From 9b1301b931915d4636fc4ad64866f30355a6b856 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Rafa=C3=ABl=20Carr=C3=A9?=
<rafael.carre@savoirfairelinux.com>
Date: Fri, 30 Sep 2011 14:24:12 -0400
Subject: [PATCH] update client kde changelog
---
.../sflphone-client-kde/debian/changelog | 1849 ++++++++++++++++-
1 file changed, 1835 insertions(+), 14 deletions(-)
diff --git a/tools/build-system/launchpad/sflphone-client-kde/debian/changelog b/tools/build-system/launchpad/sflphone-client-kde/debian/changelog
index d59275102a..a368c43102 100644
--- a/tools/build-system/launchpad/sflphone-client-kde/debian/changelog
+++ b/tools/build-system/launchpad/sflphone-client-kde/debian/changelog
@@ -1,4 +1,1825 @@
-sflphone-client-kde (0.9.6-SYSVER) SYSTEM; urgency=low
+sflphone-client-kde (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 13:51:04 -0400
+
+sflphone-client-kde (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 14:00:02 -0500
+
+sflphone-client-kde (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-client-gnome/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-client-gnome.
+ * [#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
+ ZajÄ…czkowski <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 11:23:15 -0500
+
+sflphone-client-kde (0.9.6-SYSTEM) SYSTEM; urgency=low
** 0.9.6 **
@@ -63,9 +1884,9 @@ sflphone-client-kde (0.9.6-SYSVER) SYSTEM; urgency=low
* [#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 09:53:37 -0400
+ -- SFLphone Automatic Build System <team@sflphone.org> Mon, 27 Jul 2009 09:53:00 -0400
-sflphone-client-kde (0.9.6~rc2-SYSVER) SYSTEM; urgency=low
+sflphone-client-kde (0.9.6~rc2-SYSTEM) SYSTEM; urgency=low
** 0.9.6~rc2 **
@@ -118,9 +1939,9 @@ sflphone-client-kde (0.9.6~rc2-SYSVER) SYSTEM; urgency=low
* common po files
* [#1753] Remove ILBC from pjproject
- -- SFLphone Automatic Build System <team@sflphone.org> Fri, 17 Jul 2009 19:13:11 -0400
+ -- SFLphone Automatic Build System <team@sflphone.org> Fri, 17 Jul 2009 19:12:44 -0400
-sflphone-client-kde (0.9.6~rc1-SYSVER) SYSTEM; urgency=low
+sflphone-client-kde (0.9.6~rc1-SYSTEM) SYSTEM; urgency=low
** 0.9.6~rc1 **
@@ -226,9 +2047,9 @@ sflphone-client-kde (0.9.6~rc1-SYSVER) SYSTEM; urgency=low
* [#1317] Changed tag convention
* [#1317] Cleaned git-dch
- -- SFLphone Automatic Build System <team@sflphone.org> Fri, 10 Jul 2009 15:50:55 -0400
+ -- SFLphone Automatic Build System <team@sflphone.org> Fri, 10 Jul 2009 15:49:56 -0400
-sflphone-client-kde (0.9.6~beta-SYSVER) SYSTEM; urgency=low
+sflphone-client-kde (0.9.6~beta-SYSTEM) SYSTEM; urgency=low
** 0.9.6~beta **
@@ -521,9 +2342,9 @@ sflphone-client-kde (0.9.6~beta-SYSVER) SYSTEM; urgency=low
* Config Dialog almost finished.
* Base of QT client
- -- SFLphone Automatic Build System <team@sflphone.org> Tue, 23 Jun 2009 11:15:26 -0400
+ -- SFLphone Automatic Build System <team@sflphone.org> Tue, 23 Jun 2009 11:12:06 -0400
-sflphone-client-kde (0.9.5-SYSVER) SYSTEM; urgency=low
+sflphone-client-kde (0.9.5-SYSTEM) SYSTEM; urgency=low
** 0.9.5 release **
@@ -552,9 +2373,9 @@ sflphone-client-kde (0.9.5-SYSVER) SYSTEM; urgency=low
* [#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 Automatic Build System <team@sflphone.org> Mon, 25 May 2009 11:34:40 -0400
-sflphone-client-kde (0.9.5-SYSVER~rc2) SYSTEM; urgency=low
+sflphone-client-kde (0.9.5-SYSTEM~rc2) SYSTEM; urgency=low
** 0.9.5 rc2 **
@@ -606,7 +2427,7 @@ sflphone-client-kde (0.9.5-SYSVER~rc2) SYSTEM; urgency=low
* 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 Automatic Build System <team@sflphone.org> Tue, 19 May 2009 12:08:03 -0400
sflphone-client-kde (0.9.5-0ubuntu1~rc1) SYSTEM; urgency=low
@@ -635,7 +2456,7 @@ sflphone-client-kde (0.9.5-0ubuntu1~rc1) SYSTEM; urgency=low
[ Sflphone Project ]
- -- Sflphone Project <sflphone@mtl.savoirfairelinux.net> Tue, 05 May 2009 19:16:13 -0400
+ -- Sflphone Project <sflphone@mtl.savoirfairelinux.net> Tue, 05 May 2009 19:16:09 -0400
sflphone-client-kde (0.9.5-0ubuntu1~beta) SYSTEM; urgency=low
@@ -857,7 +2678,7 @@ sflphone-client-kde (0.9.5-0ubuntu1~beta) SYSTEM; urgency=low
[ Sflphone Project ]
- -- Sflphone Project <sflphone@mtl.savoirfairelinux.net> Mon, 27 Apr 2009 17:00:03 -0400
+ -- Sflphone Project <sflphone@mtl.savoirfairelinux.net> Mon, 27 Apr 2009 16:57:00 -0400
sflphone-client-kde (0.9.4-0ubuntu2) SYSTEM; urgency=low
--
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