Diff
Modified: freeswitch/trunk/contrib/jmesquita/FSPhone/FSPhone.pro (16121 => 16122)
--- freeswitch/trunk/contrib/jmesquita/FSPhone/FSPhone.pro 2010-01-02 18:37:31 UTC (rev 16121)
+++ freeswitch/trunk/contrib/jmesquita/FSPhone/FSPhone.pro 2010-01-02 22:06:42 UTC (rev 16122)
@@ -30,3 +30,4 @@
call.h \
mod_qsettings/mod_qsettings.h
FORMS += mainwindow.ui
+RESOURCES += resources.qrc
Added: freeswitch/trunk/contrib/jmesquita/FSPhone/conf/accounts/example.xml (0 => 16122)
--- freeswitch/trunk/contrib/jmesquita/FSPhone/conf/accounts/example.xml (rev 0)
+++ freeswitch/trunk/contrib/jmesquita/FSPhone/conf/accounts/example.xml 2010-01-02 22:06:42 UTC (rev 16122)
@@ -0,0 +1,34 @@
+<include>
+ <gateway name="voicetechnology.com.ar">
+ <!--/// account username *required* ///-->
+ <param name="username" value="1001"/>
+ <!--/// 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="cerrito2p"/>
+ <!--/// 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>
+</include>
Added: freeswitch/trunk/contrib/jmesquita/FSPhone/conf/freeswitch.serial
(Binary files differ)
Added: freeswitch/trunk/contrib/jmesquita/FSPhone/conf/freeswitch.xml (0 => 16122)
--- freeswitch/trunk/contrib/jmesquita/FSPhone/conf/freeswitch.xml (rev 0)
+++ freeswitch/trunk/contrib/jmesquita/FSPhone/conf/freeswitch.xml 2010-01-02 22:06:42 UTC (rev 16122)
@@ -0,0 +1,292 @@
+<?xml version="1.0"?>
+<document type="freeswitch/xml">
+ <X-PRE-PROCESS cmd="set" data="auto_answer=false"/>
+ <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>
+ <X-PRE-PROCESS cmd="set" data="hold_music=local_stream://moh"/>
+ <X-PRE-PROCESS cmd="set" data="codec_prefs=CELT@48000h,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/>
+ <X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org"/>
+ <X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org"/>
+ <X-PRE-PROCESS cmd="set" data="outbound_caller_name=FreeSWITCH"/>
+ <X-PRE-PROCESS cmd="set" data="outbound_caller_id=0000000000"/>
+ <X-PRE-PROCESS cmd="set" data="console_loglevel=info"/>
+ <X-PRE-PROCESS cmd="set" data="default_gateway=voicetechnology.com.ar"/>
+ <X-PRE-PROCESS cmd="set" data="us-ring=%(2000, 4000, 440.0, 480.0)"/>
+ <X-PRE-PROCESS cmd="set" data="bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1400,0,350,440)"/>
+ <X-PRE-PROCESS cmd="set" data="sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7)"/>
+
+ <section name="configuration" description="Various Configuration">
+ <configuration name="cdr_csv.conf" description="CDR CSV Format">
+ <settings>
+ <param name="default-template" value="example"/>
+ <param name="rotate-on-hup" value="true"/>
+ <param name="legs" value="a"/>
+ </settings>
+ <templates>
+ <template name="example">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}","${accountcode}","${read_codec}","${write_codec}"</template>
+ </templates>
+ </configuration>
+
+ <configuration name="console.conf" description="Console Logger">
+ <mappings>
+ <map name="all" value="console,debug,info,notice,warning,err,crit,alert"/>
+ </mappings>
+ <settings>
+ <param name="colorize" value="true"/>
+ <param name="loglevel" value="$${console_loglevel}"/>
+ </settings>
+ </configuration>
+
+ <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/softphone/$1"/>
+ </routes>
+ </configuration>
+
+ <configuration name="local_stream.conf" description="stream files from local dir">
+ <directory name="moh/48000" path="$${base_dir}/sounds/music/48000">
+ <param name="rate" value="48000"/>
+ <param name="shuffle" value="true"/>
+ <param name="channels" value="1"/>
+ <param name="interval" value="10"/>
+ <param name="timer-name" value="soft"/>
+ </directory>
+ </configuration>
+
+ <configuration name="logfile.conf" description="File Logging">
+ <settings>
+ <param name="rotate-on-hup" value="true"/>
+ </settings>
+ <profiles>
+ <profile name="default">
+ <settings>
+ </settings>
+ <mappings>
+ <map name="all" value="debug,info,notice,warning,err,crit,alert"/>
+ </mappings>
+ </profile>
+ </profiles>
+ </configuration>
+
+ <configuration name="modules.conf" description="Modules">
+ <modules>
+ <load module="mod_console"/>
+ <load module="mod_logfile"/>
+ <load module="mod_enum"/>
+ <load module="mod_cdr_csv"/>
+ <load module="mod_portaudio"/>
+ <load module="mod_sofia"/>
+ <load module="mod_loopback"/>
+ <load module="mod_commands"/>
+ <load module="mod_dptools"/>
+ <load module="mod_dialplan_xml"/>
+ <load module="mod_voipcodecs"/>
+ <load module="mod_ilbc"/>
+ <load module="mod_speex"/>
+ <load module="mod_celt"/>
+ <load module="mod_siren"/>
+ <load module="mod_sndfile"/>
+ <load module="mod_tone_stream"/>
+ <load module="mod_local_stream"/>
+ <load module="mod_event_socket"/>
+ </modules>
+ </configuration>
+
+ <configuration name="portaudio.conf" description="Soundcard Endpoint">
+ <settings>
+ <param name="indev" value=""/>
+ <!-- device to use for output -->
+ <param name="outdev" value=""/>
+ <!--<param name="ringdev" value=""/>-->
+ <param name="ring-file" value="tone_stream://%(2000,4000,440.0,480.0);loops=20"/>
+ <param name="ring-interval" value="5"/>
+ <param name="hold-file" value="$${hold_music}"/>
+ <!--<param name="timer-name" value="soft"/>-->
+ <param name="dialplan" value="XML"/>
+ <param name="cid-name" value="$${outbound_caller_name}"/>
+ <param name="cid-num" value="$${outbound_caller_number}"/>
+ <param name="sample-rate" value="48000"/>
+ <param name="codec-ms" value="10"/>
+ </settings>
+ </configuration>
+
+ <configuration name="event_socket.conf" description="Socket Client">
+ <settings>
+ <param name="nat-map" value="false"/>
+ <param name="listen-ip" value="127.0.0.1"/>
+ <param name="listen-port" value="8021"/>
+ <param name="password" value="ClueCon"/>
+ <!--<param name="apply-inbound-acl" value="lan"/>-->
+ </settings>
+ </configuration>
+
+ <configuration name="post_load_modules.conf" description="Modules">
+ <modules>
+ </modules>
+ </configuration>
+
+ <configuration name="sofia.conf" description="sofia Endpoint">
+ <global_settings>
+ <param name="log-level" value="0"/>
+ <param name="auto-restart" value="true"/>
+ <param name="debug-presence" value="0"/>
+ </global_settings>
+ <profiles>
+ <profile name="softphone">
+ <gateways>
+ <X-PRE-PROCESS cmd="include" data="accounts/*.xml"/>
+ </gateways>
+ <settings>
+ <!--<param name="media-option" value="bypass-media-after-att-xfer"/>-->
+ <param name="user-agent-string" value="FreeSWITCH/SoftPhone"/>
+ <!-- <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="12345"/>
+ <param name="dialplan" value="XML"/>
+ <param name="dtmf-duration" value="100"/>
+ <param name="codec-prefs" value="$${codec_prefs}"/>
+ <param name="use-rtp-timer" value="true"/>
+ <param name="rtp-timer-name" value="soft"/>
+ <!-- ip address to use for rtp, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
+ <param name="rtp-ip" value="$${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="rfc1918"/>
+ <!--<param name="enable-timer" value="false"/>-->
+ <!--<param name="enable-100rel" value="true"/>-->
+ <!--<param name="minimum-session-expires" value="120"/>-->
+ <!--<param name="dtmf-type" value="info"/>-->
+ <param name="manage-presence" value="false"/>
+ <!--<param name="bitpacking" value="aal2"/> -->
+ <param name="max-proceeding" value="3"/>
+ <!--<param name="session-timeout" value="120"/>-->
+ <!--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"/>-->
+ <!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok-->
+ <!--<param name="inbound-late-negotiation" value="true"/>-->
+ <!--<param name="accept-blind-reg" value="true"/> -->
+ <!--<param name="accept-blind-auth" value="true"/> -->
+ <!--<param name="suppress-cng" value="true"/> -->
+ <param name="nonce-ttl" value="60"/>
+ <!--<param name="NDLB-broken-auth-hash" value="true"/>-->
+ <!--<param name="NDLB-received-in-nat-reg-contact" value="true"/>-->
+ <param name="auth-calls" value="false"/>
+ <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="disable-register" value="true"/>
+ <!--<param name="NDLB-force-rport" value="true"/>-->
+ <param name="challenge-realm" value="auto_from"/>
+ <!--<param name="disable-rtp-auto-adjust" value="true"/>-->
+ <!--<param name="inbound-use-callid-as-uuid" value="true"/>-->
+ <!--<param name="outbound-use-uuid-as-callid" value="true"/>-->
+ <!--<param name="auto-rtp-bugs" data="clear"/>-->
+ </settings>
+ </profile>
+ </profiles>
+ </configuration>
+
+ <configuration name="switch.conf" description="Core Configuration">
+ <cli-keybindings>
+ <key name="1" value="help"/>
+ <key name="2" value="status"/>
+ <key name="3" value="pa answer"/>
+ <key name="4" value="pa hangup"/>
+ <key name="5" value="sofia status"/>
+ <key name="6" value="reloadxml"/>
+ <key name="7" value="console loglevel 0"/>
+ <key name="8" value="console loglevel 7"/>
+ <key name="9" value="sofia status profile softphone"/>
+ <key name="10" value="fsctl pause"/>
+ <key name="11" value="fsctl resume"/>
+ <key name="12" value="version"/>
+ </cli-keybindings>
+ <settings>
+ <param name="colorize-console" value="true"/>
+ <param name="max-sessions" value="20"/>
+ <param name="sessions-per-second" value="5"/>
+ <param name="loglevel" value="debug"/>
+ <param name="crash-protection" value="false"/>
+ <param name="dump-cores" value="yes"/>
+ <param name="rtp-start-port" value="16384"/>
+ <param name="rtp-end-port" value="16484"/>
+ </settings>
+ </configuration>
+ </section>
+
+ <section name="dialplan" description="Regex/XML Dialplan">
+ <context name="default">
+ <extension name="codec_and_sip_uri">
+ <condition field="destination_number" expression="^sip:(.*):(.*)$">
+ <action application="bridge" data="{absolute_codec_string=$1}sofia/softphone/$2"/>
+ </condition>
+ </extension>
+ <extension name="sip_uri">
+ <condition field="destination_number" expression="^sip:(.*)$">
+ <action application="bridge" data="sofia/softphone/$1"/>
+ </condition>
+ </extension>
+ <extension name="codec_and_number">
+ <condition field="destination_number" expression="^(.*):(.*)@(.*)$">
+ <action application="bridge" data="{absolute_codec_string=$1}sofia/gateway/$3/$2"/>
+ </condition>
+ </extension>
+ <extension name="number">
+ <condition field="destination_number" expression="^(.*)@(.*)$">
+ <action application="bridge" data="sofia/gateway/$2/$1"/>
+ </condition>
+ </extension>
+ <extension name="number">
+ <condition field="destination_number" expression="^(.*)$">
+ <action application="bridge" data="sofia/gateway/${default_gateway}/$1"/>
+ </condition>
+ </extension>
+ </context>
+ <context name="public">
+ <extension name="public_extensions">
+ <condition field="$${auto_answer}" expression="^true$"/>
+ <condition field="destination_number" expression="^(.*)$">
+ <action application="info"/>
+ <action application="bridge" data="portaudio/auto_answer"/>
+ </condition>
+ </extension>
+ <extension name="public_extensions">
+ <condition field="${sip_to_params}" expression="intercom=true"/>
+ <condition field="${alert_info}" expression="Ring;Answer"/>
+ <condition field="destination_number" expression="^(.*)$">
+ <action application="info"/>
+ <action application="bridge" data="portaudio/auto_answer"/>
+ </condition>
+ </extension>
+ <extension name="public_extensions">
+ <condition field="destination_number" expression="^(.*)$">
+ <action application="info"/>
+ <action application="set" data="ringback=${us-ring}"/>
+ <action application="pre_answer"/>
+ <action application="bridge" data="portaudio"/>
+ </condition>
+ </extension>
+ </context>
+ </section>
+</document>
+
+
Modified: freeswitch/trunk/contrib/jmesquita/FSPhone/fshost.cpp (16121 => 16122)
--- freeswitch/trunk/contrib/jmesquita/FSPhone/fshost.cpp 2010-01-02 18:37:31 UTC (rev 16121)
+++ freeswitch/trunk/contrib/jmesquita/FSPhone/fshost.cpp 2010-01-02 22:06:42 UTC (rev 16122)
@@ -20,17 +20,28 @@
void FSHost::run(void)
{
- switch_core_flag_t flags = SCF_USE_SQL | SCF_USE_AUTO_NAT;
+ switch_core_flag_t flags = SCF_USE_SQL; //| SCF_USE_AUTO_NAT;
const char *err = NULL;
switch_bool_t console = SWITCH_FALSE;
switch_status_t destroy_status;
+ QDir conf_dir = QDir(QCoreApplication::applicationDirPath());
+ if (conf_dir.cd("conf"))
+ {
+ SWITCH_GLOBAL_dirs.conf_dir = (char *) malloc(strlen(conf_dir.absolutePath().toAscii().constData()) + 1);
+ if (!SWITCH_GLOBAL_dirs.conf_dir) {
+ fprintf(stderr, "Allocation error\n");
+ emit coreLoadingError(err);
+ }
+ strcpy(SWITCH_GLOBAL_dirs.conf_dir, conf_dir.absolutePath().toAscii().constData());
+ }
+
/* If you need to override configuration directories, you need to change them in the SWITCH_GLOBAL_dirs global structure */
printf("Initializing core...\n");
/* Initialize the core and load modules, that will startup FS completely */
if (switch_core_init_and_modload(flags, console, &err) != SWITCH_STATUS_SUCCESS) {
- fprintf(stderr, "Failed to initialize FreeSWITCH's core: %s\n", err);
- return;
+ fprintf(stderr, "Failed to initialize FreeSWITCH's core: %s\n", err);
+ emit coreLoadingError(err);
}
printf("Everything OK, Entering runtime loop.\n");
Modified: freeswitch/trunk/contrib/jmesquita/FSPhone/fshost.h (16121 => 16122)
--- freeswitch/trunk/contrib/jmesquita/FSPhone/fshost.h 2010-01-02 18:37:31 UTC (rev 16121)
+++ freeswitch/trunk/contrib/jmesquita/FSPhone/fshost.h 2010-01-02 22:06:42 UTC (rev 16122)
@@ -43,6 +43,7 @@
void run(void);
signals:
+ void coreLoadingError(QString);
void ready(void);
void ringing(QString);
void answered(QString);
Modified: freeswitch/trunk/contrib/jmesquita/FSPhone/main.cpp (16121 => 16122)
--- freeswitch/trunk/contrib/jmesquita/FSPhone/main.cpp 2010-01-02 18:37:31 UTC (rev 16121)
+++ freeswitch/trunk/contrib/jmesquita/FSPhone/main.cpp 2010-01-02 22:06:42 UTC (rev 16122)
@@ -1,10 +1,22 @@
#include <QtGui/QApplication>
+#include <QSplashScreen>
#include "mainwindow.h"
int main(int argc, char *argv[])
{
QApplication a(argc, argv);
- MainWindow w;
- w.show();
+ QCoreApplication::setOrganizationName("FreeSWITCH");
+ QCoreApplication::setOrganizationDomain("freeswitch.org");
+ QCoreApplication::setApplicationName("FSPhone");
+
+ QPixmap image(":/images/splash.png");
+ QSplashScreen *splash = new QSplashScreen(image);
+ splash->show();
+ splash->showMessage("Loading, please wait...", Qt::AlignRight|Qt::AlignBottom, Qt::blue);
+
+ QObject::connect(&g_FSHost, SIGNAL(ready()), splash, SLOT(close()));
+ MainWindow w;
+ QObject::connect(&g_FSHost, SIGNAL(ready()), &w, SLOT(show()));
+ g_FSHost.start();
return a.exec();
}
Modified: freeswitch/trunk/contrib/jmesquita/FSPhone/mainwindow.cpp (16121 => 16122)
--- freeswitch/trunk/contrib/jmesquita/FSPhone/mainwindow.cpp 2010-01-02 18:37:31 UTC (rev 16121)
+++ freeswitch/trunk/contrib/jmesquita/FSPhone/mainwindow.cpp 2010-01-02 22:06:42 UTC (rev 16122)
@@ -1,4 +1,5 @@
#include <QInputDialog>
+#include <QMessageBox>
#include "mainwindow.h"
#include "ui_mainwindow.h"
@@ -41,13 +42,12 @@
connect(&g_FSHost, SIGNAL(hungup(QString)), this, SLOT(hungup(QString)));
connect(&g_FSHost, SIGNAL(newOutgoingCall(QString)), this, SLOT(newOutgoingCall(QString)));
connect(&g_FSHost, SIGNAL(gwStateChange(QString,int)), this, SLOT(gwStateChanged(QString,int)));
+ /*connect(&g_FSHost, SIGNAL(coreLoadingError(QString)), this, SLOT(coreLoadingError(QString)));*/
connect(ui->newCallBtn, SIGNAL(clicked()), this, SLOT(makeCall()));
connect(ui->answerBtn, SIGNAL(clicked()), this, SLOT(paAnswer()));
connect(ui->hangupBtn, SIGNAL(clicked()), this, SLOT(paHangup()));
connect(ui->listCalls, SIGNAL(itemDoubleClicked(QListWidgetItem*)), this, SLOT(callListDoubleClick(QListWidgetItem*)));
-
- g_FSHost.start();
}
MainWindow::~MainWindow()
@@ -58,6 +58,12 @@
g_FSHost.wait();
}
+void MainWindow::coreLoadingError(QString err)
+{
+ QMessageBox::warning(this, "Error Loading Core...", err, QMessageBox::Ok);
+ QApplication::exit(255);
+}
+
void MainWindow::gwStateChanged(QString gw, int state)
{
ui->statusBar->showMessage(tr("Account %1 is %2").arg(gw, g_FSHost.getGwStateName(state)));
Modified: freeswitch/trunk/contrib/jmesquita/FSPhone/mainwindow.h (16121 => 16122)
--- freeswitch/trunk/contrib/jmesquita/FSPhone/mainwindow.h 2010-01-02 18:37:31 UTC (rev 16121)
+++ freeswitch/trunk/contrib/jmesquita/FSPhone/mainwindow.h 2010-01-02 22:06:42 UTC (rev 16122)
@@ -57,6 +57,7 @@
void dtmfDialed(QString);
private slots:
+ void coreLoadingError(QString);
void gwStateChanged(QString, int);
void dialDTMF(QString);
void callListDoubleClick(QListWidgetItem *);
Modified: freeswitch/trunk/contrib/jmesquita/FSPhone/mod_qsettings/mod_qsettings.cpp (16121 => 16122)
--- freeswitch/trunk/contrib/jmesquita/FSPhone/mod_qsettings/mod_qsettings.cpp 2010-01-02 18:37:31 UTC (rev 16121)
+++ freeswitch/trunk/contrib/jmesquita/FSPhone/mod_qsettings/mod_qsettings.cpp 2010-01-02 22:06:42 UTC (rev 16122)
@@ -33,11 +33,6 @@
#include "mod_qsettings/mod_qsettings.h"
-/*
-SWITCH_MODULE_DEFINITION(mod_qsettings, mod_qsettings_load, mod_qsettings_shutdown, NULL);
-*/
-
-
struct xml_binding {
char *bindings;
};
Added: freeswitch/trunk/contrib/jmesquita/FSPhone/resources/splash.png
(Binary files differ)
Added: freeswitch/trunk/contrib/jmesquita/FSPhone/resources.qrc (0 => 16122)
--- freeswitch/trunk/contrib/jmesquita/FSPhone/resources.qrc (rev 0)
+++ freeswitch/trunk/contrib/jmesquita/FSPhone/resources.qrc 2010-01-02 22:06:42 UTC (rev 16122)
@@ -0,0 +1,5 @@
+<RCC>
+ <qresource prefix="/images">
+ <file alias="splash.png">resources/splash.png</file>
+ </qresource>
+</RCC>