mirror of
https://git.jami.net/savoirfairelinux/jami-daemon.git
synced 2025-08-12 22:09:25 +08:00
[#1910] Remove warnings compilation in src
This commit is contained in:
@ -23,7 +23,10 @@
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#include "manager.h"
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Account::Account (const AccountID& accountID, std::string type) :
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_accountID (accountID), _link (NULL), _enabled (false), _type (type)
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_accountID (accountID)
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, _link (NULL)
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, _enabled (false)
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, _type (type)
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{
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setRegistrationState (Unregistered);
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}
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@ -178,12 +178,6 @@ class Account{
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*/
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std::string _alias;
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/*
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* The account type
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* IAX2 or SIP
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*/
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std::string _type;
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/**
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* Voice over IP Link contains a listener thread and calls
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*/
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@ -196,6 +190,12 @@ class Account{
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*/
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bool _enabled;
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/*
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* The account type
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* IAX2 or SIP
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*/
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std::string _type;
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/*
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* The registration state of the account
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*/
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@ -88,6 +88,8 @@ AudioStream::drainStream(void) {
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pa_threaded_mainloop_unlock(_mainloop);
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}
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return true;
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}
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bool
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@ -32,9 +32,7 @@ int UrlHook::addAction (std::string field_value, std::string command)
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/* Execute the command in the background to not block the application */
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command_bg = command + " " + field_value + "&" ;
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/* Execute a system call */
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RUN_COMMAND (command_bg.c_str());
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return 0;
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return RUN_COMMAND (command_bg.c_str());
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}
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@ -920,7 +920,6 @@ void
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IAXVoIPLink::iaxHandleRegReply (iax_event* event)
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{
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int new_voicemails;
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std::string account_id;
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IAXAccount *account;
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@ -86,24 +86,29 @@ main (int argc, char **argv)
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}
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} else {
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// PID file exists. Check the former process still alive or not. If alive, give user a hint.
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fgets (cOldPid, 64, fp);
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fclose (fp);
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char *res;
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res = fgets (cOldPid, 64, fp);
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if (res == NULL) perror ("Error getting string from stream");
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else
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{
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fclose (fp);
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if (kill (atoi (cOldPid), 0) == SUCCESS) {
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fprintf (stderr, "There is already a sflphoned daemon running in the system. Starting Failed.\n");
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exit (-1);
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} else {
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if ( (fp = fopen (homepid,"w")) == NULL) {
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fprintf (stderr, "Writing to PID file %s failed. Exited.\n", homepid);
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exit (-1);
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} else {
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fputs (cPid , fp);
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fclose (fp);
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}
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}
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}
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if (kill (atoi (cOldPid), 0) == SUCCESS) {
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fprintf (stderr, "There is already a sflphoned daemon running in the system. Starting Failed.\n");
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exit (-1);
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} else {
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if ( (fp = fopen (homepid,"w")) == NULL) {
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fprintf (stderr, "Writing to PID file %s failed. Exited.\n", homepid);
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exit (-1);
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} else {
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fputs (cPid , fp);
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fclose (fp);
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}
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}
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}
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}
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int sessionPort = 0;
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if (argc == 2) {
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@ -286,7 +286,6 @@ ManagerImpl::answerCall (const CallID& id)
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AccountID currentaccountid = getAccountFromCall (id);
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Call* currentcall = getAccountLink (currentaccountid)->getCall (getCurrentCallId());
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_debug ("ManagerImpl::answerCall :: current call->getState %i \n",currentcall->getState());
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if (currentcall->getState() == 1)
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isActive = true;
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@ -1050,19 +1049,12 @@ bool ManagerImpl::playATone (Tone::TONEID toneId)
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void ManagerImpl::stopTone (bool stopAudio=true)
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{
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int hasToPlayTone;
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AudioLayer *audiolayer;
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hasToPlayTone = getConfigInt (SIGNALISATION, PLAY_TONES);
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if (!hasToPlayTone)
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return;
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// if (stopAudio) {
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// audiolayer = getAudioDriver();
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// if (audiolayer) audiolayer->stopStream();
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// }
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_toneMutex.enterMutex();
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if (_telephoneTone != 0) {
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@ -1628,10 +1620,14 @@ ManagerImpl::getAudioOutputDeviceList (void)
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{
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_debug ("Get audio output device list\n");
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AlsaLayer *layer;
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std::vector <std::string> devices;
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layer = dynamic_cast<AlsaLayer*> (getAudioDriver ());
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if (layer) return layer -> getSoundCardsInfo (SFL_PCM_PLAYBACK);
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if (layer)
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devices = layer -> getSoundCardsInfo (SFL_PCM_PLAYBACK);
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return devices;
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}
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/**
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@ -1664,12 +1660,15 @@ ManagerImpl::setAudioOutputDevice (const int index)
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std::vector<std::string>
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ManagerImpl::getAudioInputDeviceList (void)
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{
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_debug ("Get audio input device list\n");
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AlsaLayer *audiolayer;
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std::vector <std::string> devices;
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audiolayer = dynamic_cast<AlsaLayer *> (getAudioDriver());
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if (audiolayer) return audiolayer->getSoundCardsInfo (SFL_PCM_CAPTURE);
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if (audiolayer)
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devices = audiolayer->getSoundCardsInfo (SFL_PCM_CAPTURE);
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return devices;
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}
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/**
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@ -1932,7 +1931,7 @@ ManagerImpl::setPulseAppVolumeControl (void)
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void ManagerImpl::setAudioManager (const int32_t& api)
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{
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int type, samplerate, framesize, numCardIn, numCardOut;
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int type;
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std::string alsaPlugin;
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_debug ("Setting audio manager \n");
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@ -1984,7 +1983,10 @@ ManagerImpl::getAudioDeviceIndex (const std::string name)
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alsalayer = dynamic_cast<AlsaLayer *> (getAudioDriver());
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if (alsalayer) return alsalayer -> soundCardGetIndex (name);
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if (alsalayer)
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return alsalayer -> soundCardGetIndex (name);
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else
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return 0;
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}
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std::string
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@ -2390,7 +2392,7 @@ ManagerImpl::getAccountList()
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{
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std::vector< std::string > v;
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std::vector< std::string > account_order;
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int i;
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unsigned int i;
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account_order = loadAccountOrder ();
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AccountMap::iterator iter;
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@ -2476,7 +2478,6 @@ void ManagerImpl::setAccountDetails (const std::string& accountID, const std::ma
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std::string accountType;
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Account *acc;
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VoIPLink *link;
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accountType = (*details.find (CONFIG_ACCOUNT_TYPE)).second;
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@ -2681,7 +2682,7 @@ ManagerImpl::loadAccountMap()
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short nbAccount = 0;
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TokenList sections = _config.getSections();
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std::string accountType;
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Account* tmpAccount;
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Account *tmpAccount = 0;
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std::vector <std::string> account_order;
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TokenList::iterator iter = sections.begin();
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@ -2811,25 +2812,6 @@ ManagerImpl::getAccountIdFromNameAndServer (const std::string& userName, const s
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return AccountNULL;
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}
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AccountMap ManagerImpl::getSipAccountMap (void)
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{
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AccountMap::iterator iter;
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AccountMap sipaccounts;
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AccountID id;
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Account *account;
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for (iter = _accountMap.begin(); iter != _accountMap.end(); ++iter) {
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if (iter->second->getType() == "sip") {
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//id = iter->first;
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//account = iter->second;
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//sipaccounts.insert( std::pair<id, account> );
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}
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}
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return sipaccounts;
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}
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void ManagerImpl::restartPJSIP (void)
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{
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SIPVoIPLink *siplink;
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@ -3133,7 +3115,6 @@ std::vector< std::string >
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ManagerImpl::getCallList (void)
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{
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std::vector< std::string > v;
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int i;
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CallAccountMap::iterator iter = _callAccountMap.begin ();
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@ -893,11 +893,6 @@ class ManagerImpl {
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void registerCurSIPAccounts(VoIPLink *link);
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/**
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* Returns a map with only the existing SIP accounts
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*/
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AccountMap getSipAccountMap( void );
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/*
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* Initialize audiodriver
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*/
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@ -207,6 +207,9 @@ bool AudioRecord::setRecording()
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recordingEnabled_ = true; // once opend file, start recording
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}
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// WARNING: Unused return value
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return true;
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}
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void AudioRecord::stopRecording()
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@ -344,6 +347,8 @@ bool AudioRecord::openExistingRawFile()
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_debug ("AudioRecord::openExistingRawFile() : could not create RAW file!\n");
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return false;
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}
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return true;
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}
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@ -386,6 +391,8 @@ bool AudioRecord::openExistingWavFile()
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if (fseek (fp, 4 , SEEK_END) != 0)
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_debug ("AudioRecors::OpenExistingWavFile : 2.Couldn't seek at the en of the file \n");
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return true;
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}
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@ -35,7 +35,14 @@ static const pj_str_t STR_RTPMAP = { (char*) "rtpmap", 6 };
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Sdp::Sdp (pj_pool_t *pool)
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: _local_media_cap(), _session_media (0), _ip_addr (""), _local_offer (NULL), _negociated_offer (NULL), _negociator (NULL), _pool (NULL), _local_extern_audio_port (0)
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: _local_media_cap()
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, _session_media (0)
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, _negociator (NULL)
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, _ip_addr ("")
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, _local_offer (NULL)
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, _negociated_offer (NULL)
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, _pool (NULL)
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, _local_extern_audio_port (0)
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{
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_pool = pool;
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}
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@ -206,8 +213,8 @@ int Sdp::receiving_initial_offer (pjmedia_sdp_session* remote)
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pj_status_t Sdp::check_sdp_answer(pjsip_inv_session *inv, pjsip_rx_data *rdata)
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{
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static const pj_str_t str_application = { "application", 11 };
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static const pj_str_t str_sdp = { "sdp", 3 };
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static const pj_str_t str_application = { (char*) "application", 11 };
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static const pj_str_t str_sdp = { (char*) "sdp", 3 };
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pj_status_t status;
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pjsip_msg * message = NULL;
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pjmedia_sdp_session * remote_sdp = NULL;
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@ -469,7 +476,7 @@ void Sdp::set_local_media_capabilities ()
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{
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CodecOrder selected_codecs;
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int i;
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unsigned int i;
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sdpMedia *audio;
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CodecsMap codecs_list;
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CodecsMap::iterator iter;
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@ -220,15 +220,15 @@ class Sdp {
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// negociation, with the compatible media from each part
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pjmedia_sdp_session *_negociated_offer;
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// The pool to allocate memory
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pj_pool_t *_pool;
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/** Local audio port */
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int _local_extern_audio_port;
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/** Remote's audio port */
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unsigned int _remote_audio_port;
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// The pool to allocate memory
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pj_pool_t *_pool;
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Sdp(const Sdp&); //No Copy Constructor
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Sdp& operator=(const Sdp&); //No Assignment Operator
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|
@ -25,10 +25,10 @@
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SIPAccount::SIPAccount (const AccountID& accountID)
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: Account (accountID, "sip")
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, _cred (NULL)
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, _regc()
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, _bRegister (false)
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, _contact ("")
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, _resolveOnce (false)
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, _regc()
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{
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/* SIPVoIPlink is used as a singleton, because we want to have only one link for all the SIP accounts created */
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/* So instead of creating a new instance, we just fetch the static instance, or create one if it is not yet */
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@ -97,8 +97,6 @@ class SIPCall : public Call
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private:
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Sdp *_local_sdp;
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int _cid;
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int _did;
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int _tid;
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@ -111,7 +109,11 @@ class SIPCall : public Call
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pjsip_evsub *_xferSub;
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pjsip_inv_session *_invSession;
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pjsip_inv_session *_invSession;
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Sdp *_local_sdp;
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};
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#endif
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@ -33,25 +33,25 @@
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#define CAN_REINVITE 1
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static char * invitationStateMap[] = {
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"PJSIP_INV_STATE_NULL",
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"PJSIP_INV_STATE_CALLING",
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"PJSIP_INV_STATE_INCOMING",
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"PJSIP_INV_STATE_EARLY",
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"PJSIP_INV_STATE_CONNECTING",
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"PJSIP_INV_STATE_CONFIRMED",
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"PJSIP_INV_STATE_DISCONNECTED"
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(char*) "PJSIP_INV_STATE_NULL",
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(char*) "PJSIP_INV_STATE_CALLING",
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(char*) "PJSIP_INV_STATE_INCOMING",
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(char*) "PJSIP_INV_STATE_EARLY",
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(char*) "PJSIP_INV_STATE_CONNECTING",
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(char*) "PJSIP_INV_STATE_CONFIRMED",
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(char*) "PJSIP_INV_STATE_DISCONNECTED"
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};
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static char * transactionStateMap[] = {
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"PJSIP_TSX_STATE_NULL" ,
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"PJSIP_TSX_STATE_CALLING",
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"PJSIP_TSX_STATE_TRYING",
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"PJSIP_TSX_STATE_PROCEEDING",
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"PJSIP_TSX_STATE_COMPLETED",
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"PJSIP_TSX_STATE_CONFIRMED",
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"PJSIP_TSX_STATE_TERMINATED",
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"PJSIP_TSX_STATE_DESTROYED",
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"PJSIP_TSX_STATE_MAX"
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(char*) "PJSIP_TSX_STATE_NULL" ,
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(char*) "PJSIP_TSX_STATE_CALLING",
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(char*) "PJSIP_TSX_STATE_TRYING",
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(char*) "PJSIP_TSX_STATE_PROCEEDING",
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(char*) "PJSIP_TSX_STATE_COMPLETED",
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(char*) "PJSIP_TSX_STATE_CONFIRMED",
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(char*) "PJSIP_TSX_STATE_TERMINATED",
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(char*) "PJSIP_TSX_STATE_DESTROYED",
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(char*) "PJSIP_TSX_STATE_MAX"
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};
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struct result
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@ -1882,6 +1882,7 @@ bool SIPVoIPLink::pjsip_shutdown (void)
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pj_shutdown();
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/* Done. */
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return true;
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}
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int getModId()
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@ -1902,7 +1903,7 @@ static void dns_cb(pj_status_t status, void *token, const struct pjsip_server_ad
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void set_voicemail_info (AccountID account, pjsip_msg_body *body)
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{
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int voicemail, pos_begin, pos_end;
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int voicemail = 0, pos_begin, pos_end;
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std::string voice_str = "Voice-Message: ";
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std::string delimiter = "/";
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std::string msg_body, voicemail_str;
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@ -2655,7 +2656,7 @@ void onCallTransfered (pjsip_inv_session *inv, pjsip_rx_data *rdata)
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return;
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}
|
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|
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SIPCall* newCall;
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SIPCall* newCall = 0;
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|
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SIPVoIPLink *link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink (accId));
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@ -2791,7 +2792,7 @@ void xfer_func_cb (pjsip_evsub *sub, pjsip_event *event)
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if (event->body.rx_msg.rdata->msg_info.msg_buf != NULL) {
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request = event->body.rx_msg.rdata->msg_info.msg_buf;
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if (request.find (noresource) != -1) {
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if ((int) request.find (noresource) != -1) {
|
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_debug ("UserAgent: NORESOURCE for transfer!\n");
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link->transferStep2();
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pjsip_evsub_terminate (sub, PJ_TRUE);
|
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@ -2800,7 +2801,7 @@ void xfer_func_cb (pjsip_evsub *sub, pjsip_event *event)
|
||||
return;
|
||||
}
|
||||
|
||||
if (request.find (ringing) != -1) {
|
||||
if ((int) request.find (ringing) != -1) {
|
||||
_debug ("UserAgent: transfered call RINGING!\n");
|
||||
link->transferStep2();
|
||||
pjsip_evsub_terminate (sub, PJ_TRUE);
|
||||
|
@ -351,9 +351,6 @@ class SIPVoIPLink : public VoIPLink
|
||||
*/
|
||||
bool pjsip_shutdown(void);
|
||||
|
||||
/** Do we use stun? */
|
||||
bool _useStun;
|
||||
|
||||
pj_status_t stunServerResolve();
|
||||
|
||||
/** Create SIP UDP Listener */
|
||||
@ -363,12 +360,6 @@ class SIPVoIPLink : public VoIPLink
|
||||
|
||||
std::string getLocalIP() {return _localExternAddress;}
|
||||
|
||||
/** Starting sound */
|
||||
AudioRtp* _audiortp;
|
||||
|
||||
/** For registration use only */
|
||||
int _regPort;
|
||||
|
||||
/* Flag to check if the STUN server is valid or not */
|
||||
bool validStunServer;
|
||||
|
||||
@ -381,6 +372,15 @@ class SIPVoIPLink : public VoIPLink
|
||||
/** Local Extern Port is the port seen by peers for SIP listener */
|
||||
unsigned int _localExternPort;
|
||||
|
||||
/** Starting sound */
|
||||
AudioRtp* _audiortp;
|
||||
|
||||
/** For registration use only */
|
||||
int _regPort;
|
||||
|
||||
/** Do we use stun? */
|
||||
bool _useStun;
|
||||
|
||||
/** Threading object */
|
||||
EventThread* _evThread;
|
||||
ost::Mutex _mutexSIP;
|
||||
|
@ -87,7 +87,6 @@ bool VoIPLink::clearCallMap()
|
||||
|
||||
Account* VoIPLink::getAccountPtr (void)
|
||||
{
|
||||
Account* account;
|
||||
AccountID id;
|
||||
|
||||
id = getAccountID();
|
||||
|
Reference in New Issue
Block a user