From f2bc81a0f2f94e81b29a582f62fc6cf22af9aa1b Mon Sep 17 00:00:00 2001
From: yanmorin <yanmorin>
Date: Mon, 7 Aug 2006 21:22:56 +0000
Subject: [PATCH] Resolving bug when calling the same account (adding codec map
to new incoming call)
Adding debugging information
---
src/audio/audiolayer.cpp | 21 ++--
src/audio/audiortp.cpp | 28 ++---
src/global.h | 6 +
src/gui/guiframework.cpp | 26 -----
src/gui/guiframework.h | 1 -
src/gui/server/guiserverimpl.cpp | 9 --
src/managerimpl.cpp | 37 +++----
src/managerimpl.h | 21 ----
src/sipcall.cpp | 2 +-
src/sipvoiplink.cpp | 182 +++++++++++++++----------------
10 files changed, 141 insertions(+), 192 deletions(-)
diff --git a/src/audio/audiolayer.cpp b/src/audio/audiolayer.cpp
index c6b50cd86e..827799453a 100644
--- a/src/audio/audiolayer.cpp
+++ b/src/audio/audiolayer.cpp
@@ -45,7 +45,7 @@ AudioLayer::~AudioLayer (void)
try {
portaudio::System::terminate();
} catch (const portaudio::PaException &e) {
- _debug("Catch an exception when portaudio tried to terminate\n");
+ _debug("! AL: Catch an exception when portaudio tried to terminate\n");
}
closeStream();
}
@@ -112,15 +112,16 @@ AudioLayer::startStream(void)
try {
ost::MutexLock guard(_mutex);
if (_stream && !_stream->isActive()) {
- _debug("Starting sound stream\n");
+ _debug("- AL Action: Starting sound stream\n");
_stream->start();
- }
- else { _debug ("stream doesn't exist or is already active\n"); }
+ } else {
+ _debug ("* AL Info: Stream doesn't exist or is already active\n");
+ }
} catch (const portaudio::PaException &e) {
- _debugException("Portaudio error: error on starting audiolayer stream");
+ _debugException("! AL: Portaudio error: error on starting audiolayer stream");
throw;
} catch(...) {
- _debugException("stream start error");
+ _debugException("! AL: Stream start error");
throw;
}
}
@@ -137,10 +138,10 @@ AudioLayer::stopStream(void)
_micRingBuffer.flush();
}
} catch (const portaudio::PaException &e) {
- _debugException("Portaudio error: stoping audiolayer stream failed");
+ _debugException("! AL: Portaudio error: stoping audiolayer stream failed");
throw;
} catch(...) {
- _debugException("stream stop error");
+ _debugException("! AL: Stream stop error");
throw;
}
}
@@ -162,7 +163,7 @@ AudioLayer::isStreamActive (void)
return true;
}
} catch (const portaudio::PaException &e) {
- _debugException("Portaudio error: isActive returned an error");
+ _debugException("! AL: Portaudio error: isActive returned an error");
}
return false;
}
@@ -239,7 +240,7 @@ AudioLayer::isStreamStopped (void)
return true;
}
} catch (const portaudio::PaException &e) {
- _debugException("Portaudio error: isStopped returned an exception");
+ _debugException("! AL: Portaudio error: isStopped returned an exception");
}
return false;
}
diff --git a/src/audio/audiortp.cpp b/src/audio/audiortp.cpp
index 095829f826..93d2f0d476 100644
--- a/src/audio/audiortp.cpp
+++ b/src/audio/audiortp.cpp
@@ -54,7 +54,7 @@ AudioRtp::createNewSession (SIPCall *ca) {
// something should stop the thread before...
if ( _RTXThread != 0 ) {
- _debug("AudioRTP Failure: Thread already exists..., stopping it\n");
+ _debug("! ARTP Failure: Thread already exists..., stopping it\n");
delete _RTXThread; _RTXThread = 0;
//return -1;
}
@@ -65,11 +65,11 @@ AudioRtp::createNewSession (SIPCall *ca) {
try {
if (_RTXThread->start() != 0) {
- _debug("AudioRTP Failure: unable to start RTX Thread\n");
+ _debug("! ARTP Failure: unable to start RTX Thread\n");
return -1;
}
} catch(...) {
- _debugException("AudioRTP try to start a thread");
+ _debugException("! ARTP Failure: when trying to start a thread");
throw;
}
return 0;
@@ -84,7 +84,7 @@ AudioRtp::closeRtpSession () {
try {
delete _RTXThread; _RTXThread = 0;
} catch(...) {
- _debugException("Exception when stopping audiortp\n");
+ _debugException("! ARTP Exception: when stopping audiortp\n");
throw;
}
}
@@ -133,7 +133,7 @@ AudioRtpRTX::~AudioRtpRTX () {
try {
this->terminate();
} catch(...) {
- _debugException("AudioRTP Thread destructor didn't terminate correctly");
+ _debugException("! ARTP: Thread destructor didn't terminate correctly");
throw;
}
//_debug("terminate audiortprtx ended...\n");
@@ -164,7 +164,7 @@ AudioRtpRTX::initAudioRtpSession (void)
//_debug("Init audio RTP session\n");
ost::InetHostAddress remote_ip(_ca->getRemoteIp().c_str());
if (!remote_ip) {
- _debug("AudioRTP Thread Error: Target IP address [%s] is not correct!\n", _ca->getRemoteIp().data());
+ _debug("! ARTP Thread Error: Target IP address [%s] is not correct!\n", _ca->getRemoteIp().data());
return;
}
@@ -186,7 +186,7 @@ AudioRtpRTX::initAudioRtpSession (void)
return;
}
if (!_sessionSend->addDestination (remote_ip, (unsigned short) _ca->getRemoteAudioPort())) {
- _debug("AudioRTP Thread Error: could not connect to port %d\n", _ca->getRemoteAudioPort());
+ _debug("! ARTP Thread Error: could not connect to port %d\n", _ca->getRemoteAudioPort());
return;
}
@@ -220,7 +220,7 @@ AudioRtpRTX::initAudioRtpSession (void)
}
}
} catch(...) {
- _debugException("AudioRTP initialisation failed");
+ _debugException("! ARTP Failure: initialisation failed");
throw;
}
}
@@ -291,7 +291,7 @@ AudioRtpRTX::sendSessionFromMic (unsigned char* data_to_send, int16* data_from_m
int compSize = audiocodec->codecEncode(data_to_send, data_from_mic_to_codec, toSize*sizeOfData);
// encode divise by two
// Send encoded audio sample over the network
- if (compSize > RTP_FRAMES2SEND) { _debug("%d should be %d\n", compSize, _nbFrames);}
+ if (compSize > RTP_FRAMES2SEND) { _debug("! ARTP: %d should be %d\n", compSize, _nbFrames);}
//fprintf(stderr, "S");
if (!_sym) {
_sessionSend->putData(timestamp, data_to_send, compSize);
@@ -300,7 +300,7 @@ AudioRtpRTX::sendSessionFromMic (unsigned char* data_to_send, int16* data_from_m
}
}
} catch(...) {
- _debugException("AudioRTP sending failed");
+ _debugException("! ARTP: sending failed");
throw;
}
}
@@ -382,7 +382,7 @@ AudioRtpRTX::receiveSessionForSpkr (int16* data_for_speakers_stereo, int16* data
delete adu; adu = NULL;
} catch(...) {
- _debugException("AudioRTP receiving failed");
+ _debugException("! ARTP: receiving failed");
throw;
}
}
@@ -430,7 +430,7 @@ AudioRtpRTX::run () {
audiolayer->flushMic();
audiolayer->startStream();
_start.post();
- _debug("AudioRTP Start\n");
+ _debug("- ARTP Action: Start\n");
while (!testCancel()) {
////////////////////////////
// Send session
@@ -451,11 +451,11 @@ AudioRtpRTX::run () {
audiolayer->stopStream();
} catch(std::exception &e) {
_start.post();
- _debug("AudioRTP Stop: %s\n", e.what());
+ _debug("! ARTP: Stop %s\n", e.what());
throw;
} catch(...) {
_start.post();
- _debugException("AudioRTP Stop");
+ _debugException("* ARTP Action: Stop");
throw;
}
delete [] data_for_speakers_stereo; data_for_speakers_stereo = 0;
diff --git a/src/global.h b/src/global.h
index 19ca5e9890..70457e9a62 100644
--- a/src/global.h
+++ b/src/global.h
@@ -29,10 +29,16 @@ typedef short int16;
#ifdef DEBUG
#define _debug(...) fprintf(stderr, "[sfl-debug] " __VA_ARGS__)
+ #define _debugStart(...) fprintf(stderr, "[sfl-debug] " __VA_ARGS__)
+ #define _debugMid(...) fprintf(stderr, __VA_ARGS__)
+ #define _debugEnd(...) fprintf(stderr, __VA_ARGS__)
#define _debugException(...) fprintf(stderr, "[sfl-excep] " __VA_ARGS__ "\n")
#define _debugInit(...) fprintf(stderr, "[sfl-init.] " __VA_ARGS__ "\n")
#else
#define _debug(...)
+ #define _debugStart(...)
+ #define _debugMid(...)
+ #define _debugEnd(...)
#define _debugException(...)
#define _debugInit(...)
#endif
diff --git a/src/gui/guiframework.cpp b/src/gui/guiframework.cpp
index b902b95d47..3c8d49437c 100644
--- a/src/gui/guiframework.cpp
+++ b/src/gui/guiframework.cpp
@@ -242,32 +242,6 @@ GuiFramework::hasLoadedSetup()
return Manager::instance().hasLoadedSetup();
}
-bool
-GuiFramework::getRegistrationState(std::string& stateCode, std::string& stateMessage) {
- ManagerImpl::REGISTRATION_STATE state = Manager::instance().getRegistrationState();
- bool returnValue = false;
- switch( state ) {
- case ManagerImpl::REGISTERED:
- returnValue = true;
- stateCode = "103";
- stateMessage = _("Registration succeed");
- break;
-
- case ManagerImpl::FAILED:
- returnValue = true;
- stateCode = "104";
- stateMessage = _("Registration failed");
- break;
-
- case ManagerImpl::UNREGISTERED:
- returnValue = false;
- stateCode = "";
- stateMessage = "";
- break;
- }
- return returnValue;
-}
-
bool
GuiFramework::setSwitch(const std::string& switchName, std::string& returnMessage)
{
diff --git a/src/gui/guiframework.h b/src/gui/guiframework.h
index 2f8daa75ea..31b209b087 100644
--- a/src/gui/guiframework.h
+++ b/src/gui/guiframework.h
@@ -97,7 +97,6 @@ public:
bool hasLoadedSetup();
const CallID& getCurrentId();
- bool getRegistrationState(std::string& stateCode, std::string& stateMessage);
protected:
std::string _message;
diff --git a/src/gui/server/guiserverimpl.cpp b/src/gui/server/guiserverimpl.cpp
index a050875fb8..b576e5e690 100644
--- a/src/gui/server/guiserverimpl.cpp
+++ b/src/gui/server/guiserverimpl.cpp
@@ -112,17 +112,8 @@ GUIServerImpl::getEvents(const std::string& sequenceId)
tk.push_back(_("Microphone volume changed"));
_requestManager.sendResponse(ResponseMessage("022", sequenceId, tk));
- std::string stateCode;
- std::string stateMessage;
-
// try to register, if not done yet...
GuiFramework::getEvents();
-
- if (GuiFramework::getRegistrationState(stateCode,stateMessage)) {
- _requestManager.sendResponse(ResponseMessage(stateCode, sequenceId, stateMessage));
- }
-
-
return true;
}
diff --git a/src/managerimpl.cpp b/src/managerimpl.cpp
index 9ae3e6c30f..39a4ad1d5e 100644
--- a/src/managerimpl.cpp
+++ b/src/managerimpl.cpp
@@ -56,8 +56,6 @@
ManagerImpl::ManagerImpl (void)
{
// Init private variables
- //_error = new Error();
-
_hasZeroconf = false;
#ifdef USE_ZEROCONF
_hasZeroconf = true;
@@ -73,13 +71,12 @@ ManagerImpl::ManagerImpl (void)
// sound
_audiodriverPA = NULL;
_dtmfKey = 0;
- _spkr_volume = 0; // Initialize after by init() -> initVolume()
+ _spkr_volume = 0; // Initialize after by init() -> initVolume()
_mic_volume = 0; // Initialize after by init() -> initVolume()
_mic_volume_before_mute = 0;
// Call
_nbIncomingWaitingCall=0;
- _registerState = UNREGISTERED;
_hasTriedToRegister = false;
// initialize random generator for call id
@@ -209,24 +206,24 @@ bool
ManagerImpl::outgoingCall(const std::string& accountid, const CallID& id, const std::string& to)
{
if (!accountExists(accountid)) {
- _debug("Outgoing Call: account doesn't exist\n");
+ _debug("! Manager Error: Outgoing Call: account doesn't exist\n");
return false;
}
if (getAccountFromCall(id) != AccountNULL) {
- _debug("Outgoing Call: call id already exists\n");
+ _debug("! Manager Error: Outgoing Call: call id already exists\n");
return false;
}
if (hasCurrentCall()) {
- _debug("There is currently a call, try to hold it\n");
+ _debug("* Manager Info: there is currently a call, try to hold it\n");
onHoldCall(getCurrentCallId());
}
- _debug("Adding Outgoing Call %s on account %s\n", id.data(), accountid.data());
+ _debug("- Manager Action: Adding Outgoing Call %s on account %s\n", id.data(), accountid.data());
if ( getAccountLink(accountid)->newOutgoingCall(id, to) ) {
associateCallToAccount( id, accountid );
switchCall(id);
return true;
} else {
- _debug("An error occur, the call was not created\n");
+ _debug("! Manager Error: An error occur, the call was not created\n");
}
return false;
}
@@ -273,7 +270,7 @@ ManagerImpl::hangupCall(const CallID& id)
AccountID accountid = getAccountFromCall( id );
if (accountid == AccountNULL) {
- _debug("Hangup Call: Call doesn't exists\n");
+ _debug("! Manager Hangup Call: Call doesn't exists\n");
return false;
}
@@ -291,7 +288,7 @@ ManagerImpl::cancelCall (const CallID& id)
stopTone();
AccountID accountid = getAccountFromCall( id );
if (accountid == AccountNULL) {
- _debug("Cancel Call: Call doesn't exists\n");
+ _debug("! Manager Cancel Call: Call doesn't exists\n");
return false;
}
@@ -311,7 +308,7 @@ ManagerImpl::onHoldCall(const CallID& id)
stopTone();
AccountID accountid = getAccountFromCall( id );
if (accountid == AccountNULL) {
- _debug("On Hold Call: Call doesn't exists\n");
+ _debug("5 Manager On Hold Call: Account ID %s or callid %s desn't exists\n", accountid.c_str(), id.c_str());
return false;
}
@@ -329,7 +326,7 @@ ManagerImpl::offHoldCall(const CallID& id)
stopTone();
AccountID accountid = getAccountFromCall( id );
if (accountid == AccountNULL) {
- _debug("OffHold Call: Call doesn't exists\n");
+ _debug("5 Manager OffHold Call: Call doesn't exists\n");
return false;
}
bool returnValue = getAccountLink(accountid)->offhold(id);
@@ -339,7 +336,7 @@ ManagerImpl::offHoldCall(const CallID& id)
try {
getAudioDriver()->startStream();
} catch(...) {
- _debugException("Off hold could not start audio stream");
+ _debugException("! Manager Off hold could not start audio stream");
}
}
return returnValue;
@@ -352,7 +349,7 @@ ManagerImpl::transferCall(const CallID& id, const std::string& to)
stopTone();
AccountID accountid = getAccountFromCall( id );
if (accountid == AccountNULL) {
- _debug("Transfer Call: Call doesn't exists\n");
+ _debug("! Manager Transfer Call: Call doesn't exists\n");
return false;
}
bool returnValue = getAccountLink(accountid)->transfer(id, to);
@@ -385,7 +382,7 @@ ManagerImpl::refuseCall (const CallID& id)
stopTone();
AccountID accountid = getAccountFromCall( id );
if (accountid == AccountNULL) {
- _debug("OffHold Call: Call doesn't exists\n");
+ _debug("! Manager OffHold Call: Call doesn't exists\n");
return false;
}
bool returnValue = getAccountLink(accountid)->refuse(id);
@@ -415,7 +412,7 @@ ManagerImpl::saveConfig (void)
bool
ManagerImpl::initRegisterVoIPLink()
{
- _debugInit("Initiate VoIP Links Registration\n");
+ _debugInit("Initiate VoIP Links Registration");
AccountMap::iterator iter = _accountMap.begin();
while( iter != _accountMap.end() ) {
if ( iter->second) {
@@ -1682,6 +1679,7 @@ ManagerImpl::getNewCallID()
short
ManagerImpl::loadAccountMap()
{
+ _debugStart("Load account:");
short nbAccount = 0;
Account* tmpAccount;
@@ -1693,7 +1691,7 @@ ManagerImpl::loadAccountMap()
tmpAccount = AccountCreator::createAccount(AccountCreator::SIP_ACCOUNT, accountName.str());
if (tmpAccount!=0) {
- _debug("Adding Account: %s\n", accountName.str().data());
+ _debugMid(" %s", accountName.str().data());
_accountMap[accountName.str()] = tmpAccount;
nbAccount++;
}
@@ -1706,11 +1704,12 @@ ManagerImpl::loadAccountMap()
accountName << "IAX" << iAccountIAX;
tmpAccount = AccountCreator::createAccount(AccountCreator::IAX_ACCOUNT, accountName.str());
if (tmpAccount!=0) {
- _debug("Adding Account: %s\n", accountName.str().data());
+ _debugMid(" %s", accountName.str().data());
_accountMap[accountName.str()] = tmpAccount;
nbAccount++;
}
}
+ _debugEnd("\n");
return nbAccount;
}
diff --git a/src/managerimpl.h b/src/managerimpl.h
index a9d23e7a4b..6ce7571ac0 100644
--- a/src/managerimpl.h
+++ b/src/managerimpl.h
@@ -51,17 +51,6 @@ class VoIPLink;
class DNSService;
#endif
-// Status
-//#define CONNECTED_STATUS "Connected"
-//#define LOGGED_IN_STATUS "Logged in"
-//#define RINGING_STATUS "Ringing"
-//#define TRYING_STATUS "Trying ..."
-//#define HANGUP_STATUS "Hang up"
-//#define ONHOLD_STATUS "On hold ..."
-//#define TRANSFER_STATUS "Transfer to:"
-//#define MUTE_ON_STATUS "Mute on"
-//#define ENTER_NUMBER_STATUS "Enter Phone Number:"
-
/**
* Define a type for a AccountMap container
*/
@@ -298,14 +287,6 @@ public:
*/
bool hasLoadedSetup() { return _setupLoaded; }
- enum REGISTRATION_STATE {
- UNREGISTERED,
- REGISTERED,
- FAILED
- };
-
- REGISTRATION_STATE getRegistrationState() { return _registerState; }
-
/** Return a new random callid that is not present in the list
* @return a brand new callid
*/
@@ -475,8 +456,6 @@ private:
bool initRegisterVoIPLink();
// true if we tried to register Once
bool _hasTriedToRegister;
- // Register state
- REGISTRATION_STATE _registerState;
// tell if we have zeroconf is enabled
int _hasZeroconf;
diff --git a/src/sipcall.cpp b/src/sipcall.cpp
index cb3fde0926..5225ed1bc7 100644
--- a/src/sipcall.cpp
+++ b/src/sipcall.cpp
@@ -420,7 +420,7 @@ SIPCall::sdp_complete_message(sdp_message_t * remote_sdp, osip_message_t * msg)
AudioCodec* audiocodec = _codecMap.getCodec((CodecType)payload);
if (audiocodec!=0 && audiocodec->isActive()) {
listCodec << payload << " ";
- listRtpMap << "a=rtpmap: " << payload << " " << audiocodec->getCodecName() << "/" << audiocodec->getClockRate();
+ listRtpMap << "a=rtpmap:" << payload << " " << audiocodec->getCodecName() << "/" << audiocodec->getClockRate();
if ( audiocodec->getChannel() != 1) {
listRtpMap << "/" << audiocodec->getChannel();
}
diff --git a/src/sipvoiplink.cpp b/src/sipvoiplink.cpp
index b1b2a8b788..d8db053c37 100644
--- a/src/sipvoiplink.cpp
+++ b/src/sipvoiplink.cpp
@@ -76,14 +76,14 @@ SIPVoIPLink::init()
{
if (!_eXosipStarted) {
if (0 != eXosip_init()) {
- _debug("Could not initialize eXosip\n");
+ _debug("! SIP Failure: Could not initialize eXosip\n");
return false;
}
_eXosipStarted = true;
// check networking capabilities
if ( !checkNetwork() ) {
- _debug("SIP FAILURE: Unable to determine network capabilities\n");
+ _debug("! SIP FAILURE: Unable to determine network capabilities\n");
return false;
}
@@ -98,7 +98,7 @@ SIPVoIPLink::init()
if (_useStun && !Manager::instance().behindNat(_stunServer, port)) {
port = RANDOM_SIP_PORT;
if (!Manager::instance().behindNat(_stunServer, port)) {
- _debug("SIP FAILURE: Unable to check NAT setting\n");
+ _debug("! SIP Failure: Unable to check NAT setting\n");
return false; // hoho we can't use the random sip port too...
}
}
@@ -106,17 +106,17 @@ SIPVoIPLink::init()
// second parameter, NULL is "::" for ipv6 and "0.0.0.0" for ipv4, we can put INADDR_ANY
errExosip = eXosip_listen_addr(IPPROTO_UDP, INADDR_ANY, port, AF_INET, 0);
if (errExosip != 0) {
- _debug("SIP ERROR: [%d/%d] could not initialize SIP listener on port %d\n", iTry, _nbTryListenAddr, port);
+ _debug("* SIP Info: [%d/%d] could not initialize SIP listener on port %d\n", iTry, _nbTryListenAddr, port);
port = RANDOM_SIP_PORT;
}
} while ( errExosip != 0 && iTry < _nbTryListenAddr );
if ( errExosip != 0 ) { // we didn't succeeded
- _debug("SIP FAILURE: SIP failed to listen on port %d\n", port);
+ _debug("! SIP Failure: SIP failed to listen on port %d\n", port);
return false;
}
_localPort = port;
- _debug("SIP Init: listening on port %d\n", port);
+ _debug(" SIP Init: listening on port %d\n", port);
if (_useStun) {
// This method is used to replace contact address with the public address of your NAT
@@ -134,7 +134,7 @@ SIPVoIPLink::init()
std::string tmp = std::string(PROGNAME_GLOBAL) + "/" + std::string(SFLPHONED_VERSION);
eXosip_set_user_agent(tmp.data());
- _debug("SIP Init: starting loop thread (SIP events)\n");
+ _debug(" SIP Init: starting loop thread (SIP events)\n");
_evThread->start();
}
return _eXosipStarted;
@@ -189,7 +189,7 @@ SIPVoIPLink::loadSIPLocalIP()
returnValue = false;
} else {
_localIPAddress = std::string(myIPAddress);
- _debug("Checking network, setting local IP address to: %s\n", myIPAddress);
+ _debug(" SIP Info: Checking network, setting local IP address to: %s\n", myIPAddress);
}
delete [] myIPAddress; myIPAddress = NULL;
}
@@ -208,11 +208,11 @@ SIPVoIPLink::getEvent()
return;
}
- _debug("SIP Event: [cdt=%4d:%4d:%4d] type=#%03d %s \n", event->cid, event->did, event->tid, event->type, event->textinfo);
+ _debug("> SIP Event: [cdt=%4d:%4d:%4d] type=#%03d %s \n", event->cid, event->did, event->tid, event->type, event->textinfo);
switch (event->type) {
/* REGISTER related events */
case EXOSIP_REGISTRATION_NEW: /** 00 < announce new registration. */
- _debug("EXOSIP_REGISTRATION_NEW event is not implemented\n");
+ _debug(" !EXOSIP_REGISTRATION_NEW event is not implemented\n");
break;
case EXOSIP_REGISTRATION_SUCCESS: /** 01 < user is successfully registred. */
Manager::instance().registrationSucceed(getAccountID());
@@ -221,10 +221,10 @@ SIPVoIPLink::getEvent()
Manager::instance().registrationFailed(getAccountID());
break;
case EXOSIP_REGISTRATION_REFRESHED: /** 03 < registration has been refreshed. */
- _debug("EXOSIP_REGISTRATION_REFRESHED event is not implemented\n");
+ _debug(" !EXOSIP_REGISTRATION_REFRESHED event is not implemented\n");
break;
case EXOSIP_REGISTRATION_TERMINATED: /** 04 < UA is not registred any more. */
- _debug("EXOSIP_REGISTRATION_TERMINATED event is not implemented\n");
+ _debug(" !EXOSIP_REGISTRATION_TERMINATED event is not implemented\n");
break;
/* INVITE related events within calls */
@@ -236,10 +236,10 @@ SIPVoIPLink::getEvent()
break;
case EXOSIP_CALL_NOANSWER: /** 07 < announce no answer within the timeout */
- _debug("EXOSIP_CALL_NOANSWER event is not implemented\n");
+ _debug(" !EXOSIP_CALL_NOANSWER event is not implemented\n");
break;
case EXOSIP_CALL_PROCEEDING: /** 08 < announce processing by a remote app */
- _debug("EXOSIP_CALL_NOANSWER event is not implemented\n");
+ _debug(" !EXOSIP_CALL_PROCEEDING event is not implemented\n");
break;
case EXOSIP_CALL_RINGING: /** 09 < announce ringback */
SIPCallRinging(event);
@@ -248,7 +248,7 @@ SIPVoIPLink::getEvent()
SIPCallAnswered(event);
break;
case EXOSIP_CALL_REDIRECTED: /** 11 < announce a redirection */
- _debug("EXOSIP_CALL_REDIRECTED event is not implemented\n");
+ _debug(" !EXOSIP_CALL_REDIRECTED event is not implemented\n");
break;
case EXOSIP_CALL_REQUESTFAILURE: /** 12 < announce a request failure */
SIPCallRequestFailure(event);
@@ -265,7 +265,7 @@ SIPVoIPLink::getEvent()
case EXOSIP_CALL_CANCELLED: /** 16 < announce that call has been cancelled */
case EXOSIP_CALL_TIMEOUT: /** 17 < announce that call has failed */
- Manager::instance().displayError("EXOSIP Call Error not implemented yet");
+ Manager::instance().displayError(" !EXOSIP Call Error not implemented yet");
break;
/* request related events within calls (except INVITE) */
@@ -279,7 +279,7 @@ SIPVoIPLink::getEvent()
case EXOSIP_CALL_MESSAGE_REQUESTFAILURE: /** 22 < announce a failure. */
case EXOSIP_CALL_MESSAGE_SERVERFAILURE: /** 23 < announce a failure. */
case EXOSIP_CALL_MESSAGE_GLOBALFAILURE: /** 24 < announce a failure. */
- Manager::instance().displayError("EXOSIP Call Message not implemented yet");
+ Manager::instance().displayError(" !EXOSIP Call Message not implemented yet");
break;
case EXOSIP_CALL_CLOSED: /** 25 < a BYE was received for this call */
@@ -299,30 +299,30 @@ SIPVoIPLink::getEvent()
case EXOSIP_MESSAGE_PROCEEDING: /** 28 < announce a 1xx for MESSAGE. */
case EXOSIP_MESSAGE_ANSWERED: /** 29 < announce a 200ok */
case EXOSIP_MESSAGE_REDIRECTED: /** 30 < announce a failure. */
- Manager::instance().displayError("EXOSIP Message not implemented yet");
+ Manager::instance().displayError(" !EXOSIP Message not implemented yet");
break;
case EXOSIP_MESSAGE_REQUESTFAILURE: /** 31 < announce a failure. */
if (event->response !=0 && event->response->status_code == SIP_METHOD_NOT_ALLOWED) {
Manager::instance().incomingMessage(getAccountID(), "Message are not allowed");
} else {
- Manager::instance().displayError("EXOSIP_MESSAGE_REQUESTFAILURE not implemented yet");
+ Manager::instance().displayError(" !EXOSIP_MESSAGE_REQUESTFAILURE not implemented yet");
}
break;
case EXOSIP_MESSAGE_SERVERFAILURE: /** 32 < announce a failure. */
case EXOSIP_MESSAGE_GLOBALFAILURE: /** 33 < announce a failure. */
- Manager::instance().displayError("EXOSIP Message not implemented yet");
+ Manager::instance().displayError(" !EXOSIP Message not implemented yet");
break;
/* Presence and Instant Messaging */
case EXOSIP_SUBSCRIPTION_UPDATE: /** 34 < announce incoming SUBSCRIBE. */
case EXOSIP_SUBSCRIPTION_CLOSED: /** 35 < announce end of subscription. */
- Manager::instance().displayError("EXOSIP Subscription not implemented yet");
+ Manager::instance().displayError(" !EXOSIP Subscription not implemented yet");
break;
case EXOSIP_SUBSCRIPTION_NOANSWER: /** 37 < announce no answer */
case EXOSIP_SUBSCRIPTION_PROCEEDING: /** 38 < announce a 1xx */
- Manager::instance().displayError("EXOSIP Subscription resposne not implemented yet");
+ Manager::instance().displayError(" !EXOSIP Subscription resposne not implemented yet");
break;
case EXOSIP_SUBSCRIPTION_ANSWERED: /** 39 < announce a 200ok */
eXosip_lock();
@@ -336,12 +336,12 @@ SIPVoIPLink::getEvent()
case EXOSIP_SUBSCRIPTION_GLOBALFAILURE: /** 43 < announce a global failure */
case EXOSIP_SUBSCRIPTION_NOTIFY: /** 44 < announce new NOTIFY request */
case EXOSIP_SUBSCRIPTION_RELEASED: /** 45 < call context is cleared. */
- Manager::instance().displayError("EXOSIP Subscription resposne not implemented yet");
+ Manager::instance().displayError(" !EXOSIP Subscription resposne not implemented yet");
break;
case EXOSIP_IN_SUBSCRIPTION_NEW: /** 46 < announce new incoming SUBSCRIBE.*/
case EXOSIP_IN_SUBSCRIPTION_RELEASED: /** 47 < announce end of subscription. */
- Manager::instance().displayError("EXOSIP Subscription not implemented yet");
+ Manager::instance().displayError(" !EXOSIP Subscription not implemented yet");
break;
case EXOSIP_EVENT_COUNT: /** 48 < MAX number of events */
@@ -354,7 +354,7 @@ bool
SIPVoIPLink::setRegister()
{
if (_eXosipRegID != EXOSIP_ERROR_STD) {
- Manager::instance().displayError("SIP Error: Registration already sent. Try to unregister");
+ Manager::instance().displayError("! SIP Error: Registration already sent. Try to unregister");
return false;
}
@@ -377,11 +377,11 @@ SIPVoIPLink::setRegister()
osip_message_t *reg = NULL;
eXosip_lock();
if (!_proxy.empty()) {
- _debug("SIP Register: From: %s to %s\n", from.data(), proxy.data());
+ _debug("* SIP Info: Register from: %s to %s\n", from.data(), proxy.data());
_eXosipRegID = eXosip_register_build_initial_register(from.data(),
proxy.data(), NULL, EXPIRES_VALUE, ®);
} else {
- _debug("SIP Register: From: %s to %s\n", from.data(), hostname.data());
+ _debug("* SIP Info: Register from: %s to %s\n", from.data(), hostname.data());
_eXosipRegID = eXosip_register_build_initial_register(from.data(),
hostname.data(), NULL, EXPIRES_VALUE, ®);
}
@@ -391,7 +391,7 @@ SIPVoIPLink::setRegister()
}
if (!sendSIPAuthentification()) {
- _debug("SIP Register: No authentication\n");
+ _debug("* SIP Info: register without authentication\n");
return false;
}
@@ -401,12 +401,12 @@ SIPVoIPLink::setRegister()
eXosip_lock();
int eXosipErr = eXosip_register_send_register(_eXosipRegID, reg);
if (eXosipErr == EXOSIP_ERROR_BUILDING) {
- _debug("SIP Failure: Cannot build registration, check the setup\n");
+ _debug("! SIP Failure: Cannot build registration, check the setup\n");
eXosip_unlock();
return false;
}
if (eXosipErr == EXOSIP_ERROR_STD) {
- _debug("SIP Failure: Registration sending failed\n");
+ _debug("! SIP Failure: Registration sending failed\n");
eXosip_unlock();
return false;
}
@@ -455,7 +455,7 @@ SIPVoIPLink::setUnregister()
eXosip_unlock();
if (eXosipErr != EXOSIP_ERROR_NO) {
- _debug("SIP Failure: Unable to build registration for setUnregister");
+ _debug("! SIP Failure: Unable to build registration for setUnregister");
return false;
}
@@ -463,12 +463,12 @@ SIPVoIPLink::setUnregister()
_debug("< Sending REGISTER (expire=0)\n");
eXosipErr = eXosip_register_send_register (_eXosipRegID, reg);
if (eXosipErr == EXOSIP_ERROR_BUILDING) {
- _debug("SIP Failure: Cannot build registration (unregister), check the setup\n");
+ _debug("! SIP Failure: Cannot build registration (unregister), check the setup\n");
eXosip_unlock();
return false;
}
if (eXosipErr == EXOSIP_ERROR_STD) {
- _debug("SIP Failure: Unable to send registration (unregister)\n");
+ _debug("! SIP Failure: Unable to send registration (unregister)\n");
}
eXosip_unlock();
_eXosipRegID = EXOSIP_ERROR_STD;
@@ -498,11 +498,11 @@ SIPVoIPLink::newOutgoingCall(const CallID& id, const std::string& toUrl)
bool
SIPVoIPLink::answer(const CallID& id)
{
- _debug("SIP Action: start answering\n");
+ _debug("- SIP Action: start answering\n");
SIPCall* call = getSIPCall(id);
if (call==0) {
- _debug("SIP Failure: SIPCall doesn't exists\n");
+ _debug("! SIP Failure: SIPCall doesn't exists\n");
return false;
}
@@ -516,10 +516,9 @@ SIPVoIPLink::answer(const CallID& id)
// Send 200 OK
osip_message_t *answerMessage = NULL;
eXosip_lock();
- _debug("< Building 200 OK\n");
int i = eXosip_call_build_answer(call->getTid(), SIP_OK, &answerMessage);
if (i != 0) {
- _debug("< Send 400 Bad Request\n");
+ _debug("< SIP Building Error: send 400 Bad Request\n");
eXosip_call_send_answer (call->getTid(), SIP_BAD_REQUEST, NULL);
} else {
// use exosip, bug locked
@@ -533,10 +532,10 @@ SIPVoIPLink::answer(const CallID& id)
sdp_message_free(remote_sdp);
}
if (i != 0) {
- _debug("< Send 415 Unsupported Media Type\n");
+ _debug("< SIP Error: send 415 Unsupported Media Type\n");
eXosip_call_send_answer (call->getTid(), SIP_UNSUPPORTED_MEDIA_TYPE, NULL);
} else {
- _debug("< Send 200 OK\n");
+ _debug("< SIP send 200 OK\n");
eXosip_call_send_answer (call->getTid(), SIP_OK, answerMessage);
}
}
@@ -544,14 +543,14 @@ SIPVoIPLink::answer(const CallID& id)
if(i==0) {
// Incoming call is answered, start the sound channel.
- _debug("SIP: Starting AudioRTP when answering\n");
+ _debug("* SIP Info: Starting AudioRTP when answering\n");
if (_audiortp.createNewSession(call) >= 0) {
call->setAudioStart(true);
call->setConnectionState(Call::Connected);
call->setState(Call::Active);
return true;
} else {
- _debug("FATAL: Unable to start sound when answering %s/%d\n", __FILE__, __LINE__);
+ _debug("! SIP Failure: Unable to start sound when answering %s/%d\n", __FILE__, __LINE__);
}
}
removeCall(call->getCallId());
@@ -574,9 +573,9 @@ bool
SIPVoIPLink::hangup(const CallID& id)
{
SIPCall* call = getSIPCall(id);
- if (call==0) { _debug("Call doesn't exist\n"); return false; }
+ if (call==0) { _debug("! SIP Error: Call doesn't exist\n"); return false; }
- _debug("Hang up call %s [cd: %3d %3d]\n", id.data(), call->getCid(), call->getDid());
+ _debug("- SIP Action: Hang up call %s [cd: %3d %3d]\n", id.data(), call->getCid(), call->getDid());
// Release SIP stack.
eXosip_lock();
eXosip_call_terminate(call->getCid(), call->getDid());
@@ -584,7 +583,7 @@ SIPVoIPLink::hangup(const CallID& id)
// Release RTP channels
if (Manager::instance().isCurrentCall(id)) {
- _debug("SIP: Stopping AudioRTP for hangup\n");
+ _debug("* SIP Info: Stopping AudioRTP for hangup\n");
_audiortp.closeRtpSession();
}
removeCall(id);
@@ -595,9 +594,9 @@ bool
SIPVoIPLink::cancel(const CallID& id)
{
SIPCall* call = getSIPCall(id);
- if (call==0) { _debug("Call doesn't exist\n"); return false; }
+ if (call==0) { _debug("! SIP Error: Call doesn't exist\n"); return false; }
- _debug("Cancel call %s [cid: %3d]\n", id.data(), call->getCid());
+ _debug("- SIP Action: Cancel call %s [cid: %3d]\n", id.data(), call->getCid());
// Release SIP stack.
eXosip_lock();
eXosip_call_terminate(call->getCid(), -1);
@@ -611,12 +610,12 @@ bool
SIPVoIPLink::onhold(const CallID& id)
{
SIPCall* call = getSIPCall(id);
- if (call==0) { _debug("Call doesn't exist\n"); return false; }
+ if (call==0) { _debug("! SIP Error: call doesn't exist\n"); return false; }
// Stop sound
call->setAudioStart(false);
call->setState(Call::Hold);
- _debug("SIP: Stopping AudioRTP when onhold\n");
+ _debug("* SIP Info: Stopping AudioRTP for onhold action\n");
_audiortp.closeRtpSession();
@@ -627,7 +626,7 @@ SIPVoIPLink::onhold(const CallID& id)
eXosip_unlock ();
if (local_sdp == NULL) {
- _debug("SIP Failure: unable to find local_sdp\n");
+ _debug("! SIP Failure: unable to find local_sdp\n");
return false;
}
@@ -639,7 +638,7 @@ SIPVoIPLink::onhold(const CallID& id)
if (exosipErr != 0) {
sdp_message_free(local_sdp);
- _debug("SIP Failure: unable to build invite method to hold call\n");
+ _debug("! SIP Failure: unable to build invite method to hold call\n");
return false;
}
@@ -651,7 +650,7 @@ SIPVoIPLink::onhold(const CallID& id)
if (i != 0) {
sdp_message_free (local_sdp);
osip_message_free (invite);
- _debug("SIP Failure: Unable to hold call in SDP\n");
+ _debug("! SIP Failure: Unable to hold call in SDP\n");
return false;
}
@@ -660,7 +659,7 @@ SIPVoIPLink::onhold(const CallID& id)
if (i != 0) {
osip_message_free (invite);
osip_free (tmp);
- _debug("SIP Failure: Unable to translate sdp message to string\n");
+ _debug("! SIP Failure: Unable to translate sdp message to string\n");
return false;
}
osip_message_set_body (invite, tmp, strlen (tmp));
@@ -669,7 +668,7 @@ SIPVoIPLink::onhold(const CallID& id)
}
// send request
- _debug("< Send on hold request\n");
+ _debug("< SIP: Send on hold request\n");
eXosip_lock ();
exosipErr = eXosip_call_send_request (did, invite);
eXosip_unlock ();
@@ -681,7 +680,7 @@ bool
SIPVoIPLink::offhold(const CallID& id)
{
SIPCall* call = getSIPCall(id);
- if (call==0) { _debug("Call doesn't exist\n"); return false; }
+ if (call==0) { _debug("! SIP Error: Call doesn't exist\n"); return false; }
int did = call->getDid();
@@ -690,7 +689,7 @@ SIPVoIPLink::offhold(const CallID& id)
eXosip_unlock ();
if (local_sdp == NULL) {
- _debug("SIP Failure: unable to find local_sdp\n");
+ _debug("! SIP Failure: unable to find local_sdp\n");
return false;
}
@@ -735,11 +734,11 @@ SIPVoIPLink::offhold(const CallID& id)
eXosip_unlock ();
// Enable audio
- _debug("SIP: Starting AudioRTP when offhold\n");
+ _debug("* SIP Info: Starting AudioRTP when offhold\n");
call->setState(Call::Active);
// it's sure that this is the current call id...
if (_audiortp.createNewSession(call) < 0) {
- _debug("SIP Failure: Unable to start sound (%s:%d)\n", __FILE__, __LINE__);
+ _debug("! SIP Failure: Unable to start sound (%s:%d)\n", __FILE__, __LINE__);
return false;
}
return true;
@@ -749,7 +748,7 @@ bool
SIPVoIPLink::transfer(const CallID& id, const std::string& to)
{
SIPCall* call = getSIPCall(id);
- if (call==0) { _debug("Call doesn't exist\n"); return false; }
+ if (call==0) { _debug("! SIP Failure: Call doesn't exist\n"); return false; }
std::string tmp_to = SIPToHeader(to);
if (tmp_to.find("@") == std::string::npos) {
@@ -762,7 +761,7 @@ SIPVoIPLink::transfer(const CallID& id, const std::string& to)
int exosipErr = eXosip_call_build_refer(call->getDid(), (char*)tmp_to.data(), &refer);
if (exosipErr == 0) {
// Send transfer request
- _debug("SIPCall: < send transfer request to %s\n", tmp_to.data());
+ _debug("< SIP send transfer request to %s\n", tmp_to.data());
exosipErr = eXosip_call_send_request(call->getDid(), refer);
}
eXosip_unlock();
@@ -881,7 +880,7 @@ SIPVoIPLink::SIPOutgoingInvite(SIPCall* call)
{
// If no SIP proxy setting for direct call with only IP address
if (!SIPStartCall(call, "")) {
- _debug("SIP Failure: call not started\n");
+ _debug("! SIP Failure: call not started\n");
return false;
}
return true;
@@ -899,7 +898,7 @@ SIPVoIPLink::SIPStartCall(SIPCall* call, const std::string& subject)
_debug(" Route: %s\n", route.data());
if (!SIPCheckUrl(from)) {
- _debug("SIP Error: Source address is invalid %s\n", from.data());
+ _debug("! SIP Error: Source address is invalid %s\n", from.data());
Manager::instance().displayConfigError("Error in source address");
return false;
}
@@ -937,7 +936,7 @@ SIPVoIPLink::SIPStartCall(SIPCall* call, const std::string& subject)
// add each payload in the list of payload
media_audio << payload << " ";
- rtpmap_attr << "a=rtpmap: " << payload << " " <<
+ rtpmap_attr << "a=rtpmap:" << payload << " " <<
iter->second->getCodecName().data() << "/" << iter->second->getClockRate();
nbChannel = iter->second->getChannel();
@@ -982,7 +981,7 @@ SIPVoIPLink::SIPStartCall(SIPCall* call, const std::string& subject)
eXosip_unlock();
return false ;
} else {
- _debug("SIP Info: Outgoing callID is %s, cid=%d\n", call->getCallId().data(), cid);
+ _debug("* SIP Info: Outgoing callID is %s, cid=%d\n", call->getCallId().data(), cid);
eXosip_call_set_reference (cid, NULL);
}
eXosip_unlock();
@@ -1053,12 +1052,12 @@ SIPVoIPLink::SIPCheckUrl(const std::string& url)
osip_from_t *to;
i = osip_from_init(&to);
if (i != 0) {
- _debug("SIP Warning: Cannot initialize osip parser\n");
+ _debug("! SIP Warning: Cannot initialize osip parser\n");
return false;
}
i = osip_from_parse(to, url.data());
if (i != 0) {
- _debug("SIP Warning: Cannot parse url %s\n", url.data());
+ _debug("! SIP Warning: Cannot parse url %s\n", url.data());
return false;
}
@@ -1099,10 +1098,11 @@ SIPVoIPLink::SIPCallInvite(eXosip_event_t *event)
SIPCall* call = new SIPCall(id, Call::Incoming);
if (!call) {
- _debug("SIP Failure: unable to create an incoming call");
+ _debug("! SIP Failure: unable to create an incoming call");
return;
}
setCallAudioLocal(call);
+ call->setCodecMap(Manager::instance().getCodecDescriptorMap());
call->setConnectionState(Call::Progressing);
if (call->SIPCallInvite(event)) {
if (Manager::instance().incomingCall(call, getAccountID())) {
@@ -1121,7 +1121,7 @@ SIPVoIPLink::SIPCallReinvite(eXosip_event_t *event)
_debug("> REINVITE (receive)\n");
SIPCall* call = findSIPCallWithCidDid(event->cid, event->did);
if (call == 0) {
- _debug("SIP Failure: unknown call\n");
+ _debug("! SIP Failure: unknown call\n");
_debug("< Send 488 Not Acceptable Here");
eXosip_lock();
eXosip_call_send_answer(event->tid, 488, NULL);
@@ -1132,7 +1132,7 @@ SIPVoIPLink::SIPCallReinvite(eXosip_event_t *event)
// STOP tone
Manager::instance().stopTone();
// STOP old rtp session
- _debug("SIP: Stopping AudioRTP when reinvite\n");
+ _debug("* SIP Info: Stopping AudioRTP when reinvite\n");
_audiortp.closeRtpSession();
call->setAudioStart(false);
}
@@ -1144,7 +1144,7 @@ SIPVoIPLink::SIPCallRinging(eXosip_event_t *event)
{
SIPCall* call = findSIPCallWithCid(event->cid);
if (!call) {
- _debug("SIP Failure: unknown call\n");
+ _debug("! SIP Failure: unknown call\n");
return;
}
// we could set the cid/did/tid and get the FROM here...
@@ -1160,7 +1160,7 @@ SIPVoIPLink::SIPCallAnswered(eXosip_event_t *event)
{
SIPCall* call = findSIPCallWithCid(event->cid);
if (!call) {
- _debug("SIP Failure: unknown call\n");
+ _debug("! SIP Failure: unknown call\n");
return;
}
call->setDid(event->did);
@@ -1174,7 +1174,7 @@ SIPVoIPLink::SIPCallAnswered(eXosip_event_t *event)
Manager::instance().peerAnsweredCall(call->getCallId());
if (Manager::instance().isCurrentCall(call->getCallId())) {
- _debug("SIP: Starting AudioRTP when answering\n");
+ _debug("* SIP Info: Starting AudioRTP when answering\n");
if ( _audiortp.createNewSession(call) < 0) {
_debug("RTP Failure: unable to create new session\n");
} else {
@@ -1182,7 +1182,7 @@ SIPVoIPLink::SIPCallAnswered(eXosip_event_t *event)
}
}
} else {
- _debug("Answering call (on/off hold to send ACK)\n");
+ _debug("* SIP Info: Answering call (on/off hold to send ACK)\n");
call->SIPCallAnswered(event);
}
}
@@ -1192,18 +1192,18 @@ SIPVoIPLink::SIPCallRequestFailure(eXosip_event_t *event)
{
if (!event->response) { return; }
// 404 error
- _debug("SIP Event: Request Failure, receive code %d\n", event->response->status_code);
+ _debug(" Request Failure, receive code %d\n", event->response->status_code);
// Handle 4XX errors
switch (event->response->status_code) {
case SIP_PROXY_AUTHENTICATION_REQUIRED:
- _debug("SIP Server ask required authentification: logging...\n");
+ _debug("- SIP Action: Server ask required authentification: logging...\n");
sendSIPAuthentification();
eXosip_lock();
eXosip_automatic_action();
eXosip_unlock();
break;
case SIP_UNAUTHORIZED:
- _debug("Request is unauthorized. SIP Server ask authentification: logging...\n");
+ _debug("- SIP Action: Request is unauthorized. SIP Server ask authentification: logging...\n");
sendSIPAuthentification();
break;
@@ -1273,7 +1273,7 @@ SIPVoIPLink::SIPCallAck(eXosip_event_t *event)
if (!call) { return; }
if (!call->isAudioStarted()) {
if (Manager::instance().isCurrentCall(call->getCallId())) {
- _debug("SIP: Starting AudioRTP when ack\n");
+ _debug("* SIP Info: Starting AudioRTP when ack\n");
if ( _audiortp.createNewSession(call) ) {
call->setAudioStart(true);
}
@@ -1286,13 +1286,13 @@ SIPVoIPLink::SIPCallMessageNew(eXosip_event_t *event)
{
if (0 == event->request) return;
- _debug("SIP: Receive a call message\n");
+ _debug(" > SIP Event: Receive a call message\n");
if (MSG_IS_INFO(event->request)) {
- _debug("SIP: It's a Request Info\n");
+ _debug("* SIP Info: It's a Request Info\n");
osip_content_type_t* c_t = event->request->content_type;
if (c_t != 0 && c_t->type != 0 && c_t->subtype != 0 ) {
- _debug("SIP: Content Type of the message: %s/%s\n", c_t->type, c_t->subtype);
+ _debug("* SIP Info: Content Type of the message: %s/%s\n", c_t->type, c_t->subtype);
// application/dtmf-relay
if (strcmp(c_t->type, "application") == 0 && strcmp(c_t->subtype, "dtmf-relay") == 0) {
handleDtmfRelay(event);
@@ -1303,10 +1303,10 @@ SIPVoIPLink::SIPCallMessageNew(eXosip_event_t *event)
osip_message_t *answerOKNewMessage;
eXosip_lock();
if ( 0 == eXosip_call_build_answer(event->tid, SIP_OK, &answerOKNewMessage)) {
- _debug("< Sending 200 OK\n");
+ _debug("< SIP Sending 200 OK\n");
eXosip_call_send_answer(event->tid, SIP_OK, answerOKNewMessage);
} else {
- _debug("SIP Failure: Could not sent an OK message\n");
+ _debug("! SIP Failure: Could not sent an OK message\n");
}
eXosip_unlock();
@@ -1323,7 +1323,7 @@ SIPVoIPLink::SIPCallClosed(eXosip_event_t *event)
call->setDid(event->did);
if (Manager::instance().isCurrentCall(id)) {
call->setAudioStart(false);
- _debug("SIP: Stopping AudioRTP when closing\n");
+ _debug("* SIP Info: Stopping AudioRTP when closing\n");
_audiortp.closeRtpSession();
}
Manager::instance().peerHungupCall(id);
@@ -1357,7 +1357,7 @@ SIPVoIPLink::SIPMessageNew(eXosip_event_t *event)
}
// Voice message
else if (MSG_IS_NOTIFY(event->request)){
- _debug("> NOTIFY Voice message\n");
+ _debug(" > NOTIFY Voice message\n");
int ii;
unsigned int pos;
unsigned int pos_slash;
@@ -1366,7 +1366,7 @@ SIPVoIPLink::SIPMessageNew(eXosip_event_t *event)
// Get the message body
ii = osip_message_get_body(event->request, 0, &body);
if (ii != 0) {
- _debug(" Cannot get body in a new EXOSIP_MESSAGE_NEW event\n");
+ _debug("! SIP Error: Cannot get body in a new EXOSIP_MESSAGE_NEW event\n");
return;
}
@@ -1403,17 +1403,17 @@ SIPVoIPLink::SIPMessageNew(eXosip_event_t *event)
// osip_content_type_t* osip_message::content_type
osip_content_type_t* c_t = event->request->content_type;
if (c_t != 0 && c_t->type != 0 && c_t->subtype != 0 ) {
- _debug(" Content Type of the message: %s/%s\n", c_t->type, c_t->subtype);
+ _debug("* SIP Info: Content Type of the message: %s/%s\n", c_t->type, c_t->subtype);
osip_body_t *body = NULL;
// Get the message body
if (0 == osip_message_get_body(event->request, 0, &body)) {
- _debug(" Body length: %d\n", body->length);
+ _debug("* SIP Info: Body length: %d\n", body->length);
if (body->body!=0 &&
strcmp(c_t->type,"text") == 0 &&
strcmp(c_t->subtype,"plain") == 0
) {
- _debug(" Text body: %s\n", body->body);
+ _debug("* SIP Info: Text body: %s\n", body->body);
Manager::instance().incomingMessage(getAccountID(), body->body);
}
}
@@ -1433,7 +1433,7 @@ SIPCall*
SIPVoIPLink::findSIPCallWithCid(int cid)
{
if (cid < 1) {
- _debug("Not enough information for this event\n");
+ _debug("! SIP Error: Not enough information for this event\n");
return 0;
}
ost::MutexLock m(_callMapMutex);
@@ -1453,7 +1453,7 @@ SIPCall*
SIPVoIPLink::findSIPCallWithCidDid(int cid, int did)
{
if (cid < 1 && did < -1) {
- _debug("Not enough information for this event\n");
+ _debug("! SIP Error: Not enough information for this event\n");
return 0;
}
ost::MutexLock m(_callMapMutex);
@@ -1494,7 +1494,7 @@ SIPVoIPLink::handleDtmfRelay(eXosip_event_t* event) {
osip_body_t *body = NULL;
// Get the message body
if (0 == osip_message_get_body(event->request, 0, &body) && body->body != 0 ) {
- _debug(" Text body: %s\n", body->body);
+ _debug("* SIP Info: Text body: %s\n", body->body);
std::string dtmfBody(body->body);
unsigned int posStart = 0;
unsigned int posEnd = 0;
@@ -1509,7 +1509,7 @@ SIPVoIPLink::handleDtmfRelay(eXosip_event_t* event) {
posEnd = dtmfBody.length();
}
signal = dtmfBody.substr(posStart, posEnd-posStart+1);
- _debug("Signal value: %s\n", signal.c_str());
+ _debug("* SIP Info: Signal value: %s\n", signal.c_str());
if (!signal.empty()) {
if (Manager::instance().isCurrentCall(call->getCallId())) {
--
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