diff --git a/tools/build-system/launchpad/sflphone/debian/changelog b/tools/build-system/launchpad/sflphone/debian/changelog new file mode 100644 index 0000000000000000000000000000000000000000..78cc17db26580026556cd5c42e8213c39c364043 --- /dev/null +++ b/tools/build-system/launchpad/sflphone/debian/changelog @@ -0,0 +1,3138 @@ +sflphone (1.1.0-rc20120607~ppa1~SYSTEM) SYSTEM; urgency=low + + ** SNAPSHOT 1.1.0-rc20120607~ppa1~SYSTEM ** + + * gnome: cleanup logging + * * #12061: gnome: fix video codec list display + * * #12050: gnome: fixed regression in active codec handling + * #11732: Do not align menu with other interface component + * * #11818: gnome: stop daemon on SIGTERM, SIGINT or SIGHUP + * #10304: updatePlaybackScale dbus method uses int 32 bit for size and + position (allow for 24 days long recording playback) + * #10304: Add time lable for seekslider + * * #10725: gnome: seekslider: fixed compiler warning due to missing + header + * * #11732: gnome:calltree: fix warning when searchbar is not created + * * #11252: daemon: removed deprecated zrtp code + * #10725: Consolidate test to determine if there is a recording, + remove log on NULL pointer + * #10725: Remove logging from stop callback in seekslider + * #10725: No need of playback related callbacks in uimanager + * #10725: Remove logging in seekslider when no selected call + * #11732: Align Icons with dialpad in gnome client + * #10304: Make sure that the playback won't be updated after reset + * #10304: Prevent playback widget reset from stoping dtmf + * #10304: Move Playback widget at the bottom of the main window + * #10304: Fix logic to determine outgoing/incoming calls for recording + icon + * * #11685: gnome: fixed memory leaks in config menus + * #10304: Fix reset seekslider that prevent scale to be updated at + first read + + -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Thu, 07 Jun 2012 16:04:59 -0400 + +sflphone (1.0.0-rc20110930~ppa1~SYSTEM) SYSTEM; urgency=low + + ** SNAPSHOT 1.0.0-rc20110930~ppa1~SYSTEM ** + + * update kde .gitignore + * Fix bug in volume widget + * More polishing for release + * Bump version to 1.0.0 + * [#7023] Add the ability to load an abstract contact backend in the + library to resolve more data, polish code + * [#7021] More cleanup for release + * Cleanup + * [#7021] Refactor KDE client dbus handling, add a missing call in + daemon and port the DataEngine to the new API + * Remove some annoying debug + * merge language scripts + * remove obsolete 'VERSION' files + * update install instructions + * Add missing translations to gnome + * language update + * Revert "Don't reference count DBus clients, exit core immediately + when one of them request it" + * Don't reference count DBus clients, exit core immediately when one + of them request it + * [7021] Add contact abstraction support + * [#7121] Polishing library (over). Indentation, spacing and naming + are now consistent + * codecs: link to libccrtp, don't use logger + * Fix a daemon bug + * [#7038] Fix adding contact + * * #7037 : stop audio stream after all calls have been hanged up + * [#7025] Add full support for bookmark + * SFLPhone KDE do not destroy history anymore + * Fix config skeleton + * Close the daemon once and for all, no more automatic respawning + * Fix "unregistered account" bug (I hope so) + * Close SFLPhone at the right place, it still respawn, I don't know + why + * Remove dead code + * Fix regressions introduced in the last commit + * Dead code elimination 1/3 + * Fix bug, add "add contact" option, fix warning + * * #7019: Fix IAX codec negociation + * Remove or comment unnecessary/unhelpful debug output + * Fix "same as local" account setting, fix IP2IP LED color + * Add support for some more advanced config options and add missing + config dialog icons + * Fix crash with noise suppressor + * Alternative can now be selected from the call view context menu + * Add drag and drop support, initial context menu and fix 3 bugs in + the account dialog + * Add basic history drag and drop support + * Complete contact support is back + * * #6991 : fix IAX problems + * Fix IAX accounts being disabled by default + * Revert "deb: forge -g flags for pjsip" + * * #5884: Disable debug code in pjsip + * echo suppressor : more assertions + * Don't let the daemon think crypto is enabled when it's not + * Simplify ToneList + * Some progress on contact support + * Remove unused getRegistrationCount() + * remove annoying debug + * revert SIP bit of e27e5c39bad27bae28f574eb2cba7717e8956229 + * Simplify CallManager::placeCallFirstAccount + * Fix crash on hold + * * #6905 : SIP refactor + * gnome client: be sure key exchange is set correctly + * Move code into createSipTransport + * Fix account registration on start + * ManagerImpl::registerAccounts(): simplify + * * #5884: don't mess with pjsip threads in echo suppressor + * * #6905 : simplify udp/stun/tls pjsip transport creation + * Restore and improve support for Call history + * fix launchpad build + * SIPVoIPLink: simplify / refactor + * Fix libwidget linking + * SIP: simplify + * IM : simplify + * gnome: remove some debug + * AudioRtpFactory::stop() cannot fail + * * #6905: simplify SIP code + * pjlib: fix build without SSLv2, fix warnings + * Port history to the new syntax + * Test a dock widget based implementation for contact and history + * Disable SSLv2 support from pjsip and sflphone + * deb: forge -g flags for pjsip + * Fix deb packaging to get debug symbols + * remove debug + * pjproject: update to last stable release (1.10) + * Require gtk >= 2.20 and glib >= 2.24 + * tlsadvanceddialog: simplify + * * #6902 : fix errors spotted by -DGSEAL_ENABLE + * Update daemon dbus XML and port KDE config backend from dbus to + local + * Remove unused but set variables + * * #6929 : fix IM widget, cleanup + * Unconditionally enable debug symbols + * Should fix many KDE issues + * * #6886 : hitting backspace on empty number have no side effects + * * #6905 : fix AudioCodecFactory access in optimized builds (-O > 0) + * Remove unsupported and broken jaunty/karmic packages + * * #6902 : avoid using some gtk deprecated functions + * Update dbus introspection files + * * #6904: removed unused contactmanager + * * #6903 : use correct dbus-cxx package name + * * #6902: don't use individual gtk headers + * Fix a segfault when config is not present + * Merge latest (0.9.13) KDE code. This version is not yet ready for + git master, but better than the previous one + * addressbook : simplify + * * #5659 : sflphone-plugins doesn't depend on libedataserverui + * * #5659 : addressbook doesn't use libedataserverui + * gnome client doesn't depend on evolution + * * #5695: addressbook: simplify + * * #5695: addressbook : remove AddrBookHandle from plugin + * * #5695 : addressbook : remove unused stuff in the client + * * #5695 : addressbook : remove unused stuff, use static mutex + * gnome client doesn't use evolution + * gnome: use proper API to set GTK_CAN_FOCUS + * * #6897: removed unused focus state vars/callbacks + * gnome: fix calls to sflphone_fill_codec_list_per_account + * * #6623: gnome: don't leak in mainwindow + * gnome: mainwindow whitespace cleanup + * gnome: actions.c parameter doesn't have to be a double pointer + * * #6895: fix memleaks, cleanup in accountconfigdialog + * * #6893: fixes segfault in client on clean history + * * #6894: fix leaks, cleanup in sflnotify + * daemon: fixed prints in main + * * #6892: simplify, fix leaks in dialpad + * * #6887: audiopreference creates audio layer + * * #6660: use const char * const, not std::string for globally + visible constants + * * #6852: Preferences now solely responsible for audiolayer creation. + * * #6860: refactor uimanager, also fixes #6865 + * * #6853: hangup as soon as all digits have been deleted + * * #6852: alsa: retry if device is busy + * * #6852: audiolayer creation depends only on preference.audioApi + * * #6850: gnome: fix build for gtk < 2.22.0 + * cleanup in iax + * alsa: typo + * pulse: if we can't peek in audio input, we can't drop samples + * * #6849: show error window if codecs are missing, instead of dying + * EchoCancel: unused, remove + * * #6629 : use number of samples as arguments for audio filters + * * #6629 : remove unused Algorithm interface + * * #6629 : use helper to call alsa functions and display error msgs + * Remove unused type + * * #6841: fix some error handling + * * #6629: simplify AlsaLayer::alsa_set_params() + * Get gdk key definition from header + * * #6828: Replace raw key codes by gdk defines + * remove some debug, enhance some other + * mainbuffer: simplify + * * #6561 : fix phantom call after transfer + * Conference Participant set : simplify + * SIPCall: remove unused functions, make invite session public + * * #6229 : remove malloc/free from pulse audio loop + * * #6629 : simplify pulse callbacks + * * #6629 + * Simplify widgets + * * #6629 : keep the correct audio module when frequency changes + * * #6751: fixed erroneous debug msgs + * callable_obj.h: removed unneeded pthread header + * alsalayer: cleanup + * * #6629: Always restart audio driver when changing parameters (ALSA + only) + * gnome GUI: don't block in DBus signal errorAlert() + * * #6629 : simplify AudioLayer creation + * * #6629 : remove unused and unconfigurable frameSize from audiolayer + * * #6629 : remove unused error message from audio layer + * Fix logic error when switching audio API + * Remove unused AudioProcessing class + * AudioRtpRecordHandler::initNoiseSuppress() : use noiseSuppress + directly + * * #6629 : use DC blocker directly in audio layers + * * #6629 : clean AudioLayer + * * #6629 : don't store mainbuffer inside audiolayer + * * #6629 : correct AudioLayer::notifyincomingCall() + * * #6554: cleanup, refactoring in sipvoiplink + * * #6554: cleanup in iaxvoiplink + * * #6554: throw exception in getSIPCall if pointer is NULL + * * #6554: make some methods of sipvoiplink static + * * #6655: cleanup in managerimpl + * * #6554: refactoring, fix memleaks in sipvoiplink + * * #6478: remove throw specs, cleanup in voiplink + * * #6629 : remove unused AudioDevice + * * #6655: removed more dependencies from managerimpl + * * #6744: simplified numbercleaner + * conference : remove one prototype + * * #6743: fix ip2ip + * Don't give glib warnings if icons are not found + * gnome: fixed includes + * Codec.h: removed unused function + * * #6742 : clean dbus & icons + * * #6699: refactor/cleanup accounts + * icons: cleanup + * timer : use second precision, not millisecond + * calltree_update_clock : use correct type, returns something + * * #6737: fixed typo in dbus call + * * #6737: removed tests for removed API + * * #6737: dbus: fixed bug from merge + * * #6737: cleanup in accountlist + * * #6737: cleanup in dbus + * * #6740 : fix history double free + * * #6740 : remove time updating thread from calls + * * #6737 : use c99 for client + * * #6738 : make history loading faster + * sipvoiplink : don't crash on transfers + * fixed typo + * Remove unused file + * Don't build networkmanager.cpp at all if NM is disabled + * _debug* -> _debug + * * #6554 : simplify sipvoiplink + * hudson: added -x to git clean command + * added git clean to hudson script + * audiocodecfactory: cleanup + * * #6718: refactored setTlsSettings into SIPAccount + * * #6718: removed more unused methods + * * #6718: refactored confmanager code into sipaccount + * remove unused functions + * * #6718: confmanager: removed more unused methods + * AudioCodecFactory : cleanup + * #6697 : Turn callableElement struct into union + * * #6718: confmanager: removed more unused methods + * * #6718: confmanager: removed more unused methods + * * #6718: removed unused dbus methods, refactoring + * * #6699: accounts: cleanup/refactoring + * * #6699: refactoring, cleanup in accounts + * * #6699: more account cleanup + * remove unused autoconf variable + * * #6714: fixed hudson script + * make distclean in hudson + * added || exit 1 to run_tests.sh call + * * #6714: fixed make distcheck for sflphone-plugins + * * #6714: fixed make distcheck for gnome client + * * #6714: fixed make distcheck for daemon + * git: #6698 split the main .gitignore file + * gnome: gpointer is already a pointer + * gnome: calltab_init: use calloc instead of malloc + * * #6699: more account cleanup + * * #6699: cleanup account + * * #6554 : more *voiplink cleanup + * * #6558 : more sipvoiplink simplification + * * #6558: saner loadSIPLocalIP prototype + * gnome: #6623 clean calllists + * * #6692: more audiolayer cleanup + * * #6692: cleanup/refactoring in audiolayers + * * #6692: more forward declarations, AudioThread->AlsaThread + * * #6692: audiolayer cleanup + * * #6692: alsalayer cleanup + * * #6558 : remove account creator + * * #6558 : clean sipvoiplink + * * #6554 : cleanup sipvoiplink + * audiortp: cleanup + * * #6657 : fix launchpad builds for good + * * #6675 : send RTP dtmf events only once + * * #6655: more cleanup + * AudioRtpSession::updateSessionMedia() : simplify + * * #6655: more cleanup in managerimpl + * * #6655: removed more code, cleanup + * * #6655: more cleanup, fixed infinite loop + * * #6655: removed more unused files + * * #6655: removed unused mutex + * * #6655 removed more unused code + * * #6655: removed unused methods + * * #6655: cleanup in main + * * #6663: fixed segfault when off hold from transfer + * * #6658: user's active codec selection is respected + * * #6660: static global string should be static const char* const + class member + * * #6659: use g_strcmp0, not strcmp for vals that may be null + * callable_obj: fix double free + * calltree_display_call_info() : simplify + * * #6657: Fix launchpad builds + * Logger::log() : simplify + * AudioRtpSession : privatize members + * * #6655: more constness, cleaned up/simplified methods + * * #6654: call DBus::_init_threading so that dbus-c++ to make it + threadaware + * set default credentials on account creation + * AudioCodecFactory::scanCodecDirectory() : simplify and correct + * * #6623: fixed typos + * * #6623: fixed more leaks + * * #6623: fixed more leaks + * * #6623: fixed more leaks, don't print codec name if null + * * #6623: more leaks fixed in client + * * #6623: fix more leaks, fixed some warnings + * * #6623: fixed leak in history + * updated gitignore + * initialize dbus dispatcher correctly + * Fix tests, hudson doesn't have a dbus daemon running + * remove unused code + * removeCall() : simplify , fix leak + * stopRtpThread() : simplify + * *CurrentCall : simplify + * Fix memleak + * fix serialization of audio api (pulse / alsa) + * account map : simplify + * remove call from callmap before terminating it, avoid use after free + * * #6630 : don't make DBusManager a singleton + * call: return confID by value + * add back history code deleted by error + * history : reverse logic + * simplify history serialization and remove some debug + * remove annoying debug + * * #6464 : replace cerr with _error + * * #6464: replace cout with logger macros + * replace printf() with logger macros + * update .gitignore + * remove unused function + * update eclipse projects + * uimanager_new() : simplify + * rename directories + * celt: simplify a bit + * Fix CELT configure.ac test + * * #6612 : template speex codecs + * * #6623: refactored conference obj + * * #6623: refactored callable object, removed leaks + * * #6623: more cleanup, fix leaks, make global vars static and rename + them + * * #6623: calltree: fixed memleaks, simplified code. + * audiolayer: init pointer members + * manager: catch exception on invalid hangup + * * #6623: don't leak on calls to create_new_call + * * #6611 : clarify codecs prototypes + * ringtones : .au and .ul files are both ulaw + * * #6611 : make sure samplerate converters are called correctly + * ManagerImpl::switchAudioManager() : simplify + * * #6623: fixed more leaks + * * #6623: fixed more leaks + * * #6623: fixed more leaks + * * #6623: fixed leak, line-endings in imwidget + * * #6627: zero-initialize pointers if they're going to be deleted + * * #6628: don't leak calls on exceptions + * Revert "audiortp: call join after calling stop on RtpThread" + * sflphone-client: more constness + * audiortp: call join after calling stop on RtpThread + * * #6625: return 0 on successful completion + * * #6624: fix segfault on servercallfailure + * * #6621: Fixed double free, unlock mutex in ManagerImpl::terminate + * * #6220: remove audio stream when peer hangs up + * * #6596: AudioSymmetricSession shouldn't self-delete + * resampler: grow internal buffers dynamically + * merge up and down sampling => resampling + * Leave test directory unchanged when running make check + * audio algorithms : remove unused prototype + * ringtone: detect codec from file extension + * *AudioFile : simplify + * * #6596: create local SDP on the stack, not the heap + * * #6596: don't call Ost::Thread::terminate from dtor + * audiofile: cleanup (samplerate -> unsigned) + * remove unused func + * samplerateconverter: cleanup + * RingBuffer::Put() : remove unused return value + * MainBuffer::putData() : remove unused return argument + * audiolayer::putMain() : remove unused func + * AudioLayer::putUrgent() : remove unused return value + * * #6618: delete any remaining ringbuffers in destructor + * RingBuffer::availForPut() : remove + * * #6617: return from main rather than calling exit + * MainBuffer::availForPut(): remove + * RingBuffer: simplify + * alsa : remove write only variable + * fix memcpy declaration + * bcopy(src, dst) -> memcpy(dst, src) + * RingBuffer::Get() : remove constant volume argument + * return a copy of the call ID, not just a reference. + * MainBuffer::getDataById() : remove volume argument (always 100) + * MainBuffer::getData() : remove constant volume argument + * RingBuffer::Put() : remove constant volume argument + * MainBuffer::putData() : remove constant (=100) volume argument + * audiolayer: remove constant _defaultvolume + * AudioRtpRecordHandler / AudioRtpSession : simplify + * mainbuffer: fix test + * iaxvoiplink : simplify + * sip registration callback: fix a dbus crash + * MainBuffer: simplify + * AudioRtpFactory: return cached type of rtp session. The rtp session + can have disappeared if the call was put on hold + * AudioRtpFactory: remove unused setters + * Fix launchpad builds + * * #6611 : remove unused bandwidth codec information + * * #6611: AudioCodec: remove useless/unused setters + * make sure buffer string is initialized correctly + * * #6596: declare certain destructors virtual + * audiolayer : cleanup + * Simplify doc build rules + * * #6270: don't build dbus-api doc with make, should require make all + * configure.ac: cleanup + * Remove copy of dbus-c++ from libs/ + * * #6596: stop clock thread when peer hangs up + * removed unused Fmtp.h + * * #6595: more logical initialization order + * * #6600 : fix account creation + * * #6601 : fix configure.ac tests + * remove unused variable + * Don't mix stack and heap based allocations + * Fix copyright (2009, 2008, 2009 -> 2008, 2009) + * Fix warnings found by clang + * * #6595: fix initialization order for AudioRTP + * * #6592: removed typedef std::string CallID + * * #6586: implement local g_slist_free_full for older glib versions + * * #6579: fix memory leaks in client (there's a lot left) + * ShortcutPreferences::setShortcuts() : simplify + * Fix merge + * * #6548: remove call to non thread-safe strerror() + * AudioRtpFactory: each instance is associated to exactly one SipCall + * create_audiocodecs_configuration() : make static + * * #6269 : refactor AudioRtpSession + * Fix AudioSymmetricRtpSession.h inclusion guard (cherry picked from + commit c3081dce1cc1370d6d3558a4c4ef5cfac0d21caf) + * * #6269: Rename AudioRtpSession to AudioSymmetricRtpSession + * * #6574: Don't exit when connection to pulseaudio server fails + * accountconfigdialog.h : remove some stuff from header + * * #6560: fix configuration test + * Fix warning in test + * * #6560: don't hide password entry in security tab + * * #6560: set initial password for SIP accounts + * * #6506: remove useless pointer indirection + * * 6560: password is now specific to IAX accounts + * * #6560 : actually use, store, restore, transmit SIP credentials + * * #6560: YamlEmitter: serialize sequences + * YamlEmitterException: typo + * ManagerImpl::computeMd5HashFromCredential() : simplify, fix memleak + * * #6561: invite_session_state_changed_cb() : simplify + * * #6561: More useful debug in VoIPLink::removeCall + * * #6561 : fix ghost call reappearing in GUI after transfer + * while -> for (make the code smaller) + * * #6558 : Account::loadConfig() : move IAX code to IAXAccount + * IAXVoIPLink::getAccountPtr : simplify + * * #6554 : access the SIPVoIPLink directly, not per account + * SIPVoIPLink is instanciated only once and is not associated to a + single account + * yamlnode: use const references when possible (still some left to do) + * Account::_accountID: constify + * VoIPLink: simplify, remove unused method + * hudson test : no need to call run_tests.sh anymore + * Remove AccountID type and AccountNULL define + * Make check runs the test (no need to call run_tests.sh manually + anymore) + * gnome GUI: Fix tests + * Revert "Move registration information from SIPAccount to + SIPVoIPLink" + * * #6392: pluginmanagertest: fix warnings reported by valgrind + * * #6547 : remove unused exceptions + * * #6547: CallManagerException: use runtime exceptions + * * #6547: InstantMessageException: use runtime exceptions + * * #6547: do not throw exceptions if some settings are not present in + config file + * * #6547: YamlParserException: use runtime exceptions + * * #6547: VoipLinkException: use runtime exceptions + * * #6547: YamlEmitterException: use runtime exceptions + * * #6547: DTMFException: use runtime exceptions + * * #6547: AudioFile: use runtime exceptions + * * 6547: AudioZRtpSession: remove impossible error case + * * #6547 : AudioRtpSession: remove impossible error case + * * #6547: AudioZrtp: use runtime exceptions + * * #6408 : send authenticationUsername to GUI + * * #6408 : store/restore authenticationUsername from config file + * SIPAccount: simplify + * Move registration information from SIPAccount to SIPVoIPLink + * SIPAccount::getAccountDetails : simplify + * * #6540: yaml parser: simplify + * sdp.cpp : fix a warning + * * #6540: yaml parser : remove std::string typedefs + * * #6540: Simplify yaml unserialization + * * #6540 : add a Conf::ScalarNode constructor for booleans + * setAccountDetails(): simplify + * * #6408: store authentication username in daemon + * * #6408: Be able to set the authentication username in the GUI + * * #6507 : do not crash if the program is not sflphoned + * Fix tests + * macroify SIPAccount::unserialize() + * Move all .cpp files from sflphoned target to libsflphone.la, except + main.c + * main() : simplify, return positive error codes + * * #6507 : find codecs dir in build directory + * * #6392: Sdp: move clean functions to destructor + * AlsaLayer::adjustVolume() : simplify + * alsalayer : reduce indentation + * malloc/free -> new/delete + * malloc/free -> new[]/delete[] + * malloc/free -> new/delete + * AudioSrtpSession: simplify base64 encoding + * * #6392: Initialize std::string from pj_str_t correctly + * * #6392: AudioRtpSession: Initialize remote port + * Audio settings : Initialize _echoCancelTailLength and + _echoCancelDelay(0) + * Initialize variable + * YamlParserException : fix use of stack variable after it has been + deallocated + * * #6392: fix memory leak in history + * * #6392 AudioCodec : fix memory leak + * * #6392 : fix memory leak in sip account + * * #6408: clean up sipaccount (cosmetics mostly) + * sipaccount.cpp serialize() : reduce number of lines + * * #6392: invalid memory access + * * #6392 : fix invalid memory access + * * #6479: merged useful code from MimeParameters into Codec interface + * * #6462: fixed hangup on IP2IP call + * added run_daemon.sh script + * test: remove unused variable + * Remove functions only used by a failing test (cherry picked from + commit fcf718cb75de7f1882dc61c07bb8d300dfa10f85) + * * #6360 : make client tests build (cherry picked from commit + 028b2835f040e51ab8ab979b32732b07b8798fce) + * * #6360 : fix warnings in check_global test (cherry picked from + commit 9e2bd6a7496dd64f6f48595e385760019aab1193) + * * 6360: updated API calls in tests, but they're not building yet + (cherry picked from commit 548f6f0f919b43772a3e9c667e5e292791281795) + * Fixed include in tests (cherry picked from commit + aeadc7525c1e31f936670ac8b02f0bcf387c38a8) + * Remove unused variables and functions + * IAX: fix warnings (cherry picked from commit + fd7a113a11cac2cd9a7c36929e88ad28195c4c35) + * Remove unused DEBUG define which interferes with logger.h (cherry + picked from commit b2f72b91d0f43cb1dd94d138882a8caa9c841c24) + * * #6392: no need to check for account NULLity since it is + dereferenced above + * * #6392: fix a memory leak, replace by stack allocation + * * #6392: remove a variable assignement which confuses cppcheck + * process_conference_participant_from_serialized() : remove unused + function + * * #6392: s/free/g_free/ + * * #6392: fix a memory leak in abookfactory_load_module() + * * #6392: remove generate_call_id() used only once + * * #6392: fix memory leak (opendir() without closedir()) + * * #6392: AudioRecorder(): ensures mbuffer is set + * Remove SFLPHONED_VERSION from global.h, use autoconf PACKAGE_VERSION + * #6298: Cleanup + * #6331: Fix deleting ringtone file after call have been answered + * * #6330: merged user_cfg into headers + * #6298: Fix conference recording file update at conference end + * #6298: Fix record file name serialization for conference + * * #6295: cleanup of codec hierarchy + * #6298: Fix gtk warnings + * * #6300: added script to run tests + * #6109: Add recording playback for conference + * * #6300: tests do not require an installed sflphone + * * #6295: re-removed clone methods + * #6109: Fix gtk_critical warnings for incoming calls + * #6109: Fix GTK_CRITICAL warning + * #6109: Fix icons when history is not activated + * #6109: Fix warnings + * #6109: Implement stop recorded file playback signal + * Revert "* #6295: removed unused clone method" + * * #6295: removed unused clone method + * * #6296: removed non existant file from Makefile.am + * #6109: Stop fileplayback for outgoing call + * #6109: Implement stop recording playback button + * Fix binding names errors in dbus introspection file + * #6109: Implement playback recorded file callback in client + * #6109: Store recorded file path on client side + * #6109: Add dbus methods for call recording playback + * * #6290: remove unused classes from utilspp + * * #6288: cleanup sdp + * * #6288: fix exception usage + * * #6288: simplify SdpException + * * #6288: cleanup in sdp.cpp/h + * #6109: Only display playback button if record file is set and valid + * * 6290: updated configure.ac to remove functor Makefile + * * #6290, #6289: removed unused classes from utilspp, fixed make + check + * #6109: Add button for history playback of recorded file + * * #6289: removed unused observer class + * * #6282: forward declare sdpMedia in sdp.h + * * #6281: renamed setCallAudioLocal->setCallMediaLocal + * #6183: Handle conference with more tahn two calls + * #6183: Fix history icons when calling back a conference from history + * #6183: Fix icons inconsistencies in history for conference hang up + * #6183: Fix toolbar actions when selecting a conference in history + * #6183: Fix conference serialization + * #6268: Serialize only calls + * * #6269: removed useless type testing + * ignore some files in test/ + * * #6268: Remove dead class AudioSymmetricRtpSession + * #6251: Do not had history calls in calllist when loading history + file + * #6251: Fix insertion in history map in before saving history file in + daemon + * #6251: Fix history unit tests + * #6251: Order the list before serailization, get rid of the hashtable + in history + * #6251: Implement history serialization using a list wether than a + map + * * #6253: remove external audioport from header, make all members + private + * * #6253: don't store external local audio port (used for NAT) in + Call + * #6251: Add start_time timestamp in history serialization + * #6251: Fix call insertion in conference items + * #6233: Fix serialized account list terminated with a ";" character + * #6238: Fix draggable history calls into current calls + * #6233: Fix toolbar updates + * #6233: Fix history + * * #6235: remove pyc files from git tree + * #6233: Handle cases when one or manuy calls are unreachable in + createConfFomrParticipantList + * #6233: Handle wrong numbers in createConferenceFromParticipantList + * #6231: Fix drag-n-drop issue + * * #6173 : move sippxml in tools + * #6231: Fix merging issue + * #6183: Implement conference unserialize + * * #6212: remove extraneous flags from globals.mak + * #6183: Unserialize conference data in conference + * #6183: Add account information in request for conference call from + history + * #5755: Add -ldl to liker in sflphone + * #5755: Fix fedora 15 compilation issue + * #6183: Serialize conference participant phone number and account + * #6183: Add conference timestamp in serialization + * * #6186: don't include global.h, just logger.h + * #6183: Fix saving history to file + * #6183: Fix removing call from calllist + * * #6184: remove pointers to Manager from AudioRtpSessions + * #6183: Calling calltree_add_call explicitely for history + * #6183: Ability to store conference inside history tab queue + * * 6181: remove unused API from sipcall + * #6171: Implment nreCallCreated callback + * #6167: Fix participant list NULL ending + * #6149: First draft of conference creation from history + * #6149: Fix multiple call/conf selection callbacks ... + * #6129: Fix place_call function called twice for pressing enter + action + * #6129: Fix double click action for history + * #6149: Add dbus call for creating conference from history + * #6129: Fix placing call from history and addressbook (still need to + fix icon) + * * #6148: removed unused AudioRtpFactory constructor + * * #6145: remove unused isAudioStarted + * * #6145: remove unused isAudioStarted + * #6129: Add conference into history, fix call/conference selection + * * #6143: don't use getType outside of serialization methods + * * #6132: forward declarations instead of includes + * * #6132: add constness, remove redundant "inline" keywords + * #6129: Add timestamp to conference object to order history entries + * * #6128: remove unused forward declarations from header + * * #6127: make noncopyable class actually noncopyable + * * #6125: don't include AudioRtpFactory in sipcall.h + * #6123: Fix alsa ringback audio file + * #6123: Fix raw audio file loading problem + * #6109: Fix daemon plugin manager unit test + * #6109: Fix history manager unit tests + * #6109: Recording filename in daemon and client for history items + + serialization + * #6109: Refactor AudioFile to play recorded call + * * #6104: AudioCodec moved to sfl namespace + * * #6099: remove active flags from codec classes + * #6095: Add notification-daemon as a runtime dependencies for rpm + packages + * #6095: Fix fedora 15 compilation in MineParameters.h + * #6095: Declare static variable explicitely for client + * #6095: Add logs to build OSC build machine + * * #6098: global variables should have file-scope to avoid name + conflicts + * #6095: Fix compilation error for Fedora 15 + * #6095: Update SFLphone version to 0.9.14 + * #6095: Add specification file in opensusse build service for + sflphone-plugins + * #6073: Fix sflphone-plugins build on launchpad + * #6093: Rename CodecDescriptor for AudioCodecFactory + * * #6089: fix warnings in make check + * * #6086: renamed codecs methods to audio_codecs + * * #6085: renamed codec related dbus calls to audio_codec + * #6065: Remove g_print from client, use DEBUG instead + * #6065: Add actions name for addressbook + * * #6085: renamed codecs* widgets/functions audiocodecs* + * #6065: Fix Addressbook runtime warnings + * #6065: Replace Codecs tab for Audio in account preference dialog + * #6065: Fix "transfert" typo + * #6065: Fix addressbook action runtime warning in uimanager + * * #6082: fixes make check by adding libcrypto libs to test + dependencies + * #6073: Rename plugin/addressbook folders for addressbook/evolution + in sflphone-plugins + * #6074: Removed AC_SUBST from configure.ac when using + PKG_CHECK_MODULE + * #6073: Fix sflphone-plugins package build + * #6073: Fix sflphone-common build + * #6065: Fix runtime gtk warning when initializing searchbar without + addressbook + * #6063: Fix mozilla-tellify gitignore + * #6063: Remove stream copy file using ifdef macro + * * #6012: fix make dist for sflphone-common + * #6063: Update .gitignore file + * #6058: Fix base64 encoding related warnings + * #6056: Fix SdpException handling + * #6055: Fix unknown pargma warning for gcc <= 4.5 + * * #5949: test gcc version before disabling unused-but-set warning + * #6054: Fix addressbook plugin compilation warning + * #6048: Fix uimanager static initialization + * #6046: Fix addressbook factory static initialization of member + addrbook + * #5979: Fix implicit function declaration warning + * #6042: Fixed discarding qualifier warnings in client + * #6041: Fix instant messaging unhandled case warning + * #5994: Implement set current addressbook name and search type in + addressbook plugin + * #5994: add rules for launchpad packaging of addressbook plugin + * #5994: Fix addressbook plugin configuration loading + * #6027: Fix addressbook enabled test from configuration + * #6027: No need of gnomedoc related macros in addressbook plugin + * #6027: Add NEWS file required for build + * #6027: Add addressbook plugin autogen.sh script + * #6027: Remove plugins from client + * #6027: Add sflphone-plugins folder at project's root level + * #5994: Move addressbook folder from contacts to plugin folder + * * #6011: removed unused Makefiles + * * #6010: remove unused headers + * * #5952: fix "string constant to char*" warnings + * * #6009 fixed warnings + * * #6003: finished cleanup of account classes + * * #6003, #6004: cleanup of account classes, defaultAccount no longer + global + * * #6000: fix memory leak of args object + * * #5998: removed using namespace std from networkmanager + * * #5998: removed "using namespace std" from ZrtpSessionCallback + * * #5998: removed using namespacestd from AudioZrtpSession.h + * * #5998: remove "using namespace std" from auriorecord.h and + MimeParameters.h + * * #5998: remove using namespace std in main + * * #5998: removed "using namespace std" from logger + * * #5949: test gcc version before disabling unused-but-set warning + * #5994: Installation of addressbook plugin + * #5979: Implement codec full addressbook search from plugin + * #5979: Implement addressbook factory and plugin + * * #5981: unused webwidget removed + * #5966: Account config synchronization fix (for stun) + * #5954: Handle media name exception + * #5954: Fix audio codec name display in client + * #5954: Clean up getSessionMedia methods + * * #5957: getRecordingSmplRate returns a value + * #5954: Clean up getCurrentCodec methods + * * #5950: remove "converting to non-pointer type 'int' from NULL" + warnings + * #5915: Full gain control version + * * #5949: remove more unused variable warnings + * * #5949: remove unused/unused-but-set variable warnings + * * #5949: show_preferences_dialog returns a success value + * * #5946: cleanup of include directives, undefined function + * * #5515: comment out SSLv2 calls in pjsip + * #5915: Implement different slope for attack tme and release time for + gain control + * #5915: use only one input signal for gain control (removed output + buffer) + * #5921: Fix no audio after holding a conference + * #5916: Add gaincontrol files + * #5916: Implement FFMPEG/CCRTP video streaming prototype + * #5903: Fix call transfer during a conference + * #5915: implement rms detector, first order averager, limiter for + gain control + * #5914: Fix call transfer when no notification request is required + * #5899: Fix conference right-click segfault + * #5884: temporary fix segfault in pjsip memory pool + * #5883: Fix compilation issues on maverick and lucid + * #5755: Fix fedora 15 compilation without patching ccrtp + * [#5855] Make echo canceller optional + * #5855: Fix echo suppression activation/deactivation + * #5855: Implement pjsip echo canceller + * #5814: Speex initialization function uses samples, not bytes + * #5814: Test using more unbalanced signals + * #5814: Fix buffer size for long echo length or long echo delay + * #5814: Adjust level for echo cancellation at runtime + * #5814: Process noise reduction before echo cancelling + * #5814: Implement speex post echo canceller processing + * #5814: Dump echo cancel file to disk + * #5814: Add parameters for echo cancel + * #5809: Add configuration parameters + * #5809: Implement speex echo canceller in audio rtp session + * #5814: Code cleanup + * #5814: Fix conf creation with several incomming ringing calls + * #5814: Fix conf creation segfault when dragging a call on hold on a + ringing call + * #5809: Added unit test for echo cancellation and implemented + "process" virtual method + * #5709: Add always recording option in configuration + * #5709: Add always recording option in audio conference panel + * #5709: Add core functionnality for always recording (missing config + options) + * #5769: Fix conference participant handling (detach/attach) and hold + actions + * #5747: Fix recording icons and state for conference when adding new + participant + * #5769: Code cleanup + * #5769: Fix hangup unsent calls + * #5769: Fix remove/add additional participant to conference + * 5769: Several fixes concerning confererence handling + * #5769: Fix compilation error + * [#5769] Fix audio streams binding in main buffer + * #5769: Removed access to audio mixer from audio layer + * #5765: Fix audio crash for illformated wavefiles + * #5765: Add maximum iteration for finding fmt and data "chunck" + * #5589: Fix compilation of libnotify under + * #5757: Fix abort signal when receiving INFO + * #5747: Add usersDetached.svg + * #5747: Handle offhold action for recording conference + * #5747: Fix off hold action for conferences + * #5747: Implement update conference in record action in calltree + * #5747: Add new icons for recording conferences + * #5747: Add recording state for conferences + * [#5738] Remove getAudioDriver call from manager (replace by + _audiodriver var) + * [#5738] Refactor mutex protecting audiolayer + * [#5737] Fix HD conference recording + * [#5730] Fix start audio session after changing sampling rate + * [#5714] Fix enter keyboard event for addressbbok and history + * [5695] Fix addressbook combo box update when no addressbook selected + * [#5695] Fix addressbook initialization and search bar update + * [#5695] Add mutex for books_data in addressbook to protect async + calls + * [#5695] Get back addressbook open from uri + * [#5695] Fix absolute addressbook URI for local addressbooks + * [#5695] Implement libebook 3.0 interface + * [#5571] Better logic for hangup (for case where call have not been + sent yet) + * [#5571] Update error handling in voip links + * [#5571] Fix compile time warnings + * [#5696] Fix installation dependencies for Natty + * [#5669] Add mention that sflphone.org is for testing only + * [#5693] Add natty in teh dput.conf file + * [#5690] Remove not useful logs + * [#5670] Use dynamic payload type for rtp dtmf + * [#5668] Clean up sflphone configuration logging + * [#5668] Fix hook checkbox configuration update + * [#5666] Fix unit tests + * [#5666] Manage event subscription + * [#5666] Emit bye request when subscription is terminated + * [#5666] Bye request should be sent after event subscription + notification is done on transfer + * [#5666] Make reinvite method static (to be called in pjsip + callbacks) + * [#5666] Hangup Call in manager for AccountNULL and IP2IP + * [#5589] Use PKG_CHECK_MODULE for every client's dependencies + * [#5623] Enlarge initial size of pjsip memory pool for calls (16k) + * [#5564] Fix audio recording resampling for g722 + * [#5571] Move attribute handling for onhold/offhold actions in SDP + session + * [#5571] Codec negotiation refactored and unittested + * [#5571] Implement tests + * [#5571] Implement pjsip negociator + * [#5571] Fix unit tests + * [#5571] Add Fmtp.h to repository + * [#5571] Integrate mime types and codec factory + * [#5571] Handle exception when SDP negotiation fails + * [#5570] Add sflphoned-sample.yml in repository + * [#5564]: Implement stereo to mono mixing for rigntone + * [#5342] Update audio stream initialization + * [#5514] Restore test ni historytest suite + * [#5514] Fix + * [#5514] Disable test_create_history_path + * [#5514] use pulseaudio in sample config file + * [#5514] Fix test: load history from file + * [#5514] Do not use X + * [#5513] Make unit tests compile successfully + * [#3947] Enable unit tests in Jenkins + * [#5454] Fix build system to handle new version number + * [#5454] Update languages from launchpad + * [#5454] Add --without-celt in OpenSuse build service + * [#5454] Change version number + * [#5331] Added first SDP session tests + * [#5273] Update nightly build version tags to conform dpkg rules + * [#5211] Refactor send register method for iaxvoiplink and + sipvoiplink + * [#3950] Remove call being transfered from calltree + * [#5211] Use appropriate memory pool for transport selector + * [#5211] Fix strict aliasing rules warning in pjsip + * [#5211] Bring back pjsip shutting down sleep to 1000 ms + * [#5211] Fix registration callback segfault when closing the + application + * [#5211] Use the dialog memory pool for Route header in INVITE + request + * [#5211] Add temporary memory pool for findLocalAddressFromUri and + findLocalPortFromUri + * [#5211] Use individual memory pool for dtmfs + * [#5211] SipVoipLink refactoring + * [#3950] Attended transfer for conference calls + * [#5284] Fix DNS resolution for Route with specified port number + * [#5284] Some code cleanup + * [#3947] Fix typo in hudson script + * [#5284] Added sip route to REGISTER, INVITE, BYE request, plus DNS + resolution + * [#5266] Use RTP dtmf as default + * [#5284] Added pjsip_process_route_set after setting routes in regc + structure + * [#5286] Fix parsing error due to long configuration file (removed + max event) + * [#5286] Fix false test in configuration emmiter + * [#5286] Code cleanup + * [#5286] Updated exception handling in configuration system + * [#4969] Fix put SRTP call on hold + * [#3950] Add debug messages + * [#3950] Ability to perform an attended transfer + * [#5276] Fix initialization problem in g722 + * [#3950] Add replace header in SIPVoIPLink::transferWithReplaces + method + * [#3950] Implemented attended method in SIPVoIPLink + * [#3950] Cleanup transaction request received callback + * [#3950] Implement dummy attended transfer in gnome-client + * [#5249] Fix audio samplerate update algorithm for g722 + * [#5249] Fix uninitialized variable used in conditional jumps + * [#5249] Fix conditional jump error in audiolayer (uninitialized + value) + * [#5267] Use autoconf 2.65 as a requirement (instead of 2.67) + * [#5267] Restore manual pjsip configuration and compilation + * [#5267] Autodetect celt version (0.9.1, 0.7.1) + * [#5267] Fix deprecated macros in gnome client configure.ac + * [#5267] Update configuration for libcelt-dev + * [#5267] Fix build autoconf and automake + * [#5227] Deactivate automatic call to astyle after compilation + * [#5242] Hangup every calls before leaving + * [#5237] Will now nightly-build for natty, Karmic deprecated + * [#5229] Use inner class for rtp thread instead of inheritance + * [#5211] Move mainbuffer unbind call in rtp final method + * [#5211] Initialize sip call memory pool using 16 kb + * [#5211] Use call memory pool in session reinvite + * [#5211] Add debug messages + * [#5211] Use and internal pool for calls + * [#5211] Reduce pjsip memory pool usage for stateless error messages + * [#5211] Refactor call deletion + * [#5212] + * [#5208] Refactor codec management for accounts + * [#5168] Remove printf from codec's encode & decode method + * [#5168] Fix celt compilation on launchpad + * [#5168] Fix sflphoned compilation warnings in audiocodec.h + * [#[#5168] Must keep the g722 specific RTP rate to avoid incoming + packet timeout + * [#5168] Fix static/dynamic payload rtp session update + * [#5168] Throw SIPVoipLink Error if codec not instantiated in new + outgoing call + * [#5168] Fix dynamic/static codec payload type ambiguity + * [#5169] Fix doubled IP2IP profile when no config file + * [#4867] Add gtkinfobar in configuration panel + * [#4867] Disable input/output/ringtone selection when using default + alsa plugin + * [#4952] Patches for possible buffer overflows + * [$4885] Fix schemas problem + * [#4885] sflphone.schemas not present during build + * [#4885] Add gconf shemas directories in opensuse build system + * [#4885] Add file/folder ownership for opensuse-factory build system + * [#4906] Fix opensuse-factory build + * [#4885] Update name dependency for libedataserver + * [#4885] Fix non-void function without return in dbus-c++ + * [#4895] Update language translation + * [#4896] Update session timestamp when updating media + * [#4896] Reapply RTP hack for G722 payload type + * [#4896] Update recording sampling rate when updating codec + * [#4897] Save codecs in config for each configuration changes + * [#4895] Do not save config when sflphone quit + * [#4885] Update date for copyright + * [#4885] Deactivate siptest that require more than one sipp instance + * [#4879] Remove inmcoming call notification from IAX + * [#4885] Some cleanup + * [#4874] Add setCancel immediate/deffered for ost::Thread + * [#4879] Fix incoming call notification + * [#4878] Set keyboard focus on searchbar when selecting addressbook + * [#4874] Fixed compilation warning + * [#4874] Fixed compilation warning in sipvoiplink + * [#4874] Fix compile time warning in RTP record handler + * [#4874] Fix conditional jump in SDP + * [#4874] Fix conditional jump based on uninitialized value + * [#4874] Store call id within rtp thread context + * [#4874] Fixed conditional jump based on uninitialised value in + conference + * [#4871] Fix default account fetching + * [#4870] Delete RTP session when Refusing an incoming call + * Restore IP to IP call + * [#4857] Fix audio codec negotiation problem + * [#3947] Adjust ressources allocated to compilation + * [#3947] Disable unit tests in Hudson + * [#4305] Free mutex only when really quiting SFLphone + * [#4859] Update copyright to 2011 in every source file + * [#3218] Character '.' stripped by the caller engine + * [#4854] Fix typos, desktop entry + * [#4847] Apply RTP modification to ZRTP session + * [#4852] Update Karmic and Lucid dependencies + * [#4852] Add Libedataserver and libedataserverui as gnome client + dependencies + * [#4852] Add authentication mechanism for EDS + * [#4851] Fix segfault when closing pulseaudio layer too rapidly + * [#4808] Some otehr cleanup + * [#4808] Made some cleanup + * [#4808] Added mutex in rtp session for codecs and noise process + * [#4847] Update audio processing when updating RTP media + * [#4842] Add support for linking with gold/ld --no-add-needed + * [#4808] Make update g722 related static/dynamic payload logic + * [#4827] Upper limit on the number of contacts to import from EDS is + hard-coded to 500 + * [#4808] Fix put call on/off hold + * [#4808] Implement early RTP start for incoming calls + * [#4808] Audio stream is no longer start within RTP session. + * [#4808] Removed coupling between audio layer and and RTP session + * [#4702] Start audio rtp session as soon as it is created + * [#4702] Init timestamp to 0 + * #4702: Send RTP packets immediately, no need of outgoing queue + * [#4784] Update dbus-c++ version from gitorious + * [#4702] Update RTP timeouts + * [#4702] Lengthen RTP timeouts + * [PATCH] Fixed compatibility with old libtool versions. + * [PATCH] Accept older libebook (Maemo 5 has 1.4.2) + * [PATCH] Fixed double-free error in preferences dialog + * [PATCH] Fixed building of sflphone-common on Maemo5 + * [PATCH] Improved Gnome client initialization error handling. 1. It + no longer segfaults when sflphoned isn't available. 2. User is + provided with GUI error dialog. + * [PATCH] Improved autogen.sh scripts 1. They do not require bash + anymore 2. Added workaround for Debian bug #565663 3. Replaced + manual autotools invocations with single autoreconf call 4. Non-zero + return status on failure + * Revert "[#4468] libtool <= 2.2 doesn't have LT_INIT macro so + AC_PROG_LIBTOOL should be used instead." + * Revert "[#4468] Libebook 1.4 is sufficient" + * Revert "[#4468] Apply big path on dbus communication system" + * [#4468] Apply big path on dbus communication system + * [#4468] Libebook 1.4 is sufficient + * [#4468] libtool <= 2.2 doesn't have LT_INIT macro so AC_PROG_LIBTOOL + should be used instead. + * [#4639] Fix determining default addressbook if this property is not + set in gconf + * [#4639] Fix memory leaks in Addressbook + * [#4637] Fix opening default addressbook at sflphone init + * [#4622] Free yaml events while parsing configuration file + * [#4623] Fix conditional jumps based on uninitialized variable + * [#4622] Fix leaks in yaml serialization engine + * [#4616] Fix addressbook warnings + * [#4514] Adjust RTP timestamp + * #4527: Rename Karmic libyaml and Celt package in debian control file + * #4495: Rework addressbook opening loop + * [#4524] Increment RTP count when sending data + * [#4524] DO NOT start RTP session twice + * [#4367] Use PKG_CHECK_MODULE for celt + * [#4367] Fedora package celt as celt (not libcelt) + * [#4367] Astyling + * [#4367] Update .po files + * [#4367] Fix segfault in gensin + * [#4354] Make celt a direct dependency on launchpad opensuse build + service + * [#4367] Make celt a required package, option --without-celt valid + * [#4367] Fix zrtp timestamping error + * [#4367] Fix audio zrtp timing + * [#4367] Dispatch ZRTP packets + * [#4367] Fix segfault when unloading account map + * [#4367] Fix zrtp session + * [#4367] Implement on packet receive + * [#4367] use symetric audio rtp session, not dual + * [#4367] Reduce packet receive/sent timeout + * [#4367] Reduce RTP timeouts + * [#4367] Move speaker data receive + * [#4367] Move speaker data receive + * [#4367] Move receive speaker data method + * [#4367] Remove debug in rtp session + * [#4367] Fix g722 codec clock rate + * [#4367] Fix noise suppression initialization + * [#4367] Fix segfault in RTP mic fadein method + * [#4367] Refactor mic data encoding in rtp session + * [#4367] Implement RTP main loop + * [#4367] Fix compilation problem + * [#4367] Fix AudioRtpclass using TRTPSessionBase + * [#4367] Fix AudioRtpSession putDtmfEvent shadowing + * [#4367] Fix AudioRtpSession putDtmfEvent shadowing + * [#4367] Refactor RTP session (phase 2) + * [#4367] Refactor RTP session (phase 1) + * [#4367] Remove Redeclaration of SymetricAudioRtpSession in + rtpfactory + * [#4265] Add continue statement in for loop for invalid addressbook + * [#4261] Makes addressbook initialization more robust + * [#4257] Add maverick in build system + * [#4233] Add sdp related unit tests + * [#4233] Add condition and signal in two incoming call test + * [#4243] Fix segfault in AudioSrtpSession + * [#4243] Fix memory leak in AudioSrtpSession + * [#4243] Make audio srtp optional in for incoming call + * [#4243] Add boolean variable to make sure remote crypto context + initialized only once + * [#4243] Add documentation to AudioSrtpSession + * [#4243] Use 80 bits authentication tags by default + * [#4243] Init audio srtp remote crypto context in + call_on_media_update + * [#4243] Move SDP negotiastion in mod_on_rx_request + * [#4243] Implement initLocalCryptoInfo to be called at different + momment + * [#4243] Init init local crypto context in when initializing audiortp + * [#4243] Change key length according to sdes negociation + * [#4243] Associate callid to accountid for incoming calls + * [#4242] Fix no SDES keys in IP2IP calls + * [#4242] Fix no SDES keys in IP2IP calls + * [#4233] Test for call on/off hold + * [#4233] Add two incoming call test + * [#4233] + * [#4233] Add 2 outgoing simultaneous call unit tests + + -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 30 Sep 2011 13:44:57 -0400 + +sflphone (0.9.7~rc1~ppa1~SYSTEM) SYSTEM; urgency=low + + ** 0.9.7~rc1~ppa1~SYSTEM ** + + * [#2462] Set explicitly the transport on incoming call too + * [#2462] fix typo + * [#2462] Use different address for SDP and call IP + * [#2462] Use published address in SIP-SDP + * [#2181] Fixed changelog files + * [#2181] Updated spec file + * [#2402] Fix pointer to int conversion warning (atoi) + * [#2402] Remove daemon warnings, make indent + * [#2459] Make sure the stream is opened when the call is answered + * [#2402] Add conference related picture in documentation + * [#2443] Not much ... + * [#2399] Fix dialing display problem + * [#2450] Fix incoming call already in conference crash + * [#2399] Display peer name on the first line and peer number on the + second + * [#2450] Handle 403 FORBIDDEN when refused + * [#2447] Bind offHold/onHold actions to button in gtk client + * [#2447] Bind hangup action to button for conference + * [#2447] Add conference action in gtk client's ToolBar + * [#2381] Disable the password hashing in config file + * [#2402] Cleanup + * [#2366] Set callback to null when deleting Pulseaudio streams + * [#1313] Fix main buffer unit test + * [#1313] Fix audio layer unit test + * [#2315] Hide pw in security tab, display when editing, sync with + basic tab + * [#1313] UnitTest change AudioRtpSession for AudioSymetricRtpSession + instance + * [#2402] Code cleanup + * [#2444] Add debug to catch occasional crash when loading client's + config + * [#2444] Add debug info to catch occasional crash when loading config + dialog + * [#2402] Restore Call menu translations + * [#2403] Use the published address if checked in GUI + * [#2442] Add protection test in sdp + * [#1841] Reapply pjsip patch concerning DNS SRV resolution + * [#2384] Tags incoming call as direct SIP call, if applicable + * [#2402] Change the monkey face + * [#2315] Enable user to display password in clear text + * [#2434] Force optimization level at 2 + * [#2284] Fix dbus_get_all_ip_interface compilation warnings + * [#2431] Popup main window on incoming if applicable + * [$2402] Fix simple warnings + * [#2402] Fix implicit variable init order in LibraryManagerException + * [#2402] Fixing implicit variable initialization warnings in + AudioRtpSession + * [#2402] Revert atoi change, fixing codec list doubled entries + * [#2402] Fix gpointer to gint conversion + * [#2402] Fix pointer casting to integer different size warning in + codec list + * [#2402] Fix warning discarting qualifiers from pointer target + * [#2402] Fix gtk tree view assignement from incompatible type warning + * [#1669] Fix audio recording folder utf-8 non compatibility issue + * [#2414] Clean up debugs + * [#2414] Use transport set in iptoip Account and update it frm + preference + * [#2348] Use macro N_() to mark ui.xml strings as translatable + * [#2414] Rename getSipAddress/setSipAddress functions + * [#2407] Fix volume controls display + * [#2407] Fixes dialpad + * [#2383] Set ip to ip config when clicking apply button + * [#2404] Update call-to script - Maxime Chambreuil + * [#2405] Client handles unknown call in current state as well + * [#2383] Add DBUS signal to send IPtoIP local address and port as + string + * [#2383] Add Ip to IP config change apply call back + * Clonflict + * [#2402] Code cleanup + * [#2383] Do the same for IPtoIP (init localn ip with first in the + list) + * [#2383] Use first interface in the list if local addresss is not + defined + * [#2403] Clean up unuseful addresses/ports + * [#2403] Use the IP profile SIP port as global SIP port + * [#2383] Fix dbus_get_all_ip_interface warnings + * [#2383] Take into account sameAsLocal when loading published address + * [#2383] Tsake into account sameAsLocal option when saving published + address + * [#2383] Update local ip address in ip to ip config + * [#2383] Save ip 2 ip local port in config + * [#2406] Update toolbar at startup + * [#2284] Remove redefinition warnings + speex warnings + * [#2383] Fix security table in account config + * [#2383] Save ip 2 ip network interface parameters in config + * [#2403] Restore sip transport selector + * [#2383] Fix filling the Localt IP Address on account creation + * [#2383] Fix Gtk-Critical when checking STUN + * [#2383] Fix reopening account configuration display issue + * [#2383] Load IPtoIP local address and port in preference iptoiptab + * [#2383] Add LocalAddress and Localport in Preference IpToIp tab + * [#2403] Use the address and port associated to the account as often + as possible + * [#1753] Removed pjsip generated files + * [#1753] Removed remaining milenage lib references + * [#2383] Add _publishedSameasLocal variable in sipaccount + * [#2383] Add PUBLISHED_SAMEAS_LOCAL variable in config + * [#2383] Fix stun set active or not when opening config + * [#2181] Added RPM 64bits dbus patch + * [#2402] Code indentation + * [#2313] Force $(HOME).cache directory creation at startup + * [#2383] Separate network interface and published address in account + config + * [#2400] Change dbus service installation path to libdir + * [#2382] Move TLS related published address options in security tab + * [#2382] Indent accountconfigdialog.c + * [#2181] Install libdbus-c++ in $pkglib instead of $lib + * [#1753] Remove ILBC code and disable it by default in the configure + * [#1753] Remove milenage directory + * [#2382] Fix switching interaface instabilities + * [#2396] Save local ip in account creation wizard + * [#2284] Remove warning on hold + * [#2387] Fixes history searching and filtering + * [#1215] Add samplerate display in the GUI + * [#1663] Voicemail icon reflects voice messages + * [#2395] Fix account registration ( specifically with callcentric) + * [#2386] Strip "sip:" on incoming call, fixing history call back + * [#2181] Updated spec files + * [#1215] Display codec name in calltree instead of status bar + * [#2390] Move back nbCalls and stopStream higher in refuseCall + * [#2392] Fix ringtone during call in IAX + * [#2391] Stop audio streams when there is 0 calls only + * [#2391] Add debug when call state is not valid + * [#2390] Clear returns in IAXvoipLink::sendAudioFromMic() method + * [#2380] Fixing IncomingCallNotification not regular + * [#2339] Query conference at client startup + * [#2339] Working conference querying at startup + * [#2339] Add conference in call tree + * [#2339] Primitives to query conferences at client startup + * [#2320] Add account selection in history + * [#2355] Temporary solution: do not delete pointer when removing + account + * [#2380] Change algorithm in AudioRtp to trigger an + IncomingCallNotification + * [#2274] Comment sdebug in MainBuffer flush method + * [#2274] Add flushMain() in ManagerImpl::addStream + * [#2274] Add getBufferID() method in ring buffer + * [#2274] Fix warning, comment debug in ringbuffer's flush method + * [#2274] Use AudioLayer flushMain() and flushUrgent() in ALSA + * [#2274] Clean up unused variable warning + * [#2274] Protect minbudffer pointer on flushing + * [#2274] Fix playATone method which writing empty buffer in urgent + ringbuffer + * [#2274] Use audio layer flushUrgent and flushMain in createStreams + * [#2274] Use flush audio calls from audiolayer + * [#2274] Flush when peer answered call + * [#2375] Flush main buffer in iax when answering a call + * [#2274] Parse displayname using c++ string method + * [#2375] Flush main buffer when off holding calls + * [#2375] Flush main buffer mon RTP startup + * [#2376] Use now Pulseaudio module-cork-music-on-phone + * Updated OSC packaging + + -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 20 Nov 2009 13:59:02 -0500 + +sflphone (0.9.7~beta~ppa1~SYSTEM) SYSTEM; urgency=low + + ** 0.9.7~beta~ppa1~SYSTEM ** + + * [#1933] Cleanup debug + * [#1933] Clean up debug + * Fix mic + * [#1933] Set the IAx format earlier + * [#1933] Move IAX sendAudioFromMic outside if (call) statement + * [#1933] Fix startstream when offhold in iax and add debug concerning + codec neg. + * [#2371] sflphone_notify_voice_mail: minor gettext message formatting + cleanup + * [#2371] select_account_cb: properly gettextize status message + * [#2371] show_account_list_config_dialog: properly gettextize status + message + * INSTALL: Minor tidyup of core install guide + * Add /sflphone/src/icons/Makefile to .gitignore + * [#2181] Updated OpenSUSE files (tmp) + * [#1933] Add debug for codec negociation for iax + * [#1933] Get rid of getMicAvail and getMicData in audiolayer (not + used anymore) + * [#1933] Add "audio codec not determined" error in IAX + * [#1933] Test flush data + * [#1933] Do not need to start audio stream in iax anymore + * [#1933] Protecting pointer + * [#2284] Remove more compilation/execution warnings + * [#2284] Cleanup debug in client, use DEBUG instead of g_print + * [#2284] Clean up uimanager + * [#2370] Remove warnings + * [#2366] Clean up other debug + * [#2366] Clean up debug + * [#2366] Call pa_xfree explicitely in writeToSpeaker + * [#2284] Remove address book warnings + * [#2365] Fixes bad cast + * [#2352] Fix continuous ringing when peer hangup and call not yet + answered + * [#2181] Added version support + * [#2181] Fixed some minor issues + * [#2360] Moved MainBuffer from AudioLayer to ManagerImpl + * [#2352] Makes getMainBuffer() everywhere + * [#2352] Use 50 sec latency on pulseaudio stream creation + * [#2352] Add alsa debug + * [#2359] Update repository documentation + * [#2354] Move pulseaudio disconnectAudioStream after stopping main + loop + * [#2352] Adjust nb byte copied in pulseaudio according to + writeableSize + * [#2352] Specify pulseaudio tlength parameters using pa_usec_to_bytes + * [#2322] Convert italian translation to UTF-8 + * [#2357] Fixes window size + * [#2357] Display only actionnable tool item + * [#2333] Update streams parameters + * [#2347] Use GNOME user settings for Menu and Toolbar appareance + * [#2349] Load/Save properly audio params + * [#2322] Update translations from Launchpad + * [#2181] Added Francois Marier script + * [#2350] Remove non-valid test + * [#2181] Updated launchpad packaging + * [#2333] Fix Pulseaudio Capture + * [#2333] Use pulseaudio ADJUST_LATENCY flag and ALSA RT-SCHEDULING + * [#2333] Pulseaudio Interpolate timing + * [#2333] Change (again) Pulseaudio settings to fit logiteck usb hdw + requirement + * [#2333] Adjust pulseaudio fragment size to 4096 (max sflphone's + frames per buffer) + * [#2284] Remove recurrent compilation warning (g++ linker problem) + * [#2333] Safer Audiostream parameters + * [#2333] Fix alsa playback to reduce underrun + * [#2333] Better audiostream parameters + * [#2181] Updated version management + * [#2333] Exclusive test in playback loop + * [#2181] Updated build system + * [#2333] Less underrun with these value + * [#2333] Update playback audiostream parameters + * [#2333] Lengthen the audio buffer reduce number of underrun in + pulseaudio + * [#2333] Add ALSA recovery functions for underrun (begin) + * [#2333] Add pa_stream_trigger in pulse audio underrun callabck + * [#2048] Reduce prebuffering in pulseaudio (which affect incomming + calls' plbck) + * [#2316] Do not display any icons to the right on the history tab + * [#2333] Comment pa_stream_trigger in pulseaudio underrun + * [#2333] Modify pulseaudio streams parameters + * [#2318] Fix transfer tool button double signal + * [#2181] Updated + * [#2333] Fix ALSA ringtone + * [#2333] Flush all main buffer before starting audio + * [#2333] Open/Close Alsa thread between calls while there is no audio + * [#2333] Add debug message and test condition on starting playback + and capture + * [#2181] Fixed gnome client makefile + * [#2181] Updated + * [#2308] Remove getTelephoneTone debug + * [#2308] Change plughw for default in ALSA + * [#2308] Oups, forgot to change function name in audiolayertest.cpp + * [#2308] Cleanup in pulseaudio code (debug, function name) + * [#2308] Fix pulseaudio stream closing assertion failure + * [#2308] Moved pulseaudio mainloop locking from AudioStream + disconnect stream + * [2308] Fix latency at the beginning of a call, when playing DTMF and + wehn starting tone + * [#2181] Updated karmic + * [#2317] [#2319] Fix address book toggle button contextual behaviour + * [#2308] Stop stream when refusing a call + * [#2308] Stop pulseaudio stream when peer hungup + * [#2308] Fix tone and ringtone + * [#2312] Display the STUN entry widget when opening the tab + * [#2308] Implement two different callbacks for capture/playback in + pulseaudio + * [#2309] Open/close pulseaudio connections in startStream/stopStream + * [2308] Leave pulseaudio stream running, do not cork/uncork them + anymore + * [#2295] Set gtk file chooser to None if nothing is set in + configuration + * [#1976] Add codec and conference documentation + * [#2209] Fix recording in regard of resamling + * [#2297] Update .gitignore + * [#2297] Update translation files + * [#2297] Add reference to our coding standards + * [#2297] Remove old docbook code + * [#2296] Reinit tls account settings after modification + * [#2253] Add DcBlocker class to remove capture's dc offset + * [#2034] Fixes for TLS transport to initialize + * [#2284] Add silent build rule + client clean warnings + * [#2274] Fix unserialize history items in cilent at startup + * [#2274] Complete display name parsing and displaying + * [#2274] Parse the Display Name in sip INVITE message + * [#2050] Fix capture volume control in ALSA + * [#1970] Volume controls disable when using pulseaudio + * [#1970] Disable volume controls when using pulseaudio + * [#2277] Fix direct ip2ip ZRTP enabling/disabling in ip2ip + preferences + * [#2181] Added launchpad debian files + * [#2181] Added spec files for OSC + * [#2274] Set display name for "Contact" sip header as the hostname + * [#2181] Fixed daemon issues + * [#2181] Fixed gnome client issues + * [#1976] Remove warnings - need to fix the transfer + * [#2006] Add init is_rec variable in ManagerImpl + * [#2006] Update codec display on call selection + * [#2006] Restore double click actions in history and contact calltree + (GTK) + * [#2176] use XDG_CACHE_HOME when initializing sfl.zid file + * [#1976] Fix calltree switching from history + * [#2209] (Re)Fix cache for zid + * [#2209] Clean up debug messages + * [#2209] Clean debug messages + * [#2209] Fix trasnfering a call during a conference + * [#2209] Speex decode must return the number of bytes + * [#2209] Change frameSize speex 32kHz + * [#2209] Fix speex codec framesize + * [#2209] Reinit converterSamplingRate in RTP sessions + * [#2209] Change speex ultra wide band framesize + * [#1747] Add pixmap data + * [#2252] Fix Receiving a server error 488 crashes the callee + * [#2209] Fix iax low rate packate sending + * [#2209] Clean up debug messages + * [#2209] Add resampling changes for IAX + * [#2209] Clean up resampling code + * [#2209] Fix latency introduced by pulseaudio + * [#2209] Fix initialization of mainbuffer's internal sampling rate + * [#2176] Fix upsampling buffer size in audiolayer + * [#2209] Add dynamic converter sampling rate in audiortp sessions + * [#1747] Fixes runtime warnings + * [#1747] Remove from repo + * [#1747] register our icons to be used as stock icons + * [#2209] Fix number of byte in alsa's write to speaker + * [#2209] Fix putting non-resampled data in RTP's mainbuffer + * [#2209] Add alsa resampler + * [#2209] Add a samplerate converter in PulseLayer + * [#2209] Add mainbuffer's internal sampling rate and flushall method + * [#2176] Add mainbuffer stateInfo debug method + * [#2209] Resampling is optimal using SRC_LINEAR not SRC_FASTEST + * [#2176] Remove debug recordings + * [#2176] Fix Holding a conference participant on new calls + * [#2224] Add confID in callable object + * [#2176] Fix putting onhold a call participating to a conference when + pressing new call + * [#2176] Reset auidio buffers when adding streams (rtp, audiolayer) + * [#1976] Use xml to describe toolbars - Add a naviguation toolbar + * [#2176] Remove conference default_id in joinParticipant + * [#2176] Display error message in alsa's snd_pcm_avail_update call + * [#2176] Alsa mic avail data debug + * [#2176] Add some debug message for mic loss problem + * [#2176] Flush mic ring buffer when offholding a call + * [#2176] Reset ringbuffers' readpointer when adding main participant + * [#2176] Fix getAvailData algorithm + * [#2176] Reset ringbuffer's readpointer when adding a new participant + to a conference + * [#1744] Regex object renamed to Pattern. Previous attempt at + providing + * [#2176] Fix detach main participant problem when adding new one + * [#1976] Use right domain to translate + * [#1976] Add xml menu description + * [#2176] Store a list of confernece participant in client + * [#2176] Fix add participant, joinparticipant methods + * [#2181] Do not install dbus-c++ headers + add return value + * [#2176] Fix minor call handling instabilities + * [#2174] Fix incoming IP call contact address + * [#2211] Add test to protect NULL pointer + * [#1163] Add Advanced account configuration section + * [#2176] Add some usefull comments and debugging info + * [#2176] Add conditions to display security icons in conference + * [#2176] Fix detaching one participant while keeping communication to + others + * [#2176] Reenable userActive.svg in call tree + * [#2176] Make user active blue (not red) + * [#2176] Fix user active picture + * [#2176] Fix "hidden" merge conflict in sipvoiplink + * [#2176] Remove iax audio stream on peer hungup + * [#2174] Multiple UDP transports functional (TESTED with 2 accounts + and 3 calls) + * [#2176] Fix fix audio stream binding in iax + * [#2174] Create a default UDP transport + use tp selector for dialogs + also + * [#2176] Register iax audio stream in mainbuffer + * [#2176] Fix getAudioCodecName in IAXvoipLink + * [#2176] Fix iax account init + * [#2176] Handle multiple account using the same sip transport + * [#2165] Add .png files + * [#2176] Small fixes concerning dtmf + * [#2176] Fix make uninstall in codecs + * [#2174] remove stund makefile generation + * [#2176] Add conference lock + * [#2174] Add transport selector for multiple accounts + * [#2176] Change userActive picture from red to blue + * [#2176] Fix security pixbuff in calltree + * [#2176] Replace sfl.zid in .cache/sflphone instead of .sflphone + * [#2176] Fix add call description + * [#2176] Remove detach button from toolbar + * [#2176] Fix calltree call description state and state code in + conferences + * [#2176] Fix pulse audio double free + * [#2176] Fix conference selection + * [#2174] Clean up - remove stun settings in client network + configuration panel + * [#2174] Remove voviva stun code + * [#2174] Rsolve STUN with pjsip - DO NOT WORK + * [#2165] Add user svg + * [#2165] Debugging sip call failed + * [#929] Link against uuid if installed + * Oops + * Fixed bugs related to libsexy (with GTK < 2.16) + * [#929] Remove uuid-dev dependency in the core + * [#2165] Debugging no negociated codecs at communicatio start + * [#2165] Fix calltree bug (gtktreestore instead of gtkliststore) + * [#2165] Fix several merge problems + * Updated opensuse packaging script + * [#1163] Add missing figures + * [#1163] Update INSTALL file + * [#2165] Fix IAX + * [#2165] Add recordabe interface + * [#2165] Finish recording refactoring for call (not for conference) + * [#2165] Enable speaker recording for two different calls + simultanously + * [#2165] Implement call recording using the Recordable interface + * [#2165] Add get and set to AudioLayer's audio recorder + * [#2165] Add class recordable from which inherit call and conference + * [#2006] Fix G722 and Speex 8khz codec conferencing + * [#2006] add recording of audio buffers + * [#1163] Add general settings section + * [#1163] Fixes makefile error + * [#2006] Fix some minor issues + * [#2006] Drag a conference call on another conference call + (difference conferences) + * [#2006] Fix dragging a conference on itself + * [#1744] Integrating some of the needed regular expression patterns + in order + * COmplete call features + * [#1744] Added support for named subgroup in the Regex object. Also, + new + * [#1744] Adds thread safety features, compile() and setPattern() + methods to the Regex class. + * [#1744] Fix inconsistency in the finditer method from the last + commit. + * [#1744] Added regex pattern object built on top of libpcre. To be + used + * [#1744] Initial commit towards implementing RFC4568. Unimplemented + in the + * [#2157] Hide "security" and "advanced" tabs for IAX under account + * [#1163] Add call features section + * [#2006] Add joinConference capabilities + * [#2006] Add dbus joinConference signal + * [#2006] Drag a conference call onto a conference to add it + * [#1163] Add addressbook section + * [#2006] Drag a conference call onto a single call to create a + conference + * [#2006] Expand rows automatically + * [#2006] Add minimal multiple conference handling + * [#2006] Add atached/detached conference icons + * [#2006] Add function processRemainingParticipant + * [#2006] Deep refactoring, fix hangup bug + * [#1163] Update documentation - Accounts part + * [#1976] Integrate user doc to gnome client build system + * [#2122] Remove double inclusion in dbus-c++/src/Makefile.am + * Remove pjproject version number + * [#2006] Fix peerHungup + * [#1976] Make Yelp accessible from the GNOME client (need to install + the sflphone.xml first) + * [#2006] Fix multiconferencing hangup + * [#2006] Fix hangup calls in a conference + * [#2150] Make IAx2 reappear + * [#2006] Fix detach participant on multiple call + * [#2006] Can remove rining call from a conference + * [#2006] Reinit confID when removing a participant + * [#2006] Remove get isCurrentCAll in hangup/peerhungup (SipVoipLink) + * [#2006] Fix refuse call + * [#2006] Fix answerring incoming call + * [#2006] Refactor conference's participant list + * [#2101] Re-integrate test compilation in main build system + * [#2101] Make the test directory compile + * [#2136] Restore history functionality + * [#2006] Fix binding main participant to himself + * [#2006] Fix add current/incoming/onHold participant to an existing + conference + * [#2006] Fix add incoming calls to an already created conference + * [#2006] Fix remove stream + * [#2006] Fix detachParticipant/removeParticipant switchCall ids + * [#2006] Fix adding a call in conference having state "CURRENT" + * [#2006] Remove/add main participant from conferences + * [#2006] Hold/unHold conference + * [#2006] Detach a partcipant from drag n drop + * [#2006] Hangup a conference + * [#2006] Add hold/unhold conference dbus messages + * [#2034] gtk-ui fix under the "basic" tab. + * [#2006] Fix dragging calls on conference calls + * [#2006] Fix detach participant from a conference + * [#2034] Added default message is status bar under the account config + dialog + * [#2112] Fix a crashed caused when a non-md5 password was sent to + pjsip. + * [#2006] Detach participant by ID + * [#2006] Fix addParticipant method in managerImpl to handle + incoming/answered calls + * [#2006] Add addParticipant method in managerimpl and related dbus + messages + * [#2111] Added the ability to configure zrtp on sip.sflphone.org from + * [#2106] Fixed problem in the account assistant under gtk-ui. Also, + assistant.c + * [#2006] Fix dragging a conference call on another conference call + (same conference) + * [#1904] Small UI fix. Assistant was moved from "Call" to "Edit" + menu. + * [#1904] Fix a wrong label under gtk-ui. + * [#2034] Renaming and source code splitting. + * [#2034] Status bar added to account window to better reflect the + registration + * [#2006] Make calltree_remove_call recursive (for GtkTreeStore) + * [#1110] Small gtk-UI fix in the account window (alignment). + * [#2006] Fix remove conference, display children which are still + active + * [#2006] Recursive function call in calltree_update_call + * [#2006] Add multilayered capabilities to calltree (GtkTreeStore) + * [#2006] Implement remove conference in calltree + * [#2034] Now useless as Direct Ip calls settings moved under + Preferences. + * [#2034] Edit/add buttons were set insensitive all the time under + gtk-ui. + * [#1887] Information about the state of the current SIP call is + displayed + * [#2006] Add call tree remove callback + * [#2006] Fix create_conference function + * [#2006] Update conference_added_cb to add new conference to the list + * [#812] Added new tab under GTK-ui Preferences. Moving Direct Ip + Calls from + * [#2121] Disable temporarily test compilation + * [#2006] Fix conferencelist to handle conference_obj_t instead of + gchar + * [#2006] Add conference_obj structure + * [#2121] Update version + * [#2006] Fix conference selection + * [#2101] Use the new source tree to fetch the right object files + * [#2006] Add conference in calltree + * [#2006] Add Dbus signal conference added/removed/changed + * [#2006] Add getConferenceDetails call on dbus + * [#1904] Registration expire now appears as a spin box under gtk-ui. + * [#812] Fixing a segmentation fault caused by a non-existing account + ID + * [#2006] Add getConfList method over dbus + * [#2006] Add a conferencelist data structure in client-gnome + * [#812] Defaults value are now sent if a non-existing account is + requested + * [#2006] Add sflphone action sflphone_join_participant + * [#2006] Fix buffer read pointer problem deletion + * [pjsip] Attempt at fixing via header incompatibility with + Freeswitch. + * [#1797] forget something + * [#2006] Add call new state conferencing in deamon + * [#2006] Remove addParticipant method for conference, use + joinParticipant only + * [#1163] Update INSTALL documentation + * [#812] Msec/sec values were not taken into account. + * [#1797] Make pjproject-1.4 compile + * [#2006] Add Detach participant method + * [#2006] Dragndrop fully functional with INCOMING and HOLD call + * [#1797] Add pjproject-1.4 + * [#1797] Remove pjproject-1.0.3 + * [#2006] Get call state in conference related function + * [#2006] Add joinParticipant (conference) method in ManagerImpl + * [#2006] Add joinConference DBUS message + * [#2006] Store the previously selected call_id on dragndrop + * [#2006] Fix GValue pointer unref in selection callback + * [#2006] Store dragged call_id + * [#2006] Update drag_data_received_cb callback to manipulate CallIDs + * [#2006] Add dragndrop signals + * [#2006] Set calltree reordable + * [#812] Adds the ability to create a TLS listener in case the user + requests + * [#812] Adds the ability to configure local/published address from + * [#1883] Move switchCall in onHoldCall function + * [#812] Deals with the published address/port problem when + integrating TLS. + * [#1883] Switch call id in managerimpl when peerHungUp + * [#1883] Switch call id before hangup + * [#1883] Add usefull and permanent debug info for conference + cretion/deletion + * [#812] Fix various segmentation faults related to Direct IP kind of + calls. + * [#1883] Fix deletion of std::map elements using iterators + * [#2014] Add libzrtpcpp build dependency + * [#1883] Still some for loop test ambiguity (while loop instead) + * [#1883] Fix for loop initial test ambiguity (use while loop instead) + * [#1883] We must discard data in urgent ring buffer if data is get in + mainbuf + * [#1883] Fix availForGet same id for ringbuffer and readpointer + * [#812] Match "sips" as a Direct IP Call when the user enter a sip + uri + * [#812] Fix segmentation fault related to SIP URI creation. + * [#812] Towards integrating multiple tls listeners at the same time. + This + * [#1883] Add debug messages in conference and fix mainbufferTest + * [#812] gkt-ui fix. Private key must be fed as a filename and not as- + is. + * [#812] TLS integration within sipvoiplink and pjsip. Also, + configure.ac + * [#1883] Fix Alsa/Pulse mallocation + * [#1883] Fix data corruption in AudioRtp's micData buffer + * [#812] Full dbus integration for all the tls related options under + gtk-ui. + * [#1883] Fix memory leaks in audiortp session + * [#1883] Fix mem leaks in audio rtp + * [#812] Fix setAccountDetails where TLS_ENABLE was set to the value + * [#812] Small gtk-ui fix. + * [#811][#812] Small gtk-ui fix. + * [#812] Introduced a mechanism for configuration files that makes + possible + * [#812] New dbus bindings added. Also, configuration compliance was + enforced + * [#1881] Remove default buffer from MainBuffer (update unit-tests) + * [#1881] Add ring buffer read pointer tests + * [#1883] Fix issues in ringbuffer reader pointers + * [#2034] Implementing a new configuration dialogue for TLS transport + settings + * [#1883] Add some usefull debug and safety checks + * [#2028] Notify the client with libnotify when the zrtp negotiation + failed. + * [#811] Harmless no to throw an exception, an makes the application + less + * [#2028] A minidialog is showed to the user under sflphone-client- + gnome + * Removed useless file. + * Ignoring Makefile in src/widget + * [#2027] Fix segmentation fault when showMessage callback is called + after + * [#2026] keyExchange was set to ZRTP instead of "1" + * [#2024] Fix the wrong summary at the end of the assistant. + * [#1883] Fix mnagerimpl conference map insertion + * [#1883] Add Mutexes in MainBuffer + * [#811] Gtk ui was not presenting the right information about zrtp + for + * [#2023] security icons were not installed in sflphone. + * [#2021] Fix a mistake in the readme from sflphone-common that gives + wrong + * [#811] The current SRTP mode was not properly displayed for the + IP2IP + * [#1743] Re-implementation of the "automatically remove error dialogs + [...]" + * [#2017] [#2019] Fix the inability to dial a number and place a + registered + * [#811] Final re-integration of ZRTP support in the main branch from + 0.9.6 + * [#1883] Fix map insertion methods + * [#811] Combo box now is now set to the active key exchange method + * [#811] ZRTP options now configurable back again from the Gtk UI. + IP2IP + * Updated hostname for git clone + * [#1883] Add minimal functionalities to create a conference + * [#811] re-integration of all the methods and signals on dbus. + ManagerImpl + * [#811] Got out of a precarious position were nothing would compile. + * [#1976] Build documentation squeleton with docbook + * [#1883] Add sflphone-client "addParticipant" button for conference + * [#1994] Better organize the source directory structure. New + subdirectories + * [#1883] Add a simple Conference class + * [#1882] Use static audio buffer in Pulse and ALSA layer (instead of + malloc) + * [#811] First commit toward re-integration and refactoring of ZRTP + * [#1882] Flush RTP ring buffer before entering mainloop + * [#1882] Fixed MainBuffer::UnBinCallID() in case there is no + ringbuffer + * [#1882] Test (and fixe) high level conference and mixing + functionalities + * [#1772] Apply patch to compile on fedora (sent by Marcin + Zajączkowski <mszpak@wp.pl>) + * [#1882] Update Bind, unBind call_id in MainBuffer + * [#1959] This adds the ability to store password as an MD5 Hash in + the + * [#1538] Fixes rules compilation + * [#1930][#1931] Fixed a mistake (again) related to index and + credential count + * [#1753] Remove ILBC from pjproject - Hacks in pjsip + * [#1930][#1931] Credential was not selected properly using realm + * [#1882] Finilize multiple reading pointer in RingBuffer + * [#1538] Remove configure from autogen.sh to respect debian upstream + authors policy + * [#1773] Remove generated files from repo + * [#1791] Use XDG_CACHE_HOME to save pid file + * [#1791] Fixes path to save history + * [#1791] Fix debian installation scripts + * [#1930][#1931] Settings are now taken into account in the server. + * [#1882] Add ringbuffer default ring buffer pointer in methods + involving mStart + * [#1882] Add default ringbuffer pointer + * [#1882] Add RingBuffer multiple read pointer basic functionnalities + * [#1882] Fix MainBuffer flushData unit test + * [#1930][#1931] Ability to save and retreive the configuration from + * [#1882] Added Multiple CallID mapping to MainBuffer + * [#1791] Not much + * [#1791] If XDG env variables are not null but empty, use default + ones + * [#1791] Make XDG_CONFIG_HOME writable + * [#1930][#1931] Partial commit. Not working yet. Cannot delete + account + * [#1881] Fixed alsa capture latency problem + * [#1881] Fixed Alsa capture temporarily + * [#1930] [#1931] Partial unbroken commit providing the ability to + * [#1881] MainBuffer implemented in AudioLayer/AudioRTP + * [#1881] Add discard and flush unit-tests + * [#1881] Add discard and flush functionnalites to MainRingBuffer + * [#1881] Add availForGet in MainBuffer + * [#1881] Add availForPut function to MainBuffer + * [#1880] Remove AudioRTP* pointer from SipVoIP (reapered while + merging master) + * [#1881] Add a map between call id and coresponding ring buffer + * [#1855] Refresh pot file and upload on Launchpad + * [#1881] MainBuffe now robust to false ids on getData and putData + * [#1881] Fix big big big memory leak + * [#1881] Add getData and putData to mainBuffer + * [#1881] Unit-test basic ring buffer functionnaities + * [#1881] Add class MainBuffer and basic buffer creation unit-tests + * [#1880] Fix call transfer (step2) issues + * [#1880] Moved AudioRtp* pointer from SIPVoIPLink to SIPCall class + * [#1791] Add postinst script to keep user data when migrating + config/history file + * [#1797] Make pjsip compile + * [#1777] Code indentation + * [#1791] Use XDG_DATA_HOME and XDG_CONFIG_HOME for sflphonedrc and + history + unit tests + * [#1746] Useless space does not appear anymore when volume sliders + and + * [#1643] GtkCheckMenuItem is used instead of icons for elements in + the + * [#1110] [#1668] STUN parameters are now located in the preferences, + under + + -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 06 Nov 2009 11:20:01 -0500 + +sflphone (0.9.6-SYSTEM) SYSTEM; urgency=low + + ** 0.9.6 ** + + * Documentation on echo test + * [redmine_down] codec names not displayed in total + * [redmine_down] crash when hanging up a dialing call because tries to + add it to history whereas no starttime + * [#1927] alternate every time screen changed to call history + * [#1886] clean code + * [#1886] debug messages when loading history removed + * [redmine_down] sflphone-kde icons + * [#1855] Update language files + * [#1502] Update version number + * [redmine_down] setHistory at close + * [#redmine_down] Handle PJ_DECLINE_SC as failure + * [#1923] Fix segmentation fault when adding a new account + * [#1923] Check on iterator before setting the config + * [#1904] Added mnemonic to tabs in sflphone. + * [#1905] The daemon was not sending the currentSelectedCodec signal + on dbus when answering a call. + * [#1922] Default values set to all account details + * [#1886] Spinbox reg expire enables apply, and address book is not + visible when disabled + * [#1905] Bug fix for segmentation fault caused by an empty string, + * [#1910] Warnings in test directory + * [#1919] Error fixed + * [#1855] Update russian translation - Hussein Abdallah + * [#1910] Remove files + * [#1919] fixed + * [#1777] Code indentation + * [#1918] fixed + * [#1917] fixed + * [#1910] Remove warnings compilation in src + * [#1886] removed AccountListModel in configskeleton + * [#1914] + * [#1911] check previous and new port + * [#1910] Remove compilation warnings in src/dbus and src/history + * [#1910] Remove compilation warnings in src/audio + * [1855] Update german translation - Sven Werlen + * [#1909] removed + * [#1906] Done + * [#1904] The registration expire value is now configurable from the + * Cleaned up debug messages. + * [#1886] separated initCallItem in two functions + * [#1886] reversed error in commit + * [#1886] clean debug + * [#1886] changed Name of classes and files + * [#1886] clean + * [#1870] In call_state_cb (dbus.c:126), _time_stop was overridden by + the actual time. + * [#1884] Added some new gpg flags to prevent tty warnings + * [#1886] Clean audio config dialog + * [#1886] No more compile warnings. + 1 comm + * [#1872] Check if the user input is smaller than PJ_MAX_HOSTNAME. + * [#1886] + * [#1785] Fixed build when no new commit + * [#1852] If chosen by the user, the hostname can now be solved and + used + * [#1871] * and # inverted back + * [#1869] Conditional compilation that checks if + * [#1309] removed test in main + * [#1425] Put actions in SFLPhone window class instead of ui view, + made a separate toolbar for screens. + + -- SFLphone Automatic Build System <team@sflphone.org> Mon, 27 Jul 2009 09:53:19 -0400 + +sflphone (0.9.6~rc2-SYSTEM) SYSTEM; urgency=low + + ** 0.9.6~rc2 ** + + * [#1755] Remove generated file + * [#1753] restore ilbc ... + * [#1866] Methods getSipPort and setSipPort now have an effect on the + * [#1753] make pjsip compile without ilbc. Use ./autogen.sh --disable- + ilbc-codec + * [#1855] Fix error in russian translation + * [#1805] Remove the old flawed signal mechanism which was failing in + * [#1855] Refresh translation + * Spanish translation finished + po README files updated + echo's in + copy-in-clients + * [#1850] Yun made the chinese HK-CN translation + * [#1848] Fix transfer interface bug + * [#1862] At install, kde client installs only french translation file + * [#1841] A new fallback mechanism was added to the internal resolver + in PJSIP. + * Started AccountList model/view + * [#1855] Remove po subdir in Makefile.am + * [#1855] Fix typo error in sflphone + * [#1855] Do not generate Makefile in sflphone-common/po + * [#1855] Copy translation files into both clients dirs + * [#1855] Remove po dir from sflphone-common + * Comments added + * [#1860] mailbox->voicemail... + * make scripts executable + * [#1855] French translation + * [#1855] Chinese zh_HK partially filled... + * [#1859] An unnamed pipe monitored by poll() was added. When we want + to + * [#1855] Sven completed the first part of the german translation + * [#1855] Cantonese manually filled for already translated, almost + equal strings + * [#1855] Merge russian translation + * [#1855] Spanish manually filled for already translated, almost equal + strings + * [#1855] Update german translation in ./lang/de + * [#1858] This problem was fixed by removing a useless line in + * [#1855] merged existing translations in lang/ sflphone.po's + * [#1842] [#1843] An attempt at improving the expected behaviour that + can't + * [#1855] added po folder in gnome client and scripts for copying from + common lang folder to clients + * [#1853] Edit before call does nothing on call history + * Put most language entries possible in common. From 300 to 250 + entries. Stays underscores problem. Scripts for copy in clients. + * commit to merge master + * [#1825] Changed "Bad authentification" to "Authentication Failed". + * common po files + * [#1753] Remove ILBC from pjproject + + -- SFLphone Automatic Build System <team@sflphone.org> Fri, 17 Jul 2009 19:12:58 -0400 + +sflphone (0.9.6~rc1-SYSTEM) SYSTEM; urgency=low + + ** 0.9.6~rc1 ** + + * Update some version number + * [#1792] Creates .sflphone directory with permission 600. Also, + "chmod 600" after + * [#1810] GUI is now notified that the call failed. Also, a segfault + was + * [#1816] Address book search disabled when disabled address book and + enabled it back plus button stays triggered + * codeclistmodel + asynchronous loading of address book + + enable/disable address book + * [#1810] Now checking SDP answer after 200 OK. Still need to + implement full + * [#1794] Can't use the interface during a call + * Updated translation files + * Russian translation integrated + * Codec list model/view started. + * [#1807] Add configure.ac in pjproject-1.0.3 + * [#1787] closeRtpSession added in some places where it should have + been + * Use Item class for contacts and accounts + * Comments + clean code + * [#1794] Improved debug messages + * [#1805] Replaced the old and unreliable mecanism that was was + waiting for + * [#1794] Can't use the interface during a call + * [#1787] For those cases where no registered SIP account is + configured + * [#1797] Make pjsip compile + * [#1787] Minor changes. Removed useless commented line. Changed order + of + * [#1777] Code indentation + * [#1797] Update package generation with new pjsip version + * [#1798] Does not hang up when the call is building up + * [#1797] Update .gitignore with new pjsip version + * [#1797] Remove generated files from repo + * [#1797] Main build system now uses pjproject-1.0.3 + * [#1797] Add pjproject-1.0.3 + * [#1797] Remove pjproject-1.0.2 + * [#1796] Computing time optimization (samplerate conversion) + * [#1787] _audiortp->start() moved away from offhold(), + SIPCallAnswered() + * [#1312] Added new states for calls initialized by other clients + * [#1795] Crashes when adding a new account, checking it and applying + * [#1782] Missing icons + * [#1793] KDE client compilation problem + * Fake ringtone files can no longer be set. + * indentation + * [#1312] Able to fetch to differentiate incoming/ringing call state + * [#1784] Use DESTDIR variable in po Makefile - fix language file + installation + * [#1785] Fixed typo + * [#1785] Fixed changelog update + * [#1759] ./autogen.sh --prefix=/usr --with-debug to use optimization + level 0 + * [#1773] Changed snapshot naming convention + * [#1773] Removed gpg agent use, added repository cache cleaning + * [#1759] Use optimization level 0 for repository, 2 for packages + * [#1777] Code indentation/formatting + * Translated new features in french + * [#1785] Added missing changelog entry + * [#1781] Window title is SFLPhone + * [#1777] Add code indentation/formatting in the buil system + * [#1774] Can't set voicemail number in KDE account creation wizard + * [#1775] Can't modify account information for account created with + the wizard + * [#1771] Add a "Default" button in context menu to disable chosen + prior account + * [#1705] + * [#1224] Remove generated file from the repo + * [#1224] Remove generated file from the repo + * [#1762] distclean target should remove kconfig generated files + (settings.h, settings.cpp). Rename them? + * [#1761] clear history button should really clear history + * Dialpad works. + * Implemented Dialpad widget instead of building it in main view. + * Removed last occurence of the old config dialog, that made the build + crash. + * [#1755] Do not consider G722 as a dynamic payload elsewhere than in + RTP layer + * [#1753] Remove ilbc Makefile generation + * [#1756] Implement a kde configuration dialog with kconfig xt and + kconfigdialog class + * [#1755] fix audiocodec folder parsing problem + * [#1450] Reinit timestamp comparison in RTP, create session in + newOutgoingCall + * [#1753] Remove milenage third party code from pjsip + * New Config Dialog integrated in GUI.(without codecs) + * [#1753] Remove ILBC codec + * kconfig started, tr2i18n -> i18n, icons folder, accountList changed + * [#1705] Fixed Audio RTP thread creation/start + * [#1714] Fix codec negociation result handling + * [#1678] Fix audiortp payload setting + * [#1678] Put bac putData method in rtp + * [#1669] gtk_file_chooser_get_filename() support UTF-8 by default + * [#1735] Add conditions to sdp update call if call declined + * [#1737] substr of recordings destination folder to remove "file://" + should be done in client rather than in daemon + * [#1731] Enlarge audio stream buffer size + * [#1714] Missing true + * [#1317] Fixed Mandriva timeout + * [#1317] Changed tag convention + * [#1317] Cleaned git-dch + + -- SFLphone Automatic Build System <team@sflphone.org> Fri, 10 Jul 2009 15:50:26 -0400 + +sflphone (0.9.6~beta-SYSTEM) SYSTEM; urgency=low + + ** 0.9.6~beta ** + + * spec files for mandriva and opensuse updated with buildrequires + libqt4-dev >=4.3 + * [#1700] Cannot build on ubuntu 8.10 and a few other distribs + * [#1502] Update version number where applicable + * [#1642] Update client icons + * [#1450] Clean up useless debug and comments in sipvoiplink and + audiortp + * [#1450] Remove Semaphore object in AudioRtp thread deletion + * [#1450] Audio RTP init now synchronized with Sip/SDP + * [#1693] kde client crashes when changing codecs order/activation + * [#1450] Deep refactoring of audiortp + * [#1450] setRtpSessionRemoteIp + * [#1689] getCallList at start + * [#1224] Change path in package files + * [#1450] Audio RTP initialized only once, payload and remote ip set + at runtime + * [#1450] Add setRtpSessionMedia and setRtpSessionRemoteIp address + * [#1642] Make GNOME GUI fresher and younger ;) + * [#1686] Status bar displaying used account + * added sflphone-kde icon so that it compiles + * [#1659] Ending a call causes the daemon to crash + * corrected introspection XMLs, po files... + * [#1211] g722 media descriptor in codecDescriptor + * [#1310] Install sflphoned in $(prefix)/lib/sflphone + * [#1502] Do not install test binaries and dbus utilitaries + * [#1224] hack for pjsip build system! + * [#1224] Remove pjsip binaries from repo + * [#1224] Upgrade to pjsip 1.0.2 + * [#1658] About SFLphone (bugs) + * [#1658] About SFLphone + * [#1660] Displaying all dialed numbers in a call + * Tested status bar. + * [#790] Optimize pulse audio streams parameters + * [#1678] Some usefull debug messages for mutex/semaphore deadlock + problem + * [#1669] Add/remove some usefull/unusefull debug + * [#1665] Fix latency related to pulse audio stream openning/closing + * [#1457] Make the menus and panels accessible in french + * [#1457] Improve broken keyboard accessibility in menus and conf + panels + * [#961] Instanciate only once the searchbar icons + * [#961] Restore transfer fonction + * [#961] Filter on the history type OK + * [#961] Fix compilation problems on hardy/intrepid + * [#1157] Commit missing files + * [#790] Reduce number of start/stop streams call on pulse audio + * [#1639] kde client crashes when no account registered + * [#1620] Fix the searchbar + * [#1620] Get back caltree as it was during gtkcritical area + * [#1620] Add history filter reinit function + * [#1335] Add a missing label in address book preferences + * [#1561] Update russian translation - Hussein Abdallah + * [#1605] Fix edit menu french translation + * [#961] Enable to search in the history according to the call type + * [#1449] Searchbar does not work anymore + * [#961] Add popup menu on the entry primary icon for history + * [#1317] Fixed KDE client package dependency + * [#936] speex 32 khz integration completed + * [#936] Use 320 frame size + * [#936] Test using a frame size at 320 smpls + * [#1214] Enable / Disable history + * [#1607] Fix compilation problem for ubuntu 8.10 (libsexy) + * [#1313] Implement processDataEncode processDataDecode in audiortp + * [#1613] codec list order can't be set + * Better handling of localisation + added languages + corrected + warnings + begginning of new config dialog with kconfig + 14px + account leds + * [#1214] Save and load history according to the limit timestamp + + unit tests + * [1609] Fix call number copy/paste feature + * [1607] Restore clear action icon in searchbar + * [#936] Try to decode using 1280 samples + * [#936] Add some debug + * [#936] Add .cpp file + * [#936] Oops Forgot speex 32 khz + * [#1214] Add configuration panel for history + D-Bus calls + * [#1313] Test rtp thread function, frame size, nbbytes, resampling + * [#790] Flush audio data before closing audio streams + * [#1214] History displays local time + * [#1214] Skip empty field on display + * [#1214] Associate an account to an history entry + * [#1342] Get addressbook options sensitive/non-sensitive + * [#1211] Clean up and comments + * [#1211] Get back to 20 ms framesize + * [#1211] Use sendImmediate instead of putData in RTP + * [#1211] Fix nb byte available in RTP + * [#1211] Clear condition on maxNbSamples in RTP + * [#1211] Fix max byte available in RTP session + * [#1211] G722: Use 160 samples per frame instead of 320 + * [#1211] Test using a dynamic payload + * [#1211] Test using a dynamic payload type + * [#1211] Rename size variable (nb_samples, nb_bytes) + * [#1211] Test g722 ip-to-ip sending twice the data lenth + * [#1211] Test g722 ip-to-ip + * [#1214] Do not select an history item by default at startup + * [#1214] Remove some compilation warnings + * [#1214] Handle empty field - remove g_print + * [#1214] Add each history item only once + * [#1214] Handle call timestamps properlier + * [#1214] Do not need timestamp files anymore + * [#1214] Use the saved date for history entry + * Clean up + * [#1214] Client doesn't crash if the D-Bus call fails + * [#1214] Client is able to save its history - still some glitches + * [#1211] Forgot 16000 for g722 + * [#1211] G722 initialization + * [#1214] Save name/number, successfully load the history if no fields + are empty + * [#1499] Fixed destination directory bug + * [#1214] Restore all the functionalities; peer name/number way more + easy to handle !! + * [#1214] Add callable_object instead of call_t, refactoring + * [#1211] Test with polycom soundstation 16000 + * [#1211] Remove C like inline function in g722 codec + * [#1342] Finalize gnome client preference window formating + * [#1214] Retrieve the history when the gnome client startsup + * [#1306] Implement localization for KDE client + * [#1593] enable accounts apply button when account checked/unchecked + * [#1214] Implement the dbus calls on server side + * [#1214] Add serialized/unserialized functions to pass data on DBUS + * [#1342] Formating gnome client configuration windows + * [#1214] Save sucessfully a map of history items + * [#1499] Removed multiple jobs compilation for KDE client (2) + * [#1214] Load history from file into memory, add unit tests + * [#1534] Throws a length_error exception in case URL exceeds + std::string max_size + * [#1499] Removed multiple jobs compilation for KDE client + * [#1565] make account leds smaller + * [1430] Fix dbus debug + * [#1562] crashes when trying to change item of a call of state "OVER" + * [#1116] Fix compilation bug + * [#1317] Added mandriva and opensuse-11 64 bits + * [#1108] Add messges in main window concerning transfer success + failure + * [#1116] Fix compilation problems + * [#1211] g722 Makefile + * [#1108] Client side transferFailed/trasferSucceded signals handling + * [#1211] G722 mostly completed, + * [#1555] make bigger toolbar (24x24) + * [#1551] remove default mailbox number in wizard and disable mailbox + button when first account doesn't have mailbox number + * [#1342] Re-add sflphone manpages + * [#1116] Fix compilation on non-jaunty distros + * [#1317] Fixed opensuse startup sleep + * [#1108] Add a signal in the client to notify successful or failed + transfer + * [#1108] Dbus signals concerning call transfer success/failure + * [#1317] Added opensuse to automatic build system + * [#1223] Fix manpages bug + * [#1060] german translation glitch + * Clean up some gnome client warnings + * [#1547] replace ugly account leds by beautiful icons + * [#1548] add close button that hides windowand just hide on clicking + the cross + * [#1549] put introspec XMLs in the client's source + * [#1312] Implement getCallList D-BUS method + * [#1116] Clear text in history and contacts + * [#1499] KDE integration + * [#1469] Modify header linkers in dbus-c++'s Makefile.am's + * [#1469] Remove examples folder from dbus-c++ + * [#1214] History integration in build system; unit test squeleton + * [#1317] Cleaning + * [#1469] Remove configure stuff in dbus-c++ + * [#1469] Add unofficial mainline dbus-c++ + * [#1469] Remove dbus-c++ from freedesktop + * [#1430] Bring account changed signal/callback back to normal + * [#1060] Update german translation - Sven Werlen + * [#1430] Add marshaller one string define + * [#1430] Send account change signal broadcast using account id + * [#1430] Remove condition on setRegistrationState, cause stun to + crash + * [#1317] Centralized version handling + * [#1317] Fixed version number on sfl-git-dch + * [#1317] Refactoring for new distributions + * [#1215] Fix account order at startup if latency + * [#1088] Restore sip dns srv + * [#1214] Add squeleton for history manager + * [#1430] Add accout id to accout changed method + * [#1430] No connectionStatusNotification (account changed) if no + changes + * [#1538] Add COPYING file + * [#1430] Add audio rtp thread tests + * [#1317] Changed version detection + * [#1538] Document license in libs/stund + * [#1317] Added version files + * [#1538] Apply François patches - debian packages + * [#1317] Updated spec files + * add files + * [#1538] Apply François patches - debian packages + * [#1535] Change program file structure (directory src...) + * [#1317] Updated build system scripts + * [#1317] Cleaning + * [#1317] Copied introspect files to gnome client + * [#1317] Added opensuse to build-system : first-shot + * [#1317] Remove spec files from configure + * [#1317] Added missing prefix + * removed debug for daemon account fix + * [#1430] Add a connection reference which most likely belong to + libdbus + * [#1430] Use shared connection instead of private + * make daemon find the account, added userMatch + * Clean code, add comments... + * [#1317] Fixed packaging rules + * [#1317] Updated autogen + * Updated autogen.sh for pjsip + * [#1526] Set accounts order + * [#1317] Fixed pjsip lib dirs + * [#1317] Updated debian packaging for new pjsip configuration script + * [#1317] Switch to autogenerated guess and sub files + * [#1317] Updated pjsip inclusion in build system + * [#1317] Replaced pjsip guess and sub files + * [#1317] Fixed compilation issues on opensuse 11 + * [#1505] account list seem to crash the application when clicking + Apply very fast... + * [#1456] Add a flag to be replaced in the control files + * [#1456] Added version dependancy handling + * put account alias in AccountWidgetItem rather than in the item with + " " before. + * [#1034] The KDE client should start sflphoned if it is not started + * [#1500] Handle options for notifications and display on incoming + call. + * [#1443] Client should not crash when receive an unexpected + stateChanged signal + * [#1403] Do not stop the notification anymore + * [#1456] Added version dependancy handling + * [#1426] Daemon crashes when get alsa plugin + * [#1422] Improved error messages + * commit for merge + * [#1424] Change logo in tray icon and put a different one when + incoming call + * [#1425] first part done, window title... + * [#1413] add manpages creating and installing in build system + * [#1417] The client should start the account creation wizard if + started for the first time (if config file doesn't exist) + * [#1421] Make volume bars horizontal when dialpad is hidden. + * Changed main window title and fixed a mistake in sflphone_const.h + * [#1412] make debian package building work + * changelog changed. + * Changed addAccount method in gnome client. + * Debian and man folders added. + * [#1388] Change project name from sflphone_kde to sflphone-client-kde + * Better handle of kabc check. + * [#1351] Automatic generation of dbus interfaces in makefile + generated by cmake + * [#1307] Implement "edit before call" in history and address book. + * [#1344] change action_call label in call history from "call" to + "call back". + * [#1308] Implement Hook feature in kde client + * Improved build system. + * #1219 : Add address book configuration page + * Better handling of registration to the daemon. + * #1039 : Add tray icon in kde. + * Issue no 1216 : Double click on item in history or address book + causes call. + * display peer name in call list and call history when called from + address book. + * Address book functionnal with photo displayed. + * Help menu kde available but actions disappeared. All fonctions in + view. + * Address book functionnal but ugly and making its own sort in the + complete address book. + * Account choice on right click, clean out includes, page address + book, fixed bugs... + * Wizard, double click, context menu... + * Removed sflphone_kde.kdevelop.filelist + * Added account creation wizard and translated interface in english. + * Transfer functionnal but ugly. + * transfer not functionnal + * Bug fixed : unholding (UNHOLD_CURRENT, UNHOLD_RECORD) + * Commit functional for push. With install.sh + * Before merge. + * Problem with enable accounts. Account display increased. + * Functional with codec order working , playDTMF. + * Commit functional. + * sflphone_kde/build added in .gitignore. + * complete commit for checkout previous. + * Commit before checkout previous version to check the display + bug(little font everywhere...) + * Functionnal client. Rest : history icons, config icons and + functionalities + * commit before merge asavard for isRecording. + * Call and Automate fusion done and seems to work. + * Commiting before putting Automate class in Call class. + * Functionnal main window without recording, history, voicemail, kio + widgets. + * client kde avec kdevelop. + * Config Dialog almost finished. + * Base of QT client + + -- SFLphone Automatic Build System <team@sflphone.org> Tue, 23 Jun 2009 11:13:42 -0400 + +sflphone (0.9.5-SYSTEM) SYSTEM; urgency=low + + ** 0.9.5 release ** + + * [#1060] FIx bug in chinese translation + * [#1313] git add rtpTest.cpp rtpTest.h + * [#1313] Add init/close rtp tests + * [#1313] Basic instanciation of the rtp layer + * [#1449] Gtk-Critical concerning history filters and new calls + * [#1400] Make the match with the hostname instead of username + * [#1324] Change status bar label for "Using %s (%s)" + * [#1403] Icon size: 60x60 px + * [#1403] Do not remove notification, improve icon quality + * [#1403] Add smaller icon for gnome notifications + * [#1403] Prevent crash when hangup && no notification + * [#1403] Remove all actions on notifications; code refactoring + * [#1451] Use stun.sflphone.org as default STUN server + * [#1060] New po files - need to be translated + * [#1060] Update french translation - Rebuild template file + * [#1456] Add a flag to be replaced in the control files + * [#1454] Make cppunit optional; remove from build deps in control + files + * [#1401] Add libexpat1-dev dependency in control files + * [#1448] Take off these ugly debug messages + * [#1448] fixed getTelephoneTone and getTelephoneFile() called + repeatedly + * [#1406] add liblog4c-dev in build-depends + * [#1409] Restore .desktop icon + + -- SFLphone Automatic Build System <team@sflphone.org> Mon, 25 May 2009 11:34:48 -0400 + +sflphone (0.9.5-SYSTEM~rc2) SYSTEM; urgency=low + + ** 0.9.5 rc2 ** + + * [#1422] Improved error message + * [#1402] Fix pjsip build + * [#1404] Clear GTK-Critical Bug at client startup + * [#1422] Added automatic VM shutdown when building on more than one + VM + * [#1422] Fixed some issues with new changelog generation script + * [#1422] Moved distribution update to specific file + * [#1422] Dropped git-dch, replace by home made implementation + * [#1402] Fix pjsip build + * [#1404] Clear GTK-Critical Bug at client startup + * Changes for name based dbus connection + * Clean changelogs + * [#1343] Gnome: Implement a callback system to handle focus on + different widgets + * Debus Session + * Refactoring Python code, PEP8 + * [#1430] Get back dbus_g_proxy_new_for_name + * [#1430] Get back DBUS_BUS_SESSION type + * [#1430] Dbus fixed owner message binding + * Second test with DBUS owner + * [#1404] Gnome -> Preferences -> Hooks + * [#1404] Gnome -> Preferences -> Recordings + * [#1404] Call History + * [#1404] Gnome -> Preferences -> Address Book + * [#1404] IF the first notification option disable the second + notification + * Dbus with fixed owner does not automatically start the deamon + * Add codec debug tests in pysflphone + * [#1407] Some print info + * [#1407] Add a scenario to pick_up action + * Test client dbus connection to a fixed owner + * Add python dbus test suite + * [#1161] Modified version handling in build system + * [#1314] Test pulse audio and audio streams connect and disconnect + * [#1402] Add info message after configure + * [#1402] Build the daemon with the local pjsip library (vs the + installed one) + * [#1009] Fix Codec Sampling Rate set to zeros + * [#1314] Add mutex to pulse layer audio streams + * [#1314] Refactoring pulseaudio stream to test connect disconnect + * [#1314] Refactoring of pulselayer to test conect/disconnect + * Add debug messages in debus calls concerning account + * [#1314] Add some return values to audio init functions + * [#1406] add liblog4c-dev in build-depends + * [#1409] Restore .desktop icon + * Bug #1405: Fix strings as requested. + * Bug #1404: Fix strings in preferences panel. + + -- SFLphone Automatic Build System <team@sflphone.org> Tue, 19 May 2009 12:08:18 -0400 + +sflphone (0.9.5-0ubuntu1~rc1) SYSTEM; urgency=low + + [ SFLphone Project ] + * [#1262] Updated changelogs for version 0.9.5-0ubuntu1 Snapshot 2009- + 05-05 + + [ Emmanuel Milou ] + * Add some python CLI client code; not really functional + * [#1108] Fix peerHungup method for IP to IP call + + [ Alexandre Savard ] + * [#1108] Correct setting of SIP contact for direct IP call + * [#1108] SIP user agent handles incoming REFER + + [ Emmanuel Milou ] + * Remove website from repository + * Update translation + + [ Alexandre Savard ] + * Sflphone icon's tooltip changed for "configured" instead of + "registered" + + [ Emmanuel Milou ] + * Update translation + + [ Sflphone Project ] + + -- Sflphone Project <sflphone@mtl.savoirfairelinux.net> Tue, 05 May 2009 19:16:13 -0400 + +sflphone (0.9.5-0ubuntu1~beta) SYSTEM; urgency=low + + [ Julien Bonjean ] + * Updated Eclipse stuff + * Improved addressbook config window + * Added sflphone Eclipse stuff + * Implemented addressbook list server side + * Moved dbus stuff in dbus directory + * Updated addressbook configuration + + [ Emmanuel Milou ] + * Remove unuseful installation scripts. Use apt-get build-dep sflphone + instead + * fix bug #1090 + + [ Alexandre Savard ] + * defining speex 16khz + + [ Emmanuel Milou ] + * Remove unuseful file from build system + * Start dns srv resolver + + [ Alexandre Savard ] + * Basic ogg/vorbis initialization + + [ Emmanuel Milou ] + * Handle incoming IP-to-IP invite correctly + + [ Alexandre Savard ] + * speex wideband 16000 + + [ Emmanuel Milou ] + * Better handling of incoming IP to IP call + * DNS SRV resolution functional + * Implement IAX2 incoming URL + * Allow user to make IP call without any accounts configured + * Add a contextual menu to edit a number from the contacts tab + * Add comments, tooltip and new button to the contextual menu + * add delete event, migrate to GTK 2.16 for sexy icons + * Resolve ticket #1118 + * Update suse spec file + * Add phone number cleanup functions, unit tests and panel + configuration + * Add pertinent test that fails + * fix dependencies for suse package + * Add contextual edit menu in history - #1120 + + [ Alexandre Savard ] + * Temporary comit: make speex wideband (16 khz) + * Temporary: shared object for speex narrow band + * Temporary: speex narrowband and wideband coexist + + [ Julien Bonjean ] + * Fixed bug when no book selected + * Fixed addressbook related compilation warnings + * Fixed GTK client remaining compilation warnings + * Fixed segfault when book removed since last sflphone run + * Fixed bug when book is unreachable (ldap error) + + [ Alexandre Savard ] + * Fix codec list in audio config window + * Active/inactive speex codec by payload + + [ Julien Bonjean ] + * Updated gitignore + * Added some comments + + [ Emmanuel Milou ] + * Add callto: handler script for browsers and al. + * Integrate test compilation in the daemon build-system + + [ Julien Bonjean ] + * Fixed g_object_unref warning for pixbuf + * Cleaned too verbose output + * Fixed toolbar update warning + * Added support for asynchornous books open (first shot) + + [ Emmanuel Milou ] + * Add a DBus call to fetch the call details from a call ID - Ticket + #928 + + [ Julien Bonjean ] + * Improved async open books + * Fixed bug #1139 + + [ Emmanuel Milou ] + * Add a way to save account order + * commit missing files + + [ Julien Bonjean ] + * Introduced log4c (ticket #1162) + + [ Emmanuel Milou ] + * Load/save account order functionnal - ticket #813 + + [ Alexandre Savard ] + * Add CELT codec (#1143) + * Make celt frame size 256 (*1143) + + [ Julien Bonjean ] + * Switched everything to log4c (ticket #1162) + * Updated eclipse settings + + [ Emmanuel Milou ] + * Restore adding account - ticket #1172 + * Add liblog4c dependecy - ticket #1179 + + [ Alexandre Savard ] + * Double maxAvailByte for frame size in rtp (#1143) + + [ Emmanuel Milou ] + * Add User-Agent SIP header - Ticket #1173 + + [ Julien Bonjean ] + * Fixed autoresize issue (#708) + + [ Emmanuel Milou ] + * Remove libcppuint dependency for the debian packages + * Look for libsexy only if gtk version < 2.16 - Ticket #1116 + * Remove libsexy dependency for jaunty. ticket #1116 + + [ Julien Bonjean ] + * Introduced unit tests (#1146) + * Updated gitignore + * Fixed Makefile (#1146) + + [ Emmanuel Milou ] + * [TICKET #1112] Add a test on the voice buffer to send through iax + packets + * Remove doublon in dependencies + * Remove warnings from the client test framework + * Update version number to 0.9.5~beta + * Update build-package script + * Add check dependency in build-deps control file field + * Create debian files for the new sflphone + * [TICKET #1212] Add Replaces field in control files + * [TICKET #1212] Fix manpages installation path + * [TICKET #1212] Add maintainer scripts to create alternatives + * [#1212] Update the manpages generation - edit preinst maintainer + script + * [#1212] Fix reference error in manpage + * [#1212] Add missing files on the client side + * [#1212] Fix debian docs files - no TODO file + * [1212] Fix manpage creation problem + * [#1220] Generate client-side glue files and marshaller at + compilation time + * [#1220] Generate server-side glue files at compilation time + * [#1212] Change binary name to sflphone + * [#1212] Update .gitignore to fit the new working tree + * [#1220] Explicitly generate glue files before building the library + * [#1220] Compile dbus directory before audio + * [#1212] Create sflphone-common at the root of the repository + * [#1212] Re-add pjproject + * [#1212] Remove Makefile from repo + * [#1220] Fix Makefile.am + * [#1212] New working directory functional + * [#1212] Update .gitignore + * [#1212] Hack to make pjsip compile.. + * [#1220] Use non-installed binary for dbusxx-xml2cpp + * [#1212] Add descriptive files, remove unuseful scripts from tools/ + + [ Alexandre Savard ] + * Restore speex codecs + * add frame size for celt (#1143) + * add framesize to codec, independant from audiolayer (#1143) + * use codec frame size in rtp (#1143) + * compute fixed_codec_framesize (#1143) + * do not resample if not required (#1143) + * add condition on resampling for decoder (#1143) + * add a condition on bytesAvail == 0 from mic data + * no maximum in rtp decode (#1143) + * compute maximum for decoding (#1143) + + [ Emmanuel Milou ] + * [#1146] Implement unitary tests on the client-side + + [ Alexandre Savard ] + * use float instead of int to compute max nb of sample (#1143) + * add nbSampleMax for unresampled data (#1143) + * make thread sleep during 5 ms insead of 20 (#1143) + * use unix usleep (#1143) + * 50 usecond thread!!!!! (#1143) + * try with the smallest compression (#1143) + * use timer set at framesize (#1143) + + [ Emmanuel Milou ] + * [#1161] Restore changelog version + + [ Alexandre Savard ] + * Remove celt stuff + + [ Emmanuel Milou ] + * [#1161] Update changelog + * [#1220] Add Conflicts: sflphone in debian control files + * [#1179] Add liblog4c3 runtime dependency + * [#1212] FIx typo error in dependency list for itnrepid + * [#1212] FIx .desktop file to point on the right exec + * [#1212] Modify changelog replacing tag + + [ Sflphone Project ] + * "[#1262] Updated changelogs for version 0.9.5-0ubuntu1~beta" + + [ Emmanuel Milou ] + * [#1212] restore changelogs + + [ Sflphone Project ] + * [#1262] Updated changelogs for version 0.9.5-0ubuntu1 Snapshot 2009- + 04-27 + + [ Emmanuel Milou ] + * [#1212] restore changelogs + + [ Sflphone Project ] + * [#1262] Updated changelogs for version 0.9.5-0ubuntu1~beta + + [ Emmanuel Milou ] + * [#1212] restore changelogs + + [ Sflphone Project ] + + -- Sflphone Project <sflphone@mtl.savoirfairelinux.net> Mon, 27 Apr 2009 17:00:03 -0400 + +sflphone (0.9.4-0ubuntu2) SYSTEM; urgency=low + + [ Alexandre Savard ] + * Restore speex and GSM detection + + [ Emmanuel Milou ] + * Fix bug #1090 + + -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 8 Apr 2009 11:29:15 -0500 + +sflphone (0.9.4-0ubuntu1) SYSTEM; urgency=low + + [ Emmanuel Milou ] + * Integrate DBus-c++ and libiax2 in the main build system + * Clean up in the working repository + * Reorder hooks configuration panel + * Protect case when no codecs are active + * Fix some return values + * Add unitary tests for the hook manager (premisces) + + [Yun Liu] + * Update chinese translation + + [Sven Werlen] + * Update german translation + + [Hussein Abdallah] + * Update russian translation + + [Maxime Chambreuil] + * Update spanish translation + + -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 3 Apr 2009 18:29:15 -0500 + + +sflphone (0.9.4-rc1) SYSTEM; urgency=low + + [ Emmanuel Milou ] + * Fix bug while trying to hold/unhold several simultaneous call + * Improve address book build system + * Implement SIP url popup on incoming call + * Improve GTK+ panel configuration + [ Julien Bonjean ] + * GTK+ client refactoring + * GTK+ clean up + * Address book improvment + + -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 27 Mar 2009 18:29:15 -0500 + +sflphone (0.9.4-0beta1) SYSTEM; urgency=low + + [ Alexandre Savard ] + * Display codec used during conversation on the GUI + * Enable/disable STUN parameters at runtime + * Refactor search bar use + [ Emmanuel Milou ] + * Build system fixes + * Implement SIP re-invite + * Implement IP to IP call + [ Julien Bonjean ] + * Integrate GNOME address book based on evolution data server + + -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 20 Mar 2009 18:29:15 -0500 + + +sflphone (0.9.3-0ubuntu3) SYSTEM; urgency=low + + [ Alexandre Savard ] + * Both playback and record streams in PA_STREAM_CORKED (pulseaudio) + * Use PLUGHW device for ALSA capture + * Functional IAX and SIP recording for voicemail + * Use the less CPU-consuming interpolator algorithm for resampling + * Display in GTK GUI the codec used in conversation + * GTK GUI use ASCII instread of utf-8 + * Add record menus in GTK GUI + * Put on hold when dialing a new number + * AccountID's are saved in the history + + [ Emmanuel Milou ] + * Integrate DBUS C++, libiax2 in the git repository + * Update website + * Use libspeexdsp only if available on the system + * Updated .gitignore file + + [Cyrille Béraud] + * Account assistant manager improvment + * Add an email request when creating a new account to receive voicemails + + -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Sat, 14 Feb 2009 13:29:15 -0500 + +sflphone (0.9.3-0ubuntu2) SYSTEM; urgency=low + + [ Emmanuel Milou ] + * Add compilation note in README + * Use default ALSA plugin for capture + * Fix the ALSA capture problem one more time + * Clean up debug messages in dbus.c + * Add libspeexdsp dependency + * Remove implicit declaration compilation warnings + * Fix links in the website, add release note + * Change capture for the website front page + * Add alsa devel dependency in build-depends control file field + * Clean up, indentation, try to handle latency problems in iax/pulseaudio + * Remove pjsip generated files from the repo + * Use the previous declared curAlias function in accountwindow + * Fix bug in history call duration when the call fails + * Remove runtime warning in the GTK+ client + * Add librsvg2-common dependency to load SVG under KDE + * Refresh .gitignore + * Update locales files + french translation + * Add configuration panel for future noise reduction + * Add configuration panel for audio record module + * Daemon less verbose; accounts don't try to access STUn options anymore + * Fix typo in configwindow + * Add content in the official website + * use a GTK_STOCK icon for the record button + * Complete description text in the assistant manager + * Add libtool flags in client configure.ac + * Remove unuseful dependency (snd) + * Fix SIP transfer problems + * Remove previous version of PJSIP from the repo + * Upgrade PJSIP to version 1.0.1 + * Add the new website source in the repository + * Use libspeexdsp for silence detection only if available + + [ Loïc Faure-Lacroix ] + * Ajout du logo gpl3 + * Ajout des images + * Ajout de la section screenshot pour le site + * Ajout du favicon dans le header + * Modification des cartes + + [ Alexandre Savard ] + * Clean up <speex/libspeexdsp> + * Small cleanup + * Save Wave fixed + * Fix new call button when recording + * libspeexdsp added + * Recording: default home folder at startup + * Minor changes to config window + * IAX recording fixed + * Set / get recording path, still need some GTK for client + * AudioRecord file name format + * Now recording in HOME folder + + [ Cyrille Béraud ] + * Fix bug in reqaccount.c + + [ Maxime Chambreuil ] + * Update spanish translation + + [Yun Liu ] + * Update chinese translation + + [ Hussein Abdallah ] + * Update russian translation + + -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Sat, 14 Feb 2009 13:29:15 -0500 + +sflphone (0.9.3-0ubuntu1) SYSTEM; urgency=low + + * Remove debug + * Join thread before leaving + * Fix implicit declaration in reqaccount + * Add REST code to build the request to server + * Fix GValue initialization warnings + * Update version number, fix implicit declaration, fix GTK markup + warnings + * Apply patch to create custom SIP account from our own server + + -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 06 Feb 2009 19:17:32 -0500 + +sflphone (0.9.2-2ubuntu9) SYSTEM; urgency=low + + [ Alexandre Savard ] + * Speex audio codec preprocessing initialization + * peer hung up segmentation fault solved + * Stop recording when transfering + * Terminate only one call + * Add isRecording() function + * Fix call_icon GTK client + * Fix SIPCallClose() function, recorded file now close properly + * Function terminateSIPCall added in sipvoiplink and managerimpl + * Fix thread destructor + * setRecordingOption function implement in audiorecord + * Record now implemented in Call class + * Record interface complete (on hold erase previous recording) + * Added recButton in client + * Added: record button related icons + * Record button added + * Overload AudioRecord::recData to get mic and speaker data mixed + * Recording now in audiortp::run() method + * Audio recording working in AudioRTP: receiveSessionForSpeaker + * Open/close a wave file when pulse audio stream start/stop + + [ Emmanuel Milou ] + * Fix path for GTK+ icons; clean up + + -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Thu, 05 Feb 2009 18:27:53 -0500 + +sflphone (0.9.2-2ubuntu8) SYSTEM; urgency=low + + [ Emmanuel Milou ] + * Update changelogs + * Fix bug in merge and in Makefile.am + * Terminate only one call + * Disable PJsip shutdown when changing STUN parameters + * Function terminateSIPCall added in sipvoiplink and managerimpl + * Add a timer to the alsa thread to not jam the CPU load + * Fix bug in sipvoiplink.cpp + * Clean shutdown of pulseaudio on quiting + * Fix DTMF at first start with Pulseaudio + * Remove zeroconf from the build system + * Add a library manager + exception handling + * Clean up in the working directory + * Better handling of capture XRUNs + * Restore mic adjust volume on ALSA layer + * Protect device ALSA operation if not opened + * Fix the switching layer bug + * Use dynamic_cast<> to use audiolayer-specific methods + * Open the audio devices only once at startup + * Refactoring of the ALSA part + * Functional plug-in manager + * Use a C++ thread to handle tones and DTMF in ALSA + * Restore IAXVoIPLink, restore Mutex + * Make the plugins registering against the plugin manager + * Migrate to 1->N relationship between voiplink and accounts + * API plugin for registration + * Use C++ thread in SIP, move everything in sipvoiplink + * Complete singleton pattern for the plugin manager + * Add -Wno-return-type compilation flag to remove warnings; Update + version number in configure.ac + * Add the dynamic loading for the plugin framework; integate unittest + + [ Yun Liu ] + * Update rpm spec file + * modify build package script and spec file for suse + + [ Alexandre Savard ] + * Add audiorecorder plugin and testaudiorecorder + * Add audio Recording class, edit global.h + + -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 04 Feb 2009 14:00:30 -0500 + +sflphone (0.9.2-2ubuntu7) SYSTEM; urgency=low + + [ Emmanuel Milou ] + * Update changelog to 0.9.2-6 + * Fix some dbus-glib implementation details on the client side + * Init history after dbus initialization + * Add error checking in useragent; Clean sipvoiplink + * Prevent crash when trying to call an empty number + * Set the volume of the playback stream to PA_VOLUME_NORM at startup + * Fix GTK+ generic value double initialization + * Fix jaunty control file dependency problems + * Fix jaunty control file dependency problems + + [ Yun Liu ] + * Fix bug ticket # 137 + * Tolerant to gsm library of OpenSuse 11 + + [ Sven Werlen ] + * Update german translation + + -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 23 Jan 2009 17:48:13 -0500 + +sflphone (0.9.2-2ubuntu6) SYSTEM; urgency=low + + [ Emmanuel Milou ] + * Migrate STUN configuration to the main config window + * Update french translation + * Other tiny memory leaks + * Fix memory leak in sampleconverter.cpp + * Generate packages from the release branch + * update the build package script + * modify the control files with architecture=any + * Remove valgring uninitialized value + * IAX and SIP use the same global variables to set account + configuration ; fix broken code + + [ Maxime Chambreuil ] + * Update spanish translation + + [ Hussein Abdallah ] + * Update russian translation + + [ Yun Liu ] + * Update translation files + * Fix the bug when user uncheck the account which fails in the + previous registration + * Add stun error status + * Fix bug ticket #143 + * Script for auto-install dependencies + * Fix bug ticket #140 + * Fix bug ticket 141 + * Fix the reregister process when user change the details of an + account + + -- Emmanuel Milou <manu@sulfur.inside.savoirfairelinux.net> Fri, 16 Jan 2009 18:19:05 -0500 + +sflphone (0.9.2-2ubuntu5) SYSTEM; urgency=low + + * Fix memory leak in the pulseaudio callback + * Update debian package generation script + * Warnings removal in GTK+ client + * Clean adjust volume method in alsalayer + * Plug the sflphone playback volume control to the pulseaudio volume + manager + * Display the date in history according to the current locale + * Generate the changelog according to the git commit messages + * Complete header in chinese translation file + * Use the right gpg key to sign the packages + * add debian jaunty jackalope support + + -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 14 Jan 2009 21:17:20 -0500 + +sflphone (0.9.2-2ubuntu4) SYSTEM; urgency=low + + [ Emmanuel Milou ] + * add german translation + + [ Yun Liu ] + * Fix GUI crash in Ubuntu8.10 64bit system + + -- Yun Liu <yun.liu@savoirfairelinux.com> Thu, 08 Jan 2009 13:08:51 -0500 + +sflphone (0.9.2-2ubuntu3) SYSTEM; urgency=low + + [ Emmanuel Milou ] + * The main thread synchronizes the ringtone thread + * disable custom ringtone for the ALSA layer + * Fix the Makefile.am in man directory, add a SEE ALSO section + + [ Yun Liu ] + * Fix daemon crash caused by the previous patch ( for bug ticket #129) + + -- Yun Liu <yun.liu@savoirfairelinux.com> Tue, 06 Jan 2009 16:18:38 -0500 + +sflphone (0.9.2-2ubuntu2) SYSTEM; urgency=low + + * Fix bug ticket #129 + + -- Yun Liu <yun.liu@savoirfairelinux.com> Wed, 5 Jan 2009 15:54:53 -0500 + +sflphone (0.9.2-2ubuntu1) SYSTEM; urgency=low + + * Migrate from eXosip library to pjsip + * Add multiple SIP accounts support + * Fix ringtones problems + * Add a pulseaudio support + * Improve audio quality with ALSA + * Add chinese translation + * Improve spanish translation + * Migrate to a maintained C++ DBus bindings + * Clean and improve the build system + * Add build-dependency on Perl because we need pod2man to generate manpages + + -- Yun Liu <yun.liu@savoirfairelinux.com> Wed, 26 Nov 2008 09:47:53 -0500 + +sflphone (0.9.1) unstable; urgency=low + * Add a search tool in the history + * Migrate some gtk_entry_new to sexy_icon_entry_new + * Bug fix (Ticket #78): The voicemail password isn't displayed anymore in + the history tab + * Add the SIP registration expire value in the user file. + + -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Thu, 22 May 2008 11:14:25 -0500 + +sflphone (0.9.0) unstable; urgency=low + * Add history features + * Call date + * Call duration + * Mouse events in the history tab + * Smooth switch from the history tab to the calls tab + * Remove most of GTK-Critical warnings + + -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 13 May 2008 16:58:25 -0500 + +sflphone (0.9-2008-06-06) unstable; urgency=low + * Audio bug correction: capture stopped after a few minutes of conversation + with USB Plantronics sound card + + -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Tue, 06 May 2008 16:58:25 -0500 + +sflphone (0.9-2008-05-06) unstable; urgency=low + * Bug correction: account creation with the assistant + * GTK+ warnings removal + * libnotify warnings removal + * Remove aliasing on the SFLphone logo + + -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Mon, 05 May 2008 16:58:25 -0500 + +sflphone (0.9) unstable; urgency=low + * Clean dependencies ( removal of libboost ) + * Several GTK improvement and updates + -account window + -configuration window + * Migrate from GtkCheckMenuItem to GtkImageMenuItem + * ALSA standard I/O transfers: MMAP instead of R/W + * Fix speex audio quality + * IAX2 protocol + -Fix hold/unhold situation + -Add on hold music + * SIP protocol + -Ringtone on incoming call + -Fix transfer situation + * Add desktop notification ( libnotify ) + * Improve the system tray icon behaviour + * Improve registration error handling + * Register/unregister from the account window takes effect without starting back SFLphone + * Compilation warnings removal + * Call history + * Add an account configuration wizard + + -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 30 Apr 2008 16:58:25 -0500 + +sflphone (0.8.2) unstable; urgency=low + * Internationalization of the GTK GUI + * English / French + * STUN support + * Slight modifications of the graphical interface ( tooltips, dialpad, ...) + + -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 21 Mar 2008 11:37:53 -0500