diff --git a/.gitignore b/.gitignore
index 6668eaa3a512bd33f0fe35cd4b2b1ae967acabd8..38beb5d7dae69d41a269a0fed6c93a4e2bab2ff9 100644
--- a/.gitignore
+++ b/.gitignore
@@ -42,7 +42,8 @@ doc/doxygen/gtk-gui-doc
doc/*.html
# Ignore built stuff
-/src/.libs
+.libs
+.deps
# PJSIP files
*.depend
diff --git a/daemon/src/audio/alsa/alsalayer.cpp b/daemon/src/audio/alsa/alsalayer.cpp
index c3416825413e4947bda9b67b60c4997e85463508..5363ba338f53e30c6933e13dd90068aa1dedb8c9 100644
--- a/daemon/src/audio/alsa/alsalayer.cpp
+++ b/daemon/src/audio/alsa/alsalayer.cpp
@@ -523,7 +523,7 @@ namespace
void AlsaLayer::capture(void)
{
- unsigned int mainBufferSampleRate = getMainBuffer()->getInternalSamplingRate();
+ unsigned int mainBufferSampleRate = Manager::instance().getMainBuffer()->getInternalSamplingRate();
bool resample = audioSampleRate_ != mainBufferSampleRate;
int toGetSamples = snd_pcm_avail_update (captureHandle_);
@@ -550,11 +550,11 @@ void AlsaLayer::capture(void)
SFLDataFormat* rsmpl_out = (SFLDataFormat*) malloc (outBytes);
converter_->resample ( (SFLDataFormat*) in, rsmpl_out, mainBufferSampleRate, audioSampleRate_, toGetSamples);
dcblocker_.process(rsmpl_out, rsmpl_out, outSamples);
- getMainBuffer()->putData (rsmpl_out, outBytes);
+ Manager::instance().getMainBuffer()->putData (rsmpl_out, outBytes);
free (rsmpl_out);
} else {
dcblocker_.process(in, in, toGetSamples);
- getMainBuffer()->putData (in, toGetBytes);
+ Manager::instance().getMainBuffer()->putData (in, toGetBytes);
}
end:
@@ -565,10 +565,10 @@ void AlsaLayer::playback(int maxSamples)
{
unsigned short spkrVolume = Manager::instance().getSpkrVolume();
- unsigned int mainBufferSampleRate = getMainBuffer()->getInternalSamplingRate();
+ unsigned int mainBufferSampleRate = Manager::instance().getMainBuffer()->getInternalSamplingRate();
bool resample = audioSampleRate_ != mainBufferSampleRate;
- int toGet = getMainBuffer()->availForGet();
+ int toGet = Manager::instance().getMainBuffer()->availForGet();
int toPut = maxSamples * sizeof(SFLDataFormat);
if (toGet <= 0) { // no audio available, play tone or silence
@@ -602,7 +602,7 @@ void AlsaLayer::playback(int maxSamples)
toGet = maxNbBytesToGet;
SFLDataFormat *out = (SFLDataFormat*) malloc (toGet);
- getMainBuffer()->getData (out, toGet);
+ Manager::instance().getMainBuffer()->getData (out, toGet);
adjustVolume(out, toGet / sizeof(SFLDataFormat), spkrVolume);
if (resample) {
@@ -644,7 +644,7 @@ void AlsaLayer::audioCallback (void)
write (out, toGet, playbackHandle_);
free (out);
// Consume the regular one as well (same amount of bytes)
- getMainBuffer()->discard (toGet);
+ Manager::instance().getMainBuffer()->discard (toGet);
} else {
// regular audio data
playback(playbackAvailSmpl);
diff --git a/daemon/src/audio/audiolayer.cpp b/daemon/src/audio/audiolayer.cpp
index 2e440b3c8503c8577600fc178ea038838bbc5310..f568abdd749a68f128c927e6d6b5bf30cf5d5a9b 100644
--- a/daemon/src/audio/audiolayer.cpp
+++ b/daemon/src/audio/audiolayer.cpp
@@ -56,7 +56,7 @@ void AudioLayer::flushMain (void)
{
ost::MutexLock guard (mutex_);
// should pass call id
- getMainBuffer()->flushAllBuffers();
+ Manager::instance().getMainBuffer()->flushAllBuffers();
}
void AudioLayer::flushUrgent (void)
diff --git a/daemon/src/audio/audiolayer.h b/daemon/src/audio/audiolayer.h
index 3fb9b1d5b67132bf5bfc19ff5fb1078934a099c3..a3243ee9b0c79765236b36651502d474bb145f45 100644
--- a/daemon/src/audio/audiolayer.h
+++ b/daemon/src/audio/audiolayer.h
@@ -111,18 +111,6 @@ class AudioLayer
return audioSampleRate_;
}
- /**
- * Get a pointer to the application MainBuffer class.
- *
- * In order to send signal to other parts of the application, one must pass through the mainbuffer.
- * Audio instances must be registered into the MainBuffer and bound together via the ManagerImpl.
- *
- * @return MainBuffer* a pointer to the MainBuffer instance
- */
- MainBuffer* getMainBuffer (void) const {
- return Manager::instance().getMainBuffer();
- }
-
/**
* Get the mutex lock for the entire audio layer
*/
diff --git a/daemon/src/audio/pulseaudio/pulselayer.cpp b/daemon/src/audio/pulseaudio/pulselayer.cpp
index 60b79e2a68ffbaa879ccf0e8cf17bf5949f3f044..c6b1148dc2be6cb2053e077d1b3435a4b1d38470 100644
--- a/daemon/src/audio/pulseaudio/pulselayer.cpp
+++ b/daemon/src/audio/pulseaudio/pulselayer.cpp
@@ -434,7 +434,7 @@ void PulseLayer::writeToSpeaker (void)
urgentRingBuffer_.Get (data, urgentBytes);
pa_stream_write (s, data, urgentBytes, NULL, 0, PA_SEEK_RELATIVE);
// Consume the regular one as well (same amount of bytes)
- getMainBuffer()->discard (urgentBytes);
+ Manager::instance().getMainBuffer()->discard (urgentBytes);
return;
}
@@ -450,7 +450,7 @@ void PulseLayer::writeToSpeaker (void)
flushUrgent(); // flush remaining samples in _urgentRingBuffer
- size_t availSamples = getMainBuffer()->availForGet() / sizeof(SFLDataFormat);
+ size_t availSamples = Manager::instance().getMainBuffer()->availForGet() / sizeof(SFLDataFormat);
if (availSamples == 0) {
pa_stream_begin_write(s, &data, &bytes);
memset(data, 0, bytes);
@@ -458,7 +458,7 @@ void PulseLayer::writeToSpeaker (void)
return;
}
- unsigned int mainBufferSampleRate = getMainBuffer()->getInternalSamplingRate();
+ unsigned int mainBufferSampleRate = Manager::instance().getMainBuffer()->getInternalSamplingRate();
bool resample = audioSampleRate_ != mainBufferSampleRate;
// how much samples we can write in the output
@@ -479,7 +479,7 @@ void PulseLayer::writeToSpeaker (void)
size_t inBytes = inSamples * sizeof (SFLDataFormat);
pa_stream_begin_write(s, &data, &inBytes);
- getMainBuffer()->getData (data, inBytes);
+ Manager::instance().getMainBuffer()->getData (data, inBytes);
if (resample) {
SFLDataFormat* rsmpl_out = (SFLDataFormat*) pa_xmalloc (outBytes);
@@ -502,7 +502,7 @@ void PulseLayer::readFromMic (void)
return;
}
- unsigned int mainBufferSampleRate = getMainBuffer()->getInternalSamplingRate();
+ unsigned int mainBufferSampleRate = Manager::instance().getMainBuffer()->getInternalSamplingRate();
bool resample = audioSampleRate_ != mainBufferSampleRate;
if (resample) {
double resampleFactor = (double) audioSampleRate_ / mainBufferSampleRate;
@@ -520,7 +520,7 @@ void PulseLayer::readFromMic (void)
converter_->resample((SFLDataFormat*)data, mic_buffer_, mainBufferSampleRate, audioSampleRate_, samples);
dcblocker_.process(mic_buffer_, resample ? mic_buffer_ : (SFLDataFormat*)data, samples);
- getMainBuffer()->putData(mic_buffer_, bytes);
+ Manager::instance().getMainBuffer()->putData(mic_buffer_, bytes);
if (pa_stream_drop (record_->pulseStream()) < 0)
_error ("Audio: Error: capture stream drop failed: %s" , pa_strerror (pa_context_errno (context_)));
diff --git a/daemon/src/iax/iaxcall.cpp b/daemon/src/iax/iaxcall.cpp
index fa09091f32ce79e6d9dfab091f09e3e983a813dd..e2a8436d724ee68286bb2cb80362d1d7c52563c8 100644
--- a/daemon/src/iax/iaxcall.cpp
+++ b/daemon/src/iax/iaxcall.cpp
@@ -72,17 +72,10 @@ namespace {
}
}
-IAXCall::IAXCall (const std::string& id, Call::CallType type) : Call (id, type), _session (NULL)
+IAXCall::IAXCall (const std::string& id, Call::CallType type) : Call (id, type), session (NULL)
{
}
-void
-IAXCall::setFormat (int format)
-{
- _format = format;
- setAudioCodec(ASTFormatToCodec(format));
-}
-
int
IAXCall::getSupportedFormat (const std::string &accountID) const
{
@@ -118,7 +111,7 @@ int IAXCall::getFirstMatchingFormat (int needles, const std::string &accountID)
return 0;
}
-int IAXCall::getAudioCodec()
+int IAXCall::getAudioCodec(void)
{
- return _audioCodec;
+ return ASTFormatToCodec(format);
}
diff --git a/daemon/src/iax/iaxcall.h b/daemon/src/iax/iaxcall.h
index a4579e4d1bc9a752f6735156251908ce976d71ee..64765f97716dc1c95f8a31f527b4dbced43af268 100644
--- a/daemon/src/iax/iaxcall.h
+++ b/daemon/src/iax/iaxcall.h
@@ -50,38 +50,6 @@ class IAXCall : public Call
*/
IAXCall (const std::string& id, Call::CallType type);
- /**
- * @return iax_session* The session pointer or NULL
- */
- iax_session* getSession() const {
- return _session;
- }
-
- /**
- * Set the session pointer
- * @param session the session pointer to assign
- */
- void setSession (iax_session* session) {
- _session = session;
- }
-
- /**
- * Set format (one single bit)
- * This function sets the _audioCodec variable with the correct
- * codec.
- * @param format The format representing the codec
- */
- void setFormat (int format);
-
- /**
- * Get format for the voice codec used
- * @return int Bitmask for codecs defined in iax/frame.h
- */
- int getFormat() const {
- return _format;
- }
-
-
/**
* @return int The bitwise list of supported formats
*/
@@ -102,32 +70,10 @@ class IAXCall : public Call
*/
int getFirstMatchingFormat (int needles, const std::string &accountID) const;
- /**
- * Return audio codec [mutex protected]
- * @return int The payload of the codec
- */
- int getAudioCodec();
-
- private:
- /** Each call is associated with an iax_session */
- iax_session* _session;
+ int getAudioCodec(void);
- /**
- * Set the audio codec used. [not protected]
- * @param audioCodec The payload of the codec
- */
- void setAudioCodec (int audioCodec) {
- _audioCodec = audioCodec;
- }
-
- /** Codec pointer */
- int _audioCodec;
-
- /**
- * Format currently in use in the conversation,
- * sent in each outgoing voice packet.
- */
- int _format;
+ int format;
+ iax_session* session;
};
#endif
diff --git a/daemon/src/iax/iaxvoiplink.cpp b/daemon/src/iax/iaxvoiplink.cpp
index 742ea4b254aae73cbf04450d3df2a17859908709..46827988746b0ce0c360ddc8123fb6d503cb535e 100644
--- a/daemon/src/iax/iaxvoiplink.cpp
+++ b/daemon/src/iax/iaxvoiplink.cpp
@@ -41,151 +41,89 @@
#include <cmath>
#include <dlfcn.h>
-#define IAX_BLOCKING 1
-#define IAX_NONBLOCKING 0
-
-#define IAX_SUCCESS 0
-#define IAX_FAILURE -1
-
-#define RANDOM_IAX_PORT rand() % 64000 + 1024
-
-#define MUSIC_ONHOLD true
-
-#define CHK_VALID_CALL if (call == NULL) { _debug("IAX: Call doesn't exists"); \
- return false; }
-
-namespace {
- const char * const HOOKS = "Hooks"; /** Hooks section */
- const char * const URLHOOK_IAX2_ENABLED = "Hooks.iax2_enabled";
- const char * const URLHOOK_COMMAND = "Hooks.url_command";
-} // end anonymous namespace
-
IAXVoIPLink::IAXVoIPLink (const std::string& accountID) :
evThread_(new EventThread(this))
, regSession_(NULL)
, nextRefreshStamp_(0)
- , audiolayer_(NULL)
- , converterSamplingRate_(0)
- , converter_(new SamplerateConverter (44100))
+ , converter_(44100)
, initDone_(false)
, accountID_(accountID)
{
- // to get random number for RANDOM_PORT
- srand(time(NULL));
+ srand(time(NULL)); // to get random number for RANDOM_PORT
}
IAXVoIPLink::~IAXVoIPLink()
{
- delete converter_;
delete evThread_;
- regSession_ = NULL; // shall not delete it
+ regSession_ = NULL; // shall not delete it // XXX: but why?
terminate();
}
void
IAXVoIPLink::init()
{
- // If it was done, don't do it again, until we call terminate()
if (initDone_)
return;
- int port = IAX_DEFAULT_PORTNO;
-
- int last_port = 0;
-
- int nbTry = 3;
-
- while (port != IAX_FAILURE && nbTry) {
- last_port = port;
- port = iax_init (port);
-
- if (port < 0) {
- _debug ("IAX Warning: already initialize on port %d", last_port);
- port = RANDOM_IAX_PORT;
- } else if (port == IAX_FAILURE) {
- _debug ("IAX Fail to start on port %d", last_port);
- port = RANDOM_IAX_PORT;
- } else {
- _debug ("IAX Info: listening on port %d", last_port);
+ for (int port = IAX_DEFAULT_PORTNO, nbTry = 0; nbTry < 3 ; port = rand() % 64000 + 1024, nbTry++) {
+ if (iax_init(port) >= 0) {
evThread_->start();
-
- audiolayer_ = Manager::instance().getAudioDriver();
-
- // may be different than one already setted
- converterSamplingRate_ = Manager::instance().getMainBuffer()->getInternalSamplingRate();
-
- break;
+ initDone_ = true;
+ break;
}
-
- nbTry--;
-
- initDone_ = true;
- }
-
- if (port == IAX_FAILURE or nbTry == 0) {
- _debug ("Fail to initialize iax");
- initDone_ = false;
}
}
void
IAXVoIPLink::terminate()
{
- // If it was done, don't do it again, until we call init()
if (!initDone_)
return;
- // Hangup all calls
- terminateIAXCall();
-
- initDone_ = false;
-}
-
-void
-IAXVoIPLink::terminateIAXCall()
-{
ost::MutexLock m(_callMapMutex);
for (CallMap::iterator iter = _callMap.begin(); iter != _callMap.end(); ++iter) {
IAXCall *call = dynamic_cast<IAXCall*> (iter->second);
if (call) {
- {
- ost::MutexLock m(mutexIAX_);
- iax_hangup (call->getSession(), (char*) "Dumped Call");
- }
- call->setSession (NULL);
+ ost::MutexLock m(mutexIAX_);
+ iax_hangup (call->session, (char*) "Dumped Call");
delete call;
}
}
_callMap.clear();
+
+ initDone_ = false;
}
void
IAXVoIPLink::getEvent()
{
- {
- ost::MutexLock m(mutexIAX_);
- iax_event *event;
- while ((event = iax_get_event (IAX_NONBLOCKING)) != NULL) {
- // If we received an 'ACK', libiax2 tells apps to ignore them.
- if (event->etype == IAX_EVENT_NULL)
- continue;
-
- IAXCall *call = iaxFindCallBySession (event->session);
-
- if (call)
- iaxHandleCallEvent (event, call);
- else if (event->session && event->session == regSession_) // This is a registration session, deal with it
- iaxHandleRegReply (event);
- else // We've got an event before it's associated with any call
- iaxHandlePrecallEvent (event);
-
- iax_event_free (event);
- }
- free(event);
- }
+ mutexIAX_.enter();
+ iax_event *event;
+ while ((event = iax_get_event(0)) != NULL) {
+ // If we received an 'ACK', libiax2 tells apps to ignore them.
+ if (event->etype == IAX_EVENT_NULL) {
+ iax_event_free (event);
+ continue;
+ }
+
+ IAXCall *call = iaxFindCallBySession (event->session);
+
+ if (call)
+ iaxHandleCallEvent (event, call);
+ else if (event->session && event->session == regSession_)
+ iaxHandleRegReply (event); // This is a registration session, deal with it
+ else // We've got an event before it's associated with any call
+ iaxHandlePrecallEvent (event);
+
+ iax_event_free (event);
+ }
+ mutexIAX_.leave();
+
+ if (nextRefreshStamp_ && nextRefreshStamp_ < time (NULL))
+ sendRegister(dynamic_cast<IAXAccount *> (Manager::instance().getAccount(accountID_)));
sendAudioFromMic();
@@ -196,10 +134,6 @@ IAXVoIPLink::getEvent()
void
IAXVoIPLink::sendAudioFromMic (void)
{
- // We have to update the audio layer type in case we switched
- // TODO Find out a better way to do it
- updateAudiolayer();
-
for (CallMap::const_iterator iter = _callMap.begin(); iter != _callMap.end() ; ++iter) {
IAXCall *currentCall = dynamic_cast<IAXCall*>(iter->second);
if (!currentCall or currentCall->getState() != Call::Active)
@@ -208,21 +142,21 @@ IAXVoIPLink::sendAudioFromMic (void)
int codecType = currentCall->getAudioCodec();
sfl::AudioCodec *audioCodec = static_cast<sfl::AudioCodec *>(Manager::instance().audioCodecFactory.getCodec(codecType));
- if (!audioCodec or !audiolayer_)
+ if (!audioCodec)
continue;
Manager::instance().getMainBuffer()->setInternalSamplingRate (audioCodec->getClockRate());
- unsigned int mainBufferSampleRate = audiolayer_->getMainBuffer()->getInternalSamplingRate();
+ unsigned int mainBufferSampleRate = Manager::instance().getMainBuffer()->getInternalSamplingRate();
// we have to get 20ms of data from the mic *20/1000 = /50
// rate/50 shall be lower than IAX__20S_48KHZ_MAX
int bytesNeeded = mainBufferSampleRate * 20 / 1000 * sizeof (SFLDataFormat);
- if (audiolayer_->getMainBuffer()->availForGet (currentCall->getCallId()) < bytesNeeded)
+ if (Manager::instance().getMainBuffer()->availForGet (currentCall->getCallId()) < bytesNeeded)
continue;
// Get bytes from micRingBuffer to data_from_mic
- int bytes = audiolayer_->getMainBuffer()->getData (decData, bytesNeeded, currentCall->getCallId());
+ int bytes = Manager::instance().getMainBuffer()->getData (decData, bytesNeeded, currentCall->getCallId());
int samples = bytes / sizeof(SFLDataFormat);
int compSize;
@@ -230,7 +164,7 @@ IAXVoIPLink::sendAudioFromMic (void)
int outSamples;
SFLDataFormat *in;
if (audioRate != mainBufferSampleRate) {
- converter_->resample (decData, resampledData, audioRate, mainBufferSampleRate, samples);
+ converter_.resample (decData, resampledData, audioRate, mainBufferSampleRate, samples);
in = resampledData;
outSamples = 0;
} else {
@@ -239,16 +173,11 @@ IAXVoIPLink::sendAudioFromMic (void)
}
compSize = audioCodec->encode (encodedData, in, DEC_BUFFER_SIZE);
- // Send it out!
- {
- ost::MutexLock m(mutexIAX_);
-
- // Make sure the session and the call still exists.
- if (currentCall->getSession() and (bytes > 0)) {
- if (iax_send_voice (currentCall->getSession(), currentCall->getFormat(), encodedData, compSize, outSamples) == -1)
- _error ("IAX: Error sending voice data.");
- }
- }
+ if (currentCall->session and bytes > 0) {
+ ost::MutexLock m(mutexIAX_);
+ if (iax_send_voice (currentCall->session, currentCall->format, encodedData, compSize, outSamples) == -1)
+ _error ("IAX: Error sending voice data.");
+ }
}
}
@@ -287,16 +216,15 @@ IAXVoIPLink::sendRegister (Account *a)
void
IAXVoIPLink::sendUnregister (Account *a)
{
- ost::MutexLock m(mutexIAX_);
if (regSession_) {
+ ost::MutexLock m(mutexIAX_);
iax_destroy (regSession_);
regSession_ = NULL;
}
nextRefreshStamp_ = 0;
- IAXAccount *account = dynamic_cast<IAXAccount*>(a);
- account->setRegistrationState (Unregistered);
+ dynamic_cast<IAXAccount*>(a)->setRegistrationState(Unregistered);
}
Call*
@@ -307,14 +235,10 @@ IAXVoIPLink::newOutgoingCall (const std::string& id, const std::string& toUrl)
call->setPeerNumber (toUrl);
call->initRecFileName (toUrl);
- if (iaxOutgoingInvite (call)) {
- call->setConnectionState (Call::Progressing);
- call->setState (Call::Active);
- addCall (call);
- } else {
- delete call;
- call = NULL;
- }
+ iaxOutgoingInvite (call);
+ call->setConnectionState (Call::Progressing);
+ call->setState (Call::Active);
+ addCall (call);
return call;
}
@@ -327,16 +251,14 @@ IAXVoIPLink::answer (Call *c)
Manager::instance().addStream (call->getCallId());
- {
- ost::MutexLock m(mutexIAX_);
- iax_answer(call->getSession());
- }
+ mutexIAX_.enter();
+ iax_answer(call->session);
+ mutexIAX_.leave();
call->setState(Call::Active);
call->setConnectionState(Call::Connected);
- // Flush main buffer
- audiolayer_->flushMain();
+ Manager::instance().getMainBuffer()->flushAllBuffers();
}
void
@@ -350,11 +272,11 @@ IAXVoIPLink::hangup (const std::string& id)
Manager::instance().getMainBuffer()->unBindAll(call->getCallId());
- mutexIAX_.enterMutex();
- iax_hangup (call->getSession(), (char*) "Dumped Call");
- mutexIAX_.leaveMutex();
+ mutexIAX_.enter();
+ iax_hangup (call->session, (char*) "Dumped Call");
+ mutexIAX_.leave();
- call->setSession(NULL);
+ call->session = NULL;
removeCall(id);
}
@@ -371,7 +293,7 @@ IAXVoIPLink::peerHungup (const std::string& id)
Manager::instance().getMainBuffer()->unBindAll (call->getCallId());
- call->setSession (NULL);
+ call->session = NULL;
removeCall (id);
}
@@ -387,9 +309,9 @@ IAXVoIPLink::onhold (const std::string& id)
Manager::instance().getMainBuffer()->unBindAll (call->getCallId());
- mutexIAX_.enterMutex();
- iax_quelch_moh (call->getSession(), MUSIC_ONHOLD);
- mutexIAX_.leaveMutex();
+ mutexIAX_.enter();
+ iax_quelch_moh (call->session, true);
+ mutexIAX_.leave();
call->setState (Call::Hold);
}
@@ -403,10 +325,10 @@ IAXVoIPLink::offhold (const std::string& id)
Manager::instance().addStream (call->getCallId());
- mutexIAX_.enterMutex();
- iax_unquelch (call->getSession());
- mutexIAX_.leaveMutex();
- audiolayer_->startStream();
+ mutexIAX_.enter();
+ iax_unquelch (call->session);
+ mutexIAX_.leave();
+ Manager::instance().getAudioDriver()->startStream();
call->setState (Call::Active);
}
@@ -420,31 +342,28 @@ IAXVoIPLink::transfer (const std::string& id, const std::string& to)
char callto[to.length() +1];
strcpy (callto, to.c_str());
- mutexIAX_.enterMutex();
- iax_transfer (call->getSession(), callto);
- mutexIAX_.leaveMutex();
+ mutexIAX_.enter();
+ iax_transfer (call->session, callto);
+ mutexIAX_.leave();
}
bool
IAXVoIPLink::attendedTransfer(const std::string& /*transferID*/, const std::string& /*targetID*/)
{
- // TODO implement attended transfer for IAX
- return false;
+ return false; // TODO
}
-bool
+void
IAXVoIPLink::refuse (const std::string& id)
{
IAXCall* call = getIAXCall (id);
- CHK_VALID_CALL;
-
- mutexIAX_.enterMutex();
- iax_reject (call->getSession(), (char*) "Call rejected manually.");
- mutexIAX_.leaveMutex();
-
- removeCall (id);
+ if (call) {
+ mutexIAX_.enter();
+ iax_reject (call->session, (char*) "Call rejected manually.");
+ mutexIAX_.leave();
- return true;
+ removeCall (id);
+ }
}
@@ -453,9 +372,9 @@ IAXVoIPLink::carryingDTMFdigits (const std::string& id, char code)
{
IAXCall* call = getIAXCall (id);
if (call) {
- mutexIAX_.enterMutex();
- iax_send_dtmf (call->getSession(), code);
- mutexIAX_.leaveMutex();
+ mutexIAX_.enter();
+ iax_send_dtmf (call->session, code);
+ mutexIAX_.leave();
}
}
@@ -465,12 +384,11 @@ IAXVoIPLink::sendTextMessage (sfl::InstantMessaging *module,
const std::string& /*from*/)
{
IAXCall* call = getIAXCall (callID);
- if (!call)
- return;
-
- mutexIAX_.enterMutex();
- module->send_iax_message (call->getSession(), callID, message.c_str());
- mutexIAX_.leaveMutex();
+ if (call) {
+ mutexIAX_.enter();
+ module->send_iax_message (call->session, callID, message.c_str());
+ mutexIAX_.leave();
+ }
}
@@ -483,19 +401,12 @@ IAXVoIPLink::getCurrentCodecName(Call *c) const
}
-bool
+void
IAXVoIPLink::iaxOutgoingInvite (IAXCall* call)
{
ost::MutexLock m(mutexIAX_);
- iax_session *newsession = iax_session_new();
-
- if (!newsession) {
- _debug ("IAX Error: Can't make new session for a new call");
- return false;
- }
-
- call->setSession(newsession);
+ call->session = iax_session_new();
IAXAccount *account = dynamic_cast<IAXAccount *>(Manager::instance().getAccount(accountID_));
std::string username(account->getUsername());
@@ -505,46 +416,31 @@ IAXVoIPLink::iaxOutgoingInvite (IAXCall* call)
int audio_format_preferred = call->getFirstMatchingFormat(call->getSupportedFormat(accountID_), accountID_);
int audio_format_capability = call->getSupportedFormat(accountID_);
- _debug ("IAX New call: %s", strNum.c_str());
- iax_call(newsession, username.c_str(), username.c_str(), strNum.c_str(),
+ iax_call(call->session, username.c_str(), username.c_str(), strNum.c_str(),
NULL, 0, audio_format_preferred, audio_format_capability);
-
- return true;
}
IAXCall*
IAXVoIPLink::iaxFindCallBySession (struct iax_session* session)
{
- // access to callMap shoud use that
- // the code below is like findSIPCallWithCid()
ost::MutexLock m(_callMapMutex);
for (CallMap::const_iterator iter = _callMap.begin(); iter != _callMap.end(); ++iter) {
IAXCall* call = dynamic_cast<IAXCall*> (iter->second);
- if (call and call->getSession() == session)
+ if (call and call->session == session)
return call;
}
- return NULL; // not found
+ return NULL;
}
void
IAXVoIPLink::iaxHandleCallEvent (iax_event* event, IAXCall* call)
{
- // call should not be 0
- // note activity?
- //
std::string id = call->getCallId();
switch (event->etype) {
-
case IAX_EVENT_HANGUP:
-
- if (Manager::instance().isCurrentCall (id)) {
- // stop audio
- // audiolayer->stopStream();
- }
-
Manager::instance().peerHungupCall (id);
removeCall (id);
@@ -552,19 +448,14 @@ IAXVoIPLink::iaxHandleCallEvent (iax_event* event, IAXCall* call)
case IAX_EVENT_REJECT:
call->setConnectionState (Call::Connected);
-
call->setState (Call::Error);
Manager::instance().callFailure (id);
removeCall (id);
break;
case IAX_EVENT_ACCEPT:
- // Call accepted over there by the computer, not the user yet.
- _debug ("IAX_EVENT_ACCEPT: codec format: %d", event->ies.format);
-
if (event->ies.format)
- call->setFormat (event->ies.format);
-
+ call->format = event->ies.format;
break;
case IAX_EVENT_ANSWER:
@@ -576,19 +467,13 @@ IAXVoIPLink::iaxHandleCallEvent (iax_event* event, IAXCall* call)
call->setConnectionState (Call::Connected);
call->setState (Call::Active);
- // audiolayer->startStream();
-
- _debug ("IAX_EVENT_ANSWER: codec format: %d", event->ies.format);
-
- // Should not get here, should have been set in EVENT_ACCEPT
if (event->ies.format)
- call->setFormat (event->ies.format);
+ call->format = event->ies.format;
Manager::instance().peerAnsweredCall (id);
- // start audio here?
- audiolayer_->startStream();
- audiolayer_->flushMain();
+ Manager::instance().getAudioDriver()->startStream();
+ Manager::instance().getMainBuffer()->flushAllBuffers();
break;
@@ -600,9 +485,6 @@ IAXVoIPLink::iaxHandleCallEvent (iax_event* event, IAXCall* call)
break;
case IAX_EVENT_VOICE:
- //if (!audiolayer->isCaptureActive ())
- // audiolayer->startStream ();
- // _debug("IAX_EVENT_VOICE: ");
iaxHandleVoiceEvent (event, call);
break;
@@ -622,10 +504,8 @@ IAXVoIPLink::iaxHandleCallEvent (iax_event* event, IAXCall* call)
break;
case IAX_EVENT_URL:
-
- if (Manager::instance().getConfigString (HOOKS, URLHOOK_IAX2_ENABLED) == "1")
- UrlHook::runAction (Manager::instance().getConfigString (HOOKS, URLHOOK_COMMAND), (char*) event->data);
-
+ if (Manager::instance().getConfigString ("Hooks", "Hooks.iax2_enabled") == "1")
+ UrlHook::runAction (Manager::instance().getConfigString ("Hooks", "Hooks.url_command"), (char*) event->data);
break;
}
}
@@ -639,11 +519,6 @@ IAXVoIPLink::iaxHandleVoiceEvent (iax_event* event, IAXCall* call)
if (!event->datalen)
return;
- if (audiolayer_) {
- _debug ("IAX: incoming audio, but no sound card open");
- return;
- }
-
sfl::AudioCodec *audioCodec = static_cast<sfl::AudioCodec *>(Manager::instance().audioCodecFactory.getCodec(call->getAudioCodec()));
if (!audioCodec)
return;
@@ -651,35 +526,27 @@ IAXVoIPLink::iaxHandleVoiceEvent (iax_event* event, IAXCall* call)
Manager::instance().getMainBuffer()->setInternalSamplingRate(audioCodec->getClockRate());
unsigned int mainBufferSampleRate = Manager::instance().getMainBuffer()->getInternalSamplingRate();
- // On-the-fly codec changing (normally, when we receive a full packet)
- // as per http://tools.ietf.org/id/draft-guy-iax-03.txt
- // - subclass holds the voiceformat property.
-
- if (event->subclass && event->subclass != call->getFormat()) {
- _debug ("iaxHandleVoiceEvent: no format found in call setting it to %i", event->subclass);
- call->setFormat (event->subclass);
- }
+ if (event->subclass)
+ call->format = event->subclass;
unsigned char *data = (unsigned char*) event->data;
unsigned int size = event->datalen;
- // Decode data with relevant codec
unsigned int max = audioCodec->getClockRate() * 20 / 1000;
- if (size > max) {
- _debug ("The size %d is bigger than expected %d. Packet cropped. Ouch!", size, max);
+ if (size > max)
size = max;
- }
int samples = audioCodec->decode (decData, data , size);
int outSize = samples * sizeof(SFLDataFormat);
+ SFLDataFormat *out = decData;
unsigned int audioRate = audioCodec->getClockRate();
if (audioRate != mainBufferSampleRate) {
outSize = (double)outSize * (mainBufferSampleRate / audioRate);
- converter_->resample (decData, resampledData, mainBufferSampleRate, audioRate, samples);
- audiolayer_->getMainBuffer()->putData (resampledData, outSize, call->getCallId());
- } else
- audiolayer_->getMainBuffer()->putData (decData, outSize, call->getCallId());
+ converter_.resample (decData, resampledData, mainBufferSampleRate, audioRate, samples);
+ out = resampledData;
+ }
+ Manager::instance().getMainBuffer()->putData (out, outSize, call->getCallId());
}
/**
@@ -689,150 +556,69 @@ void
IAXVoIPLink::iaxHandleRegReply (iax_event* event)
{
IAXAccount *account = dynamic_cast<IAXAccount *>(Manager::instance().getAccount(accountID_));
+ if (event->etype != IAX_EVENT_REGREJ && event->etype != IAX_EVENT_REGACK)
+ return;
- if (event->etype == IAX_EVENT_REGREJ) {
- /* Authentication failed! */
- {
- ost::MutexLock m(mutexIAX_);
- iax_destroy(regSession_);
- }
- regSession_ = NULL;
- // Update the account registration state
- account->setRegistrationState (ErrorAuth);
- }
+ ost::MutexLock m(mutexIAX_);
+ iax_destroy(regSession_);
+ regSession_ = NULL;
- else if (event->etype == IAX_EVENT_REGACK) {
- /* Authentication succeeded */
- {
- ost::MutexLock m(mutexIAX_);
- iax_destroy (regSession_);
- }
- regSession_ = NULL;
+ account->setRegistrationState((event->etype == IAX_EVENT_REGREJ) ? ErrorAuth : Registered);
- // I mean, save the timestamp, so that we re-register again in the REFRESH time.
- // Defaults to 60, as per draft-guy-iax-03.
+ if (event->etype == IAX_EVENT_REGACK)
nextRefreshStamp_ = time (NULL) + (event->ies.refresh ? event->ies.refresh : 60);
- account->setRegistrationState (Registered);
- }
}
-int IAXVoIPLink::processIAXMsgCount (int msgcount)
-{
- // IAX sends the message count under a specific format:
- // 1
- // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
- // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- // | 0x18 | 0x02 |
- // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- // | Old messages | New messages |
- // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
- // For now we just need the new messages informations.
- // Thus:
- // 0 <= msgcount <= 255 => msgcount new messages, 0 old messages
- // msgcount >= 256 => msgcount/256 old messages , msgcount%256 new messages (RULES)
-
- return msgcount % 256;
-}
-
-
void
IAXVoIPLink::iaxHandlePrecallEvent (iax_event* event)
{
- IAXCall* call = NULL;
+ IAXCall *call;
std::string id;
+ int format;
switch (event->etype) {
- case IAX_EVENT_REGACK:
-
- case IAX_EVENT_REGREJ:
- _debug ("IAX Registration Event in a pre-call setup");
- break;
-
- case IAX_EVENT_REGREQ:
- // Received when someone wants to register to us!?!
- // Asterisk receives and answers to that, not us, we're a phone.
- _debug ("Registration by a peer, don't allow it");
- break;
-
case IAX_EVENT_CONNECT:
- // We've got an incoming call! Yikes!
- _debug ("> IAX_EVENT_CONNECT (receive)");
-
id = Manager::instance().getNewCallID();
call = new IAXCall (id, Call::Incoming);
-
- // Setup the new IAXCall
- // Associate the call to the session.
- call->setSession (event->session);
-
+ call->session = event->session;
call->setConnectionState (Call::Progressing);
-
if (event->ies.calling_number)
- call->setPeerNumber (std::string (event->ies.calling_number));
+ call->setPeerNumber(event->ies.calling_number);
if (event->ies.calling_name)
call->setPeerName (std::string (event->ies.calling_name));
// if peerNumber exist append it to the name string
call->initRecFileName (std::string (event->ies.calling_number));
+ Manager::instance().incomingCall(call, accountID_);
- if (Manager::instance().incomingCall (call, accountID_)) {
- /** @todo Faudra considérer éventuellement le champ CODEC PREFS pour
- * l'établissement du codec de transmission */
-
- // Remote lists its capabilities
- int format = call->getFirstMatchingFormat (event->ies.capability, accountID_);
- // Remote asks for preferred codec voiceformat
- int pref_format = call->getFirstMatchingFormat (event->ies.format, accountID_);
-
- // Priority to remote's suggestion. In case it's a forwarding, no transcoding
- // will be needed from the server, thus less latency.
-
- if (pref_format)
- format = pref_format;
-
- iax_accept (event->session, format);
-
- iax_ring_announce (event->session);
+ format = call->getFirstMatchingFormat (event->ies.format, accountID_);
+ if (!format)
+ format = call->getFirstMatchingFormat (event->ies.capability, accountID_);
- addCall (call);
-
- // Set the audio codec now, to avoid non-set codec problem later
- call->setFormat (format);
-
- } else {
- // reject call, unable to add it
- iax_reject (event->session, (char*) "Error ringing user.");
- delete call;
- }
+ iax_accept (event->session, format);
+ iax_ring_announce (event->session);
+ addCall (call);
+ call->format = format;
break;
case IAX_EVENT_HANGUP:
- // Remote peer hung up
- call = iaxFindCallBySession (event->session);
- id = call->getCallId();
- _debug ("IAXVoIPLink::hungup::iaxHandlePrecallEvent");
+ id = iaxFindCallBySession(event->session)->getCallId();
Manager::instance().peerHungupCall (id);
removeCall (id);
break;
case IAX_EVENT_TIMEOUT: // timeout for an unknown session
- break;
-
case IAX_IE_MSGCOUNT:
- break;
-
+ case IAX_EVENT_REGACK:
+ case IAX_EVENT_REGREJ:
+ case IAX_EVENT_REGREQ:
+ // Received when someone wants to register to us!?!
+ // Asterisk receives and answers to that, not us, we're a phone.
default:
- _debug ("IAXVoIPLink::iaxHandlePrecallEvent: Unknown event type (in precall): %d", event->etype);
+ break;
}
}
-
-void IAXVoIPLink::updateAudiolayer (void)
-{
- ost::MutexLock m(mutexIAX_);
- audiolayer_ = Manager::instance().getAudioDriver();
-}
diff --git a/daemon/src/iax/iaxvoiplink.h b/daemon/src/iax/iaxvoiplink.h
index ed8757dc17f2f1f8602aa8842001f13695ce5462..5dab2f2dcfdd0a72019a9f9c5cb2fb20c16cb4e7 100644
--- a/daemon/src/iax/iaxvoiplink.h
+++ b/daemon/src/iax/iaxvoiplink.h
@@ -36,12 +36,12 @@
#include <iax-client.h>
#include "audio/codecs/audiocodec.h" // for DEC_BUFFER_SIZE
#include "global.h"
+#include "audio/samplerateconverter.h"
namespace sfl {
class InstantMessaging;
}
-class SamplerateConverter;
class EventThread;
class IAXCall;
@@ -164,10 +164,8 @@ class IAXVoIPLink : public VoIPLink
/**
* Refuse a call
* @param id The ID of the call
- * @return bool true on success
- * false otherwise
*/
- virtual bool refuse (const std::string& id);
+ virtual void refuse (const std::string& id);
/**
* Send DTMF
@@ -185,11 +183,6 @@ class IAXVoIPLink : public VoIPLink
*/
virtual std::string getCurrentCodecName(Call *c) const;
-
- public: // iaxvoiplink only
-
- void updateAudiolayer (void);
-
private:
/*
* Decode the message count IAX send.
@@ -208,11 +201,6 @@ class IAXVoIPLink : public VoIPLink
*/
IAXCall* getIAXCall (const std::string& id);
- /**
- * Delete every call
- */
- void terminateIAXCall();
-
/**
* Find a iaxcall by iax session number
* @param session an iax_session valid pointer
@@ -255,7 +243,7 @@ class IAXVoIPLink : public VoIPLink
* Send an outgoing call invite to iax
* @param call An IAXCall pointer
*/
- bool iaxOutgoingInvite (IAXCall* call);
+ void iaxOutgoingInvite (IAXCall* call);
/** Threading object */
EventThread* evThread_;
@@ -272,17 +260,13 @@ class IAXVoIPLink : public VoIPLink
* iax_stuff inside this class. */
ost::Mutex mutexIAX_;
- /** Connection to audio card/device */
- AudioLayer* audiolayer_;
-
/** encoder/decoder/resampler buffers */
SFLDataFormat decData[DEC_BUFFER_SIZE];
SFLDataFormat resampledData[DEC_BUFFER_SIZE];
unsigned char encodedData[DEC_BUFFER_SIZE];
- int converterSamplingRate_;
/** Sample rate converter object */
- SamplerateConverter* converter_;
+ SamplerateConverter converter_;
/** Whether init() was called already or not
* This should be used in init() and terminate(), to
diff --git a/daemon/src/managerimpl.cpp b/daemon/src/managerimpl.cpp
index 1724cba6b26af62fd42764aab36fbe67a3789afb..222da33d093082b4f02ccff21cd220aff3b378e6 100644
--- a/daemon/src/managerimpl.cpp
+++ b/daemon/src/managerimpl.cpp
@@ -585,10 +585,8 @@ bool ManagerImpl::attendedTransfer(const std::string& transferID, const std::str
}
//THREAD=Main : Call:Incoming
-bool ManagerImpl::refuseCall (const std::string& id)
+void ManagerImpl::refuseCall (const std::string& id)
{
- bool returnValue;
-
stopTone();
if (getCallList().size() <= 1) {
@@ -600,31 +598,25 @@ bool ManagerImpl::refuseCall (const std::string& id)
/* Direct IP to IP call */
if (getConfigFromCall (id) == Call::IPtoIP)
- returnValue = SIPVoIPLink::instance ()-> refuse (id);
+ SIPVoIPLink::instance()->refuse (id);
else {
/* Classic call, attached to an account */
std::string accountid = getAccountFromCall (id);
if (accountid.empty())
- return false;
+ return;
- returnValue = getAccountLink (accountid)->refuse (id);
+ getAccountLink(accountid)->refuse(id);
removeCallAccount (id);
}
- // if the call was outgoing or established, we didn't refuse it
- // so the method did nothing
- if (returnValue) {
- removeWaitingCall (id);
- _dbus.getCallManager()->callStateChanged (id, "HUNGUP");
- }
+ removeWaitingCall(id);
+ _dbus.getCallManager()->callStateChanged(id, "HUNGUP");
// Disconnect streams
removeStream(id);
getMainBuffer()->stateInfo();
-
- return returnValue;
}
Conference*
@@ -1361,8 +1353,6 @@ bool ManagerImpl::incomingCallWaiting ()
void ManagerImpl::addWaitingCall (const std::string& id)
{
- _info ("Manager: Add waiting call %s (%d calls)", id.c_str(), _nbIncomingWaitingCall);
-
ost::MutexLock m(_waitingCallMutex);
_waitingCall.insert (id);
_nbIncomingWaitingCall++;
@@ -1370,12 +1360,8 @@ void ManagerImpl::addWaitingCall (const std::string& id)
void ManagerImpl::removeWaitingCall (const std::string& id)
{
- _info ("Manager: Remove waiting call %s (%d calls)", id.c_str(), _nbIncomingWaitingCall);
-
ost::MutexLock m(_waitingCallMutex);
- // should return more than 1 if it erase a call
-
- if (_waitingCall.erase (id))
+ if (_waitingCall.erase(id))
_nbIncomingWaitingCall--;
}
@@ -1388,17 +1374,14 @@ bool ManagerImpl::isWaitingCall (const std::string& id)
// Management of event peer IP-phone
////////////////////////////////////////////////////////////////////////////////
// SipEvent Thread
-bool ManagerImpl::incomingCall (Call* call, const std::string& accountId)
+void ManagerImpl::incomingCall (Call* call, const std::string& accountId)
{
assert(call);
stopTone();
- _debug ("Manager: Incoming call %s for account %s", call->getCallId().data(), accountId.c_str());
-
associateCallToAccount (call->getCallId(), accountId);
- // If account is null it is an ip to ip call
- if (accountId.empty())
+ if (accountId == "")
associateConfigToCall (call->getCallId(), Call::IPtoIP);
else {
// strip sip: which is not required and bring confusion with ip to ip calls
@@ -1407,45 +1390,26 @@ bool ManagerImpl::incomingCall (Call* call, const std::string& accountId)
size_t startIndex = peerNumber.find ("sip:");
- if (startIndex != std::string::npos) {
- std::string strippedPeerNumber = peerNumber.substr (startIndex + 4);
- call->setPeerNumber (strippedPeerNumber);
- }
+ if (startIndex != std::string::npos)
+ call->setPeerNumber (peerNumber.substr (startIndex + 4));
}
if (!hasCurrentCall()) {
- _debug ("Manager: Has no current call start ringing");
call->setConnectionState (Call::Ringing);
ringtone (accountId);
-
- } else
- _debug ("Manager: has current call, beep in current audio stream");
+ }
addWaitingCall (call->getCallId());
std::string from(call->getPeerName());
std::string number(call->getPeerNumber());
- std::string display_name(call->getDisplayName());
- if (not from.empty() and not number.empty()) {
- from.append (" <");
- from.append (number);
- from.append (">");
- } else if (from.empty()) {
- from.append ("<");
- from.append (number);
- from.append (">");
- }
-
- /* Broadcast a signal over DBus */
- _debug ("Manager: From: %s, Number: %s, Display Name: %s", from.c_str(), number.c_str(), display_name.c_str());
+ if (not from.empty() and not number.empty())
+ from += " ";
- std::string display(display_name);
- display.append(" ");
- display.append(from);
+ from += "<" + number + ">";
- _dbus.getCallManager()->incomingCall(accountId, call->getCallId(), display);
- return true;
+ _dbus.getCallManager()->incomingCall(accountId, call->getCallId(), call->getDisplayName() + " " + from);
}
diff --git a/daemon/src/managerimpl.h b/daemon/src/managerimpl.h
index 616def92beddd00191c4882e52dcf60b41bca0bb..024ec8992d91b70d2db5698e65421c8613796a47 100644
--- a/daemon/src/managerimpl.h
+++ b/daemon/src/managerimpl.h
@@ -231,7 +231,7 @@ class ManagerImpl
* Refuse the call
* @param id The call identifier
*/
- bool refuseCall (const std::string& id);
+ void refuseCall (const std::string& id);
/**
* Create a new conference given two participant
@@ -365,9 +365,8 @@ class ManagerImpl
* and notify user
* @param call A call pointer
* @param accountId an account id
- * @return bool True if the call was added correctly
*/
- bool incomingCall (Call* call, const std::string& accountId);
+ void incomingCall (Call* call, const std::string& accountId);
/**
* Notify the user that the recipient of the call has answered and the put the
diff --git a/daemon/src/sip/sipvoiplink.cpp b/daemon/src/sip/sipvoiplink.cpp
index 53594c6a6258e2235e8e2cc05ee28c31dcbdc1b9..decbbf4474160f9864ba3c246f4b9fc404b9fa66 100644
--- a/daemon/src/sip/sipvoiplink.cpp
+++ b/daemon/src/sip/sipvoiplink.cpp
@@ -667,28 +667,26 @@ bool SIPVoIPLink::attendedTransfer(const std::string& id, const std::string& to)
return transferCommon(getSIPCall(id), &dst);
}
-bool
+void
SIPVoIPLink::refuse (const std::string& id)
{
SIPCall *call = getSIPCall (id);
if (!call->isIncoming() or call->getConnectionState() == Call::Connected)
- return false;
+ return;
call->getAudioRtp()->stop();
pjsip_tx_data *tdata;
if (pjsip_inv_end_session (call->inv, PJSIP_SC_DECLINE, NULL, &tdata) != PJ_SUCCESS)
- return false;
+ return;
if (pjsip_inv_send_msg (call->inv, tdata) != PJ_SUCCESS)
- return false;
+ return;
// Make sure the pointer is NULL in callbacks
call->inv->mod_data[_mod_ua.id] = NULL;
removeCall (id);
-
- return true;
}
std::string
@@ -1730,8 +1728,12 @@ static pj_bool_t transaction_request_cb (pjsip_rx_data *rdata)
call->getAudioRtp()->start (static_cast<sfl::AudioCodec *>(audiocodec));
pjsip_dialog* dialog;
- if (pjsip_dlg_create_uas (pjsip_ua_instance(), rdata, NULL, &dialog) != PJ_SUCCESS)
- goto fail;
+ if (pjsip_dlg_create_uas (pjsip_ua_instance(), rdata, NULL, &dialog) != PJ_SUCCESS) {
+ delete call;
+ pjsip_endpt_respond_stateless (_endpt, rdata, PJSIP_SC_INTERNAL_SERVER_ERROR, NULL, NULL, NULL);
+ return false;
+
+ }
pjsip_inv_create_uas (dialog, rdata, call->getLocalSDP()->getLocalSdpSession(), 0, &call->inv);
@@ -1767,18 +1769,11 @@ static pj_bool_t transaction_request_cb (pjsip_rx_data *rdata)
call->setConnectionState (Call::Ringing);
- if (!Manager::instance().incomingCall (call, account_id))
- goto fail;
-
+ Manager::instance().incomingCall (call, account_id);
Manager::instance().getAccountLink (account_id)->addCall (call);
}
return true;
-
-fail:
- delete call;
- pjsip_endpt_respond_stateless (_endpt, rdata, PJSIP_SC_INTERNAL_SERVER_ERROR, NULL, NULL, NULL);
- return false;
}
static pj_bool_t transaction_response_cb (pjsip_rx_data *rdata)
diff --git a/daemon/src/sip/sipvoiplink.h b/daemon/src/sip/sipvoiplink.h
index 4d50d830ef710a287c78e422f17fc84bee65e9a5..edaa5e2fb9fc799e2172922be89650451d83fbb4 100644
--- a/daemon/src/sip/sipvoiplink.h
+++ b/daemon/src/sip/sipvoiplink.h
@@ -164,9 +164,8 @@ class SIPVoIPLink : public VoIPLink
/**
* Refuse the call
* @param id The call identifier
- * @return bool True on success
*/
- virtual bool refuse (const std::string& id);
+ virtual void refuse (const std::string& id);
/**
* Send DTMF refering to account configuration
diff --git a/daemon/src/voiplink.h b/daemon/src/voiplink.h
index 35f7a368e02da17fce3f82a94201e25d88065c0d..0936e3c4630edf4ae044d649f3731b87ef00a57a 100644
--- a/daemon/src/voiplink.h
+++ b/daemon/src/voiplink.h
@@ -148,9 +148,8 @@ class VoIPLink
/**
* Refuse incoming call
* @param id The call identifier
- * @return bool True on success
*/
- virtual bool refuse (const std::string& id) = 0;
+ virtual void refuse (const std::string& id) = 0;
/**
* Send DTMF