diff --git a/configure.ac b/configure.ac
index b3a168ba65b780caf6fd54ac4cf5a24484e34dbe..988fc35e537e21c1680ba68c5e02b282227e57d0 100644
--- a/configure.ac
+++ b/configure.ac
@@ -188,11 +188,18 @@ PKG_CHECK_MODULES(DBUSCPP, dbus-c++-1 >= ${LIBDBUSCPP_MIN_VERSION})
AC_SUBST(DBUSCPP_LIBS)
AC_SUBST(DBUSCPP_CFLAGS)
+LIBOPENSSL_MIN_VERSION=0.9.8
+PKG_CHECK_MODULES(LIBOPENSSL, libssl >= ${LIBOPENSSL_MIN_VERSION})
+AC_SUBST(LIBOPENSSL_LIBS)
+AC_SUBST(LIBOPENSSL_CFLAGS)
+
dnl Check for libcppunit-dev
CPPUNIT_MIN_VERSION=1.12
PKG_CHECK_MODULES(CPPUNIT, cppunit >= ${CPPUNIT_MIN_VERSION})
AC_SUBST(CPPUNIT_LIBS)
+LIBSSL_MIN_VERSION=0.9.8
+
# check for libgsm1 (doesn't use pkg-config)
dnl Check for libgsm
AC_ARG_WITH([gsm],
diff --git a/sflphone-gtk/src/calltree.c b/sflphone-gtk/src/calltree.c
index 591f276ca0a91642aaf6cc3b18536017da73983f..7359a34bb1e6dde63b06c4e80bf607b236a5e4aa 100644
--- a/sflphone-gtk/src/calltree.c
+++ b/sflphone-gtk/src/calltree.c
@@ -378,6 +378,7 @@ void row_activated(GtkTreeView *tree_view UNUSED,
{
case CALL_STATE_INCOMING:
dbus_accept(selectedCall);
+ stop_notification();
break;
case CALL_STATE_HOLD:
dbus_unhold(selectedCall);
diff --git a/src/audio/alsalayer.cpp b/src/audio/alsalayer.cpp
index b1b0dd69a6eef9e913d7b7d9fd7cdee912230a9d..2d2811b363722e41d4a2d774182eec2ef21681e4 100644
--- a/src/audio/alsalayer.cpp
+++ b/src/audio/alsalayer.cpp
@@ -194,9 +194,9 @@ void AlsaLayer::stopCaptureStream (void)
if(_CaptureHandle){
err = snd_pcm_drop (_CaptureHandle);
- _debug("AlsaLayer::stopCaptureStream snd_pcm_drop returned vaue : %i\n",err);
+
stop_capture ();
- _debug("Wroking here !!!!!!!!!!!!!!!\n");
+
}
}
diff --git a/src/audio/audiostream.cpp b/src/audio/audiostream.cpp
index 778192643d5ee2ad3b2e2ba8540e2266a8cb370f..c031b4ac0d4a5119a7444736eb26a417d21d869f 100644
--- a/src/audio/audiostream.cpp
+++ b/src/audio/audiostream.cpp
@@ -66,6 +66,7 @@ AudioStream::stream_state_callback( pa_stream* s, void* user_data UNUSED )
case PA_STREAM_READY:
_debug("Stream successfully created, connected to %s\n", pa_stream_get_device_name( s ));
+ pa_stream_cork( s, 0, NULL, NULL);
break;
case PA_STREAM_UNCONNECTED:
_debug("Stream unconnected\n");
diff --git a/src/audio/pulselayer.cpp b/src/audio/pulselayer.cpp
index 40de7e886fa86a6282b40e5870c402d3af0b4a18..8101e5e23839497707a618fe7f7afd1c08c533a2 100644
--- a/src/audio/pulselayer.cpp
+++ b/src/audio/pulselayer.cpp
@@ -140,12 +140,12 @@ void PulseLayer::createStreams( pa_context* c )
playback = new AudioStream(c, PLAYBACK_STREAM, PLAYBACK_STREAM_NAME, _manager->getSpkrVolume());
pa_stream_set_write_callback( playback->pulseStream(), audioCallback, this);
- //pa_stream_set_overflow_callback( playback->pulseStream() , overflow , this);
+ // pa_stream_set_overflow_callback( playback->pulseStream() , overflow , this);
// pa_stream_set_suspended_callback( playback->pulseStream(), stream_suspended_callback, this);
record = new AudioStream(c, CAPTURE_STREAM, CAPTURE_STREAM_NAME , _manager->getMicVolume());
pa_stream_set_read_callback( record->pulseStream() , audioCallback, this);
- //pa_stream_set_underflow_callback( record->pulseStream() , underflow , this);
+ // pa_stream_set_underflow_callback( record->pulseStream() , underflow , this);
// pa_stream_set_suspended_callback(record->pulseStream(), stream_suspended_callback, this);
@@ -173,6 +173,8 @@ bool PulseLayer::openDevice(int indexIn UNUSED, int indexOut UNUSED, int sampleR
assert(context);
connectPulseAudioServer();
+
+ startStream();
_debug("Connection Done!! \n");
return true;
@@ -254,8 +256,6 @@ void PulseLayer::stream_suspended_callback(pa_stream *s, void *userdata UNUSED )
void PulseLayer::processData( void )
{
- _debug("PulseLayer::processData() \n");
-
// Handle the mic
// We check if the stream is ready
if( (record->pulseStream()) && (pa_stream_get_state( record->pulseStream()) == PA_STREAM_READY))
diff --git a/src/iaxvoiplink.cpp b/src/iaxvoiplink.cpp
index 231e024d4619433cbc8ad291411a1155a1183150..79466dc5028ab6093d3c3565d0d2ed8818162977 100644
--- a/src/iaxvoiplink.cpp
+++ b/src/iaxvoiplink.cpp
@@ -74,6 +74,8 @@ IAXVoIPLink::~IAXVoIPLink()
audiolayer = NULL;
+ delete converter;
+
delete [] micData; micData = NULL;
delete [] micDataConverted; micDataConverted = NULL;
delete [] micDataEncoded; micDataEncoded = NULL;
diff --git a/src/managerimpl.cpp b/src/managerimpl.cpp
index c24affc647efee0a88119fc1d99cf61bd7dd42db..4ad23121b347059f92cc050def291349b38f7c52 100644
--- a/src/managerimpl.cpp
+++ b/src/managerimpl.cpp
@@ -124,7 +124,7 @@ ManagerImpl::init()
AudioLayer *audiolayer = getAudioDriver();
- if (audiolayer!=0) {
+ if (audiolayer != 0) {
unsigned int sampleRate = audiolayer->getSampleRate();
_debugInit("Load Telephone Tone");
@@ -134,6 +134,9 @@ ManagerImpl::init()
_debugInit("Loading DTMF key");
_dtmfKey = new DTMF(sampleRate);
}
+
+ if (audiolayer == 0)
+ audiolayer->stopStream();
}
void ManagerImpl::terminate()
@@ -190,6 +193,9 @@ ManagerImpl::outgoingCall(const std::string& accountid, const CallID& id, const
{
_debug("ManagerImpl::outgoingCall() method \n");
+ stopTone(false);
+ playTone();
+
if (!accountExists(accountid)) {
_debug("! Manager Error: Outgoing Call: account doesn't exist\n");
return false;
@@ -221,6 +227,14 @@ ManagerImpl::outgoingCall(const std::string& accountid, const CallID& id, const
bool
ManagerImpl::answerCall(const CallID& id)
{
+ bool isActive = false;
+
+ AccountID currentaccountid = getAccountFromCall( id );
+ Call* currentcall = getAccountLink(currentaccountid)->getCall(getCurrentCallId());
+ _debug("ManagerImpl::answerCall :: current call->getState %i \n",currentcall->getState());
+
+ if (currentcall->getState() == 1)
+ isActive = true;
stopTone(false);
_debug("Try to answer call: %s\n", id.data());
@@ -230,10 +244,27 @@ ManagerImpl::answerCall(const CallID& id)
return false;
}
- if (id != getCurrentCallId()) {
+ /*
+ _debug("_nbIncomingWaitingCall =======>>>>>>>> %i \n",_nbIncomingWaitingCall);
+
+ CallIDSet::iterator iter = _waitingCall.begin();
+ while (iter != _waitingCall.end()) {
+ CallID ident = *iter;
+ AccountID acc = getAccountFromCall( ident );
+ Call* call = getAccountLink(acc)->getCall(ident);
+ _debug("ManagerImpl::answerCall :: incoming call ident: %s \n",ident.c_str());
+ _debug("ManagerImpl::answerCall :: incoming call state: %i \n",call->getState());
+ ++iter;
+ }
+ */
+
+ // if (id != getCurrentCallId()) {
+ if (isActive) {
_debug("* Manager Info: there is currently a call, try to hold it\n");
+
onHoldCall(getCurrentCallId());
}
+
if (!getAccountLink(accountid)->answer(id)) {
// error when receiving...
@@ -313,11 +344,11 @@ ManagerImpl::cancelCall (const CallID& id)
bool
ManagerImpl::onHoldCall(const CallID& id)
{
- _debug("*************** ON HOLD ***********************************\n");
+
stopTone(true);
AccountID accountid = getAccountFromCall( id );
if (accountid == AccountNULL) {
- _debug("5 Manager On Hold Call: Account ID %s or callid %s desn't exists\n", accountid.c_str(), id.c_str());
+ _debug("5 Manager On Hold Call: Account ID %s or callid %s doesn't exists\n", accountid.c_str(), id.c_str());
return false;
}
@@ -336,7 +367,7 @@ ManagerImpl::onHoldCall(const CallID& id)
bool
ManagerImpl::offHoldCall(const CallID& id)
{
- _debug("*************** OFF HOLD ***********************************\n");
+
stopTone(false);
AccountID accountid = getAccountFromCall( id );
if (accountid == AccountNULL) {
@@ -598,13 +629,21 @@ ManagerImpl::incomingCall(Call* call, const AccountID& accountId)
associateCallToAccount(call->getCallId(), accountId);
+ _debug("ManagerImpl::incomingCall :: hasCurrentCall() %i \n",hasCurrentCall());
+
if ( !hasCurrentCall() ) {
call->setConnectionState(Call::Ringing);
ringtone();
switchCall(call->getCallId());
- } else {
+
+ }
+ /*
+ else {
addWaitingCall(call->getCallId());
}
+ */
+
+ addWaitingCall(call->getCallId());
from = call->getPeerName();
number = call->getPeerNumber();
@@ -618,6 +657,15 @@ ManagerImpl::incomingCall(Call* call, const AccountID& accountId)
from.append(number);
from.append(">");
}
+
+ /*
+ CallIDSet::iterator iter = _waitingCall.begin();
+ while (iter != _waitingCall.end()) {
+ CallID ident = *iter;
+ _debug("ManagerImpl::incomingCall :: CALL iteration: %s \n",ident.c_str());
+ ++iter;
+ }
+ */
/* Broadcast a signal over DBus */
_dbus->getCallManager()->incomingCall(accountId, call->getCallId(), from);
@@ -1592,6 +1640,8 @@ void ManagerImpl::setAudioManager( const int32_t& api )
{
int manager;
+ _debug(" ManagerImpl::setAudioManager :: %i \n",api);
+
manager = api;
if( manager == PULSEAUDIO )
{
@@ -1705,8 +1755,8 @@ ManagerImpl::initAudioDriver(void)
if (error == -1) {
_debug("Init audio driver: %i\n", error);
}
- }
-
+ }
+
}
/**
@@ -1756,6 +1806,7 @@ ManagerImpl::selectAudioDriver (void)
/* Notify the error if there is one */
if( _audiodriver -> getErrorMessage() != -1 )
notifyErrClient( _audiodriver -> getErrorMessage());
+
}
void ManagerImpl::switchAudioManager (void)
@@ -1796,6 +1847,16 @@ void ManagerImpl::switchAudioManager (void)
_audiodriver->openDevice( numCardIn , numCardOut, samplerate, framesize, SFL_PCM_BOTH, alsaPlugin );
if( _audiodriver -> getErrorMessage() != -1 )
notifyErrClient( _audiodriver -> getErrorMessage());
+
+ _debug("Current device: %i \n", type);
+ _debug("has current call: %i \n", hasCurrentCall());
+
+ // need to stop audio streams if there is currently no call
+ if( (type != PULSEAUDIO) && (!hasCurrentCall())) {
+ _debug("There is currently a call!!\n");
+ _audiodriver->stopStream();
+
+ }
}
/**
diff --git a/src/sipvoiplink.cpp b/src/sipvoiplink.cpp
index 33e738f345a47cb7145525ed6a6a0fba92ea60ab..a1fec949673df9f2ae270c6031a46a987574ee7f 100644
--- a/src/sipvoiplink.cpp
+++ b/src/sipvoiplink.cpp
@@ -267,6 +267,7 @@ SIPVoIPLink::getEvent()
int SIPVoIPLink::sendRegister( AccountID id )
{
+ _debug("sendRegister called!!!!!!!!!!!!!!!!!!!!!!! \n");
pj_status_t status;
int expire_value;
char contactTmp[256];
@@ -1348,36 +1349,34 @@ void SIPVoIPLink::setStunServer( const std::string &server )
pj_sockaddr_in bound_addr;
pjsip_host_port a_name;
char tmpIP[32];
- pj_sock_t sock;
+ pj_sock_t sock;
// Init bound address to ANY
pj_memset(&bound_addr, 0, sizeof (bound_addr));
- bound_addr.sin_addr.s_addr = PJ_INADDR_ANY;
+
+
+ bound_addr.sin_addr.s_addr = pj_htonl(PJ_INADDR_ANY);
+ bound_addr.sin_port = pj_htons((pj_uint16_t) _localPort);
+ bound_addr.sin_family = PJ_AF_INET;
+ pj_bzero(bound_addr.sin_zero, sizeof(bound_addr.sin_zero));
- // Create UDP server socket
- status = pj_sock_socket(PJ_AF_INET, PJ_SOCK_DGRAM, 0, &sock);
- if (status != PJ_SUCCESS) {
- _debug("UserAgent: (%d) UDP socket() error\n", status);
- return status;
- }
- status = pj_sock_bind_in(sock, pj_ntohl(bound_addr.sin_addr.s_addr), (pj_uint16_t) _localPort);
- if (status != PJ_SUCCESS) {
- _debug("UserAgent: (%d) UDP bind() error\n", status);
- pj_sock_close(sock);
- return status;
- }
- _debug("UserAgent: Use IP: %s\n", _localExternAddress.data());
+ _debug("bound_addr.sin_port %i \n", bound_addr.sin_port);
+
+ _debug("UserAgent: Use IP: %s\n", _localExternAddress.data());
+
+
// Create UDP-Server (default port: 5060)
strcpy(tmpIP, _localExternAddress.data());
pj_strdup2(_pool, &a_name.host, tmpIP);
a_name.port = (pj_uint16_t) _localExternPort;
- _debug("a_name: host: %s - port : %i\n", a_name.host.ptr, a_name.port);
- status = pjsip_udp_transport_attach(_endpt, sock, &a_name, 1, NULL);
+ // status = pjsip_tls_transport_start(_endpt, NULL, &bound_addr, &a_name, 1, NULL);
+ status = pjsip_udp_transport_start(_endpt, &bound_addr, &a_name, 1, NULL);
+
if (status != PJ_SUCCESS) {
_debug("UserAgent: (%d) Unable to start UDP transport!\n", status);
return -1;