch_core_session_sprintf(tech_pvt->session, "\"%s\"<sip:%s@%s>",
tech_pvt->caller_profile->caller_id_name,
tech_pvt->caller_profile->caller_id_number, rpid_domain);
}
}
if (switch_test_flag(caller_profile, SWITCH_CPF_HIDE_NUMBER)) {
@@ -2503,7 +2601,7 @@ switch_status_t sofia_glue_do_invite(switch_core_session_t *session)
} else {
tech_pvt->session_refresher = nua_no_refresher;
}
if (sofia_use_soa(tech_pvt)) {
nua_invite(tech_pvt->nh,
NUTAG_AUTOANSWER(0),
@@ -2517,8 +2615,8 @@ switch_status_t sofia_glue_do_invite(switch_core_session_t *session)
TAG_IF(invite_full_from, SIPTAG_FROM_STR(invite_full_from)),
TAG_IF(invite_full_to, SIPTAG_TO_STR(invite_full_to)),
TAG_IF(tech_pvt->redirected, NUTAG_URL(tech_pvt->redirected)),
TAG_IF(!zstr(tech_pvt->user_via), SIPTAG_VIA_STR(tech_pvt->user_via)),
TAG_IF(!zstr(recover_via), SIPTAG_VIA_STR(recover_via)),
TAG_IF(!zstr(tech_pvt->user_via), SIPTAG_VIA_STR(tech_pvt->user_via)),
TAG_IF(!zstr(tech_pvt->rpid), SIPTAG_REMOTE_PARTY_ID_STR(tech_pvt->rpid)),
TAG_IF(!zstr(tech_pvt->preferred_id), SIPTAG_P_PREFERRED_IDENTITY_STR(tech_pvt->preferred_id)),
TAG_IF(!zstr(tech_pvt->asserted_id), SIPTAG_P_ASSERTED_IDENTITY_STR(tech_pvt->asserted_id)),
@@ -2551,8 +2649,8 @@ switch_status_t sofia_glue_do_invite(switch_core_session_t *session)
TAG_IF(invite_full_from, SIPTAG_FROM_STR(invite_full_from)),
TAG_IF(invite_full_to, SIPTAG_TO_STR(invite_full_to)),
TAG_IF(tech_pvt->redirected, NUTAG_URL(tech_pvt->redirected)),
TAG_IF(!zstr(tech_pvt->user_via), SIPTAG_VIA_STR(tech_pvt->user_via)),
TAG_IF(!zstr(recover_via), SIPTAG_VIA_STR(recover_via)),
TAG_IF(!zstr(tech_pvt->user_via), SIPTAG_VIA_STR(tech_pvt->user_via)),
TAG_IF(!zstr(tech_pvt->rpid), SIPTAG_REMOTE_PARTY_ID_STR(tech_pvt->rpid)),
TAG_IF(!zstr(tech_pvt->preferred_id), SIPTAG_P_PREFERRED_IDENTITY_STR(tech_pvt->preferred_id)),
TAG_IF(!zstr(tech_pvt->asserted_id), SIPTAG_P_ASSERTED_IDENTITY_STR(tech_pvt->asserted_id)),
@@ -3475,6 +3573,14 @@ switch_status_t sofia_glue_activate_rtp(private_object_t *tech_pvt, switch_rtp_f
switch_channel_set_variable(tech_pvt->channel, SWITCH_REMOTE_MEDIA_IP_VARIABLE, tech_pvt->remote_sdp_audio_ip);
switch_channel_set_variable(tech_pvt->channel, SWITCH_REMOTE_MEDIA_PORT_VARIABLE, tmp);
if (switch_channel_test_flag(tech_pvt->channel, CF_ZRTP_PASS)) {
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(tech_pvt->session), SWITCH_LOG_INFO, "Activating ZRTP PROXY MODE\n");
sofia_clear_flag(tech_pvt, TFLAG_NOTIMER_DURING_BRIDGE);
switch_rtp_udptl_mode(tech_pvt->rtp_session);
}
video:
sofia_glue_check_video_codecs(tech_pvt);
@@ -3969,7 +4075,6 @@ int sofia_glue_toggle_hold(private_object_t *tech_pvt, int sendonly)
}
sofia_set_flag_locked(tech_pvt, TFLAG_SIP_HOLD);
switch_channel_set_flag(tech_pvt->channel, CF_LEG_HOLDING);
switch_channel_mark_hold(tech_pvt->channel, SWITCH_TRUE);
switch_channel_presence(tech_pvt->channel, "unknown", msg, NULL);
changed = 1;
@@ -3997,7 +4102,6 @@ int sofia_glue_toggle_hold(private_object_t *tech_pvt, int sendonly)
} else {
if (sofia_test_flag(tech_pvt, TFLAG_HOLD_LOCK)) {
sofia_set_flag(tech_pvt, TFLAG_SIP_HOLD);
switch_channel_set_flag(tech_pvt->channel, CF_LEG_HOLDING);
switch_channel_mark_hold(tech_pvt->channel, SWITCH_TRUE);
changed = 1;
}
@@ -4030,7 +4134,6 @@ int sofia_glue_toggle_hold(private_object_t *tech_pvt, int sendonly)
}
sofia_clear_flag_locked(tech_pvt, TFLAG_SIP_HOLD);
switch_channel_clear_flag(tech_pvt->channel, CF_LEG_HOLDING);
switch_channel_mark_hold(tech_pvt->channel, SWITCH_FALSE);
switch_channel_presence(tech_pvt->channel, "unknown", "unhold", NULL);
changed = 1;
@@ -4398,9 +4501,12 @@ uint8_t sofia_glue_negotiate_sdp(switch_core_session_t *session, const char *r_s
continue;
}
if (!strcasecmp(attr->a_name, "sendonly") || !strcasecmp(attr->a_name, "inactive")) {
if (!strcasecmp(attr->a_name, "sendonly")) {
sendonly = 1;
switch_channel_set_variable(tech_pvt->channel, "media_audio_mode", "recvonly");
} else if (!strcasecmp(attr->a_name, "inactive")) {
sendonly = 1;
switch_channel_set_variable(tech_pvt->channel, "media_audio_mode", "inactive");
} else if (!strcasecmp(attr->a_name, "recvonly")) {
switch_channel_set_variable(tech_pvt->channel, "media_audio_mode", "sendonly");
recvonly = 1;
@@ -4580,12 +4686,13 @@ uint8_t sofia_glue_negotiate_sdp(switch_core_session_t *session, const char *r_s
if (!strcasecmp(attr->a_name, "rtcp") && attr->a_value) {
switch_channel_set_variable(tech_pvt->channel, "sip_remote_audio_rtcp_port", attr->a_value);
}
if (!strcasecmp(attr->a_name, "ptime") && attr->a_value) {
} else if (!strcasecmp(attr->a_name, "ptime") && attr->a_value) {
ptime = atoi(attr->a_value);
} else if (!strcasecmp(attr->a_name, "maxptime") && attr->a_value) {
maxptime = atoi(attr->a_value);
} else if (!strcasecmp(attr->a_name, "zrtp-hash") && attr->a_value) {
switch_channel_set_variable(tech_pvt->channel, "sdp_zrtp_hash_string", attr->a_value);
switch_channel_set_flag(tech_pvt->channel, CF_ZRTP_HASH);
} else if (!got_crypto && !strcasecmp(attr->a_name, "crypto") && !zstr(attr->a_value)) {
int crypto_tag;
@@ -4599,7 +4706,8 @@ uint8_t sofia_glue_negotiate_sdp(switch_core_session_t *session, const char *r_s
crypto_tag = atoi(crypto);
if (tech_pvt->remote_crypto_key && switch_rtp_ready(tech_pvt->rtp_session)) {
if (crypto_tag && crypto_tag == tech_pvt->crypto_tag) {
/* Compare all the key. The tag may remain the same even if key changed */
if (crypto && !strcmp(crypto, tech_pvt->remote_crypto_key)) {
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "Existing key is still valid.\n");
} else {
const char *a = switch_stristr("AES", tech_pvt->remote_crypto_key);
@@ -5139,6 +5247,8 @@ char *sofia_glue_get_url_from_contact(char *buf, uint8_t to_dup)
{
char *url = NULL, *e;
switch_assert(buf);
while(*buf == ' ') {
buf++;
}
@@ -5956,7 +6066,7 @@ int sofia_glue_init_sql(sofia_profile_t *profile)
" presence_id VARCHAR(255),\n"
" presence_data VARCHAR(255),\n"
" call_info VARCHAR(255),\n"
" call_info_state VARCHAR(255),\n"
" call_info_state VARCHAR(255) default '',\n"
" expires INTEGER default 0,\n"
" status VARCHAR(255),\n"
" rpid VARCHAR(255),\n"
@@ -6868,24 +6978,6 @@ char *sofia_glue_gen_contact_str(sofia_profile_t *profile, sip_t const *sip, nua
np->is_nat = NULL;
}
if (zstr(contact_host)) {
np->is_nat = "No contact host";
}
if (np->is_nat && !np->fs_path) {
contact_host = np->network_ip;
switch_snprintf(new_port, sizeof(new_port), ":%d", np->network_port);
port = NULL;
}
if (port) {
switch_snprintf(new_port, sizeof(new_port), ":%s", port);
}
ipv6 = strchr(contact_host, ':');
if (np->is_nat && np->fs_path) {
char *full_contact = sip_header_as_string(nh->nh_home, (void *) contact);
char *full_contact_dup;
@@ -6918,6 +7010,25 @@ char *sofia_glue_gen_contact_str(sofia_profile_t *profile, sip_t const *sip, nua
free(path_val);
} else {
if (zstr(contact_host)) {
np->is_nat = "No contact host";
}
if (np->is_nat) {
contact_host = np->network_ip;
switch_snprintf(new_port, sizeof(new_port), ":%d", np->network_port);
port = NULL;
}
if (port) {
switch_snprintf(new_port, sizeof(new_port), ":%s", port);
}
ipv6 = strchr(contact_host, ':');
if (contact->m_url->url_params) {
contact_str = switch_mprintf("%s <sip:%s@%s%s%s%s;%s>%s",
display, contact->m_url->url_user,

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