diff --git a/sflphone-common/src/audio/pulseaudio/pulselayer.cpp b/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
index 094ac14df50ea841b756d9e825a0d4a36cd1cc8b..bc83dcc6261d7fe3e9eb306a15f93c6c13483ba9 100644
--- a/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
+++ b/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
@@ -401,7 +401,7 @@ void PulseLayer::writeToSpeaker (void)
_mainBuffer.getData (out, toGet, 100);
pa_stream_write (playback->pulseStream(), out, toGet, NULL, 0, PA_SEEK_RELATIVE);
- _urgentRingBuffer.Discard (toGet);
+
} else {
bzero (out, framesPerBuffer * sizeof (SFLDataFormat));
diff --git a/sflphone-common/src/managerimpl.cpp b/sflphone-common/src/managerimpl.cpp
index d2fc50d23dd2c00616f027ca4e93f888ee37a9d1..e55b69b34e0b7f4008a1a1166dc8a52dc2fc4059 100644
--- a/sflphone-common/src/managerimpl.cpp
+++ b/sflphone-common/src/managerimpl.cpp
@@ -1492,7 +1492,7 @@ ManagerImpl::sendDtmf (const CallID& id, char code)
AccountID accountid = getAccountFromCall (id);
if (accountid == AccountNULL) {
- //_debug("Send DTMF: call doesn't exists\n");
+ _debug("Send DTMF: call doesn't exists\n");
playDtmf (code, false);
return false;
}
@@ -1535,13 +1535,19 @@ ManagerImpl::playDtmf (char code, bool isTalking)
bool hasToPlayTone = getConfigBool (SIGNALISATION, PLAY_DTMF);
if (!hasToPlayTone)
+ {
+ _debug(" playDtmf: Do not have to play a tone...\n");
return false;
+ }
// length in milliseconds
pulselen = getConfigInt (SIGNALISATION, PULSE_LENGTH);
if (!pulselen)
+ {
+ _debug(" playDtmf: Pulse length is not set...\n");
return false;
+ }
// numbers of int = length in milliseconds / 1000 (number of seconds)
// = number of seconds * SAMPLING_RATE by SECONDS
@@ -1551,13 +1557,16 @@ ManagerImpl::playDtmf (char code, bool isTalking)
// fast return, no sound, so no dtmf
if (audiolayer==0 || _dtmfKey == 0)
+ {
+ _debug(" playDtmf: Error no audio layer...\n");
return false;
+ }
// number of data sampling in one pulselen depends on samplerate
// size (n sampling) = time_ms * sampling/s
// ---------------------
// ms/s
- size = (int) (pulselen * ( (float) audiolayer->getSampleRate() /1000));
+ size = (int) ((pulselen * (float) audiolayer->getSampleRate()) / 1000);
// this buffer is for mono
// TODO <-- this should be global and hide if same size
@@ -1574,6 +1583,9 @@ ManagerImpl::playDtmf (char code, bool isTalking)
audiolayer->startStream();
audiolayer->putUrgent (buf, size * sizeof (SFLDataFormat));
}
+ else {
+ _debug(" playDtmf: Error cannot play dtmf");
+ }
ret = true;
diff --git a/sflphone-common/src/sipvoiplink.h b/sflphone-common/src/sipvoiplink.h
deleted file mode 100644
index 204e8639031e69e626d61e80d685aa3b57517c0e..0000000000000000000000000000000000000000
--- a/sflphone-common/src/sipvoiplink.h
+++ /dev/null
@@ -1,390 +0,0 @@
-/*
- * Copyright (C) 2004-2009 Savoir-Faire Linux inc.
- *
- * Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
- * Author: Yun Liu <yun.liu@savoirfairelinux.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- */
-
-#ifndef SIPVOIPLINK_H
-#define SIPVOIPLINK_H
-
-#include "voiplink.h"
-#include "hooks/urlhook.h"
-
-//////////////////////////////
-/* PJSIP imports */
-#include <pjsip.h>
-#include <pjlib.h>
-#include <pjsip_ua.h>
-#include <pjlib-util.h>
-#include <pjnath/stun_config.h>
-///////////////////////////////
-
-class EventThread;
-class SIPCall;
-
-#define RANDOM_LOCAL_PORT ((rand() % 27250) + 5250)*2
-#define RANDOM_SIP_PORT rand() % 64000 + 1024
-
-// To set the verbosity. From 0 (min) to 6 (max)
-#define PJ_LOG_LEVEL 0
-
-/**
- * @file sipvoiplink.h
- * @brief Specific VoIPLink for SIP (SIP core for incoming and outgoing events).
- * This class is based on the singleton design pattern.
- * One SIPVoIPLink can handle multiple SIP accounts, but all the SIP accounts have all the same SIPVoIPLink
- */
-
-class SIPVoIPLink : public VoIPLink
-{
- public:
-
- /**
- * Singleton method. Enable to retrieve the unique static instance
- * @return SIPVoIPLink* A pointer on the object
- */
- static SIPVoIPLink* instance( const AccountID& id );
-
- /**
- * Destructor
- */
- ~SIPVoIPLink();
-
- /* Copy Constructor */
- SIPVoIPLink(const SIPVoIPLink& rh);
-
- /* Assignment Operator */
- SIPVoIPLink& operator=( const SIPVoIPLink& rh);
-
- /**
- * Try to initiate the pjsip engine/thread and set config
- * @return bool True if OK
- */
- bool init(void);
-
- /**
- * Shut the library and clean up
- */
- void terminate( void );
-
- /**
- * Event listener. Each event send by the call manager is received and handled from here
- */
- void getEvent(void);
-
- /**
- * Build and send SIP registration request
- * @return bool True on success
- * false otherwise
- */
- int sendRegister(AccountID id);
-
- /**
- * Build and send SIP unregistration request
- * @return bool True on success
- * false otherwise
- */
- int sendUnregister(AccountID id);
-
- /**
- * Place a new call
- * @param id The call identifier
- * @param toUrl The Sip address of the recipient of the call
- * @return Call* The current call
- */
- Call* newOutgoingCall(const CallID& id, const std::string& toUrl);
-
- /**
- * Answer the call
- * @param id The call identifier
- * @return int True on success
- */
- bool answer(const CallID& id);
-
- /**
- * Hang up the call
- * @param id The call identifier
- * @return bool True on success
- */
- bool hangup(const CallID& id);
-
- /**
- * Hang up the call
- * @param id The call identifier
- * @return bool True on success
- */
- bool peerHungup(const CallID& id);
-
- /**
- * Cancel the call
- * @param id The call identifier
- * @return bool True on success
- */
- bool cancel(const CallID& id);
-
- /**
- * Put the call on hold
- * @param id The call identifier
- * @return bool True on success
- */
- bool onhold(const CallID& id);
-
- /**
- * Put the call off hold
- * @param id The call identifier
- * @return bool True on success
- */
- bool offhold(const CallID& id);
-
- /**
- * Transfer the call
- * @param id The call identifier
- * @param to The recipient of the transfer
- * @return bool True on success
- */
- bool transfer(const CallID& id, const std::string& to);
-
- /** Handle the incoming refer msg, not finished yet */
- bool transferStep2(SIPCall* call);
-
- /**
- * Refuse the call
- * @param id The call identifier
- * @return bool True on success
- */
- bool refuse (const CallID& id);
-
- /**
- * Send DTMF
- * @param id The call identifier
- * @param code The char code
- * @return bool True on success
- */
- bool carryingDTMFdigits(const CallID& id, char code);
-
- /**
- * If set to true, we check for a firewall
- * @param use true if we use STUN
- */
- inline void useStun(bool use) { _useStun=use; }
-
- inline bool useStun( void ) { return _useStun; }
-
- /**
- * The name of the STUN server
- * @param server Server FQDN/IP
- */
- void setStunServer(const std::string& server);
-
- std::string getStunServer (void) { return _stunServer; }
-
- /**
- * Terminate every call not hangup | brutal | Protected by mutex
- */
- void terminateSIPCall();
-
- /**
- * Terminate only one call
- */
- void terminateOneCall(const CallID& id);
-
- /**
- * Build a sip address with the number that you want to call
- * Example: sip:124@domain.com
- * @param to The header of the recipient
- * @return std::string Result as a string
- */
- std::string SIPToHeader(const std::string& to);
-
- /**
- * Check if an url is sip-valid
- * @param url The url to check
- * @return bool True if osip tell that is valid
- */
- bool SIPCheckUrl(const std::string& url);
-
-
- /**
- * Send an outgoing call invite
- * @param call The current call
- * @return bool True if all is correct
- */
- bool SIPOutgoingInvite(SIPCall* call);
-
- /**
- * Start a SIP Call
- * @param call The current call
- * @param subject Undocumented
- * @return true if all is correct
- */
- bool SIPStartCall(SIPCall* call, const std::string& subject);
-
- /**
- * Get the Sip TO url (add sip:, add @host, etc...)
- * @param to_url The To url
- * @return std::string The SIP to address
- */
- std::string getSipTo(const std::string& to_url, std::string hostname);
-
- /**
- * Tell the user that the call was answered
- * @param
- */
- void SIPCallAnswered(SIPCall *call, pjsip_rx_data *rdata);
-
- /**
- * Handling 5XX/6XX error
- * @param
- */
- void SIPCallServerFailure(SIPCall *call);
-
- /**
- * Peer close the connection
- * @param
- */
- void SIPCallClosed(SIPCall *call);
-
- /**
- * The call pointer was released
- * If the call was not cleared before, report an error
- * @param
- */
- void SIPCallReleased(SIPCall *call);
-
- /**
- * Handle a re-invite request by the remote peer.
- * A re-invite is an invite request inside a dialog.
- * When receiving a re-invite, we close the current rtp session and create a new one with the updated information
- */
- void handle_reinvite (SIPCall *call);
-
- /**
- * SIPCall accessor
- * @param id The call identifier
- * @return SIPCall* A pointer on SIPCall object
- */
- SIPCall* getSIPCall(const CallID& id);
-
- /** when we init the listener, how many times we try to bind a port? */
- int _nbTryListenAddr;
-
- /** Increment the number of SIP account connected to this link */
- void incrementClients (void) { _clients++; }
-
- /** Decrement the number of SIP account connected to this link */
- void decrementClients (void);
-
-
- /**
- * Set Recording
- * @param id The call identifier
- */
- // void setRecording(const CallID& id);
-
- /**
- * Returning state (true recording)
- * @param id The call identifier
- */
- // bool isRecording(const CallID& id);
-
-
- /**
- * Return the codec protocol used for this call
- * @param id The call identifier
- */
- std::string getCurrentCodecName();
-
- int inv_session_reinvite (SIPCall *call, std::string direction="");
-
- bool new_ip_to_ip_call (const CallID& id, const std::string& to);
-
- std::string get_useragent_name (void);
-
- private:
- /**
- * Constructor
- * @param accountID The account identifier
- */
- SIPVoIPLink(const AccountID& accountID);
-
- /* The singleton instance */
- static SIPVoIPLink* _instance;
-
- /**
- * Enable the SIP SRV resolver
- * @param endpt The SIP endpoint
- * @param p_resv Pointer to receive The DNS resolver instance
- *
- * @return pj_status_t PJ_SUCCESS on success
- */
- pj_status_t enable_dns_srv_resolver (pjsip_endpoint *endpt, pj_dns_resolver ** p_resv);
-
- void busy_sleep(unsigned msec);
-
- /**
- * Initialize the PJSIP library
- * Must be called before any other calls to the SIP layer
- *
- * @return bool True on success
- */
- bool pjsip_init();
-
- /**
- * Delete link-related stuuf like calls
- */
- bool pjsip_shutdown(void);
-
- pj_status_t stunServerResolve();
-
- /** Create SIP UDP Listener */
- int createUDPServer();
-
- bool loadSIPLocalIP();
-
- std::string getLocalIP() {return _localExternAddress;}
-
- /* Flag to check if the STUN server is valid or not */
- bool validStunServer;
-
- /** The current STUN server address */
- std::string _stunServer;
-
- /** Local Extern Address is the IP address seen by peers for SIP listener */
- std::string _localExternAddress;
-
- /** 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;
-
- /* Number of SIP accounts connected to the link */
- int _clients;
-};
-
-#endif