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Commit 4f672be0 authored by Rafaël Carré's avatar Rafaël Carré
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Update debian changelogs

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sflphone-client-gnome (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-client-gnome
* #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-client-gnome.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-client-gnome (0.9.7~rc1~ppa1~SYSTEM) SYSTEM; urgency=low sflphone-client-gnome (0.9.7~rc1~ppa1~SYSTEM) SYSTEM; urgency=low
** 0.9.7~rc1~ppa1~SYSTEM ** ** 0.9.7~rc1~ppa1~SYSTEM **
......
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