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sflphone-common (1.0.0-rc20110930~ppa1~SYSTEM) SYSTEM; urgency=low
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** SNAPSHOT 1.0.0-rc20110930~ppa1~SYSTEM **
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* update kde .gitignore
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* Fix bug in volume widget
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* More polishing for release
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* Bump version to 1.0.0
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* [#7023] Add the ability to load an abstract contact backend in the
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library to resolve more data, polish code
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* [#7021] More cleanup for release
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* [#7021] Refactor KDE client dbus handling, add a missing call in
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daemon and port the DataEngine to the new API
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* Remove some annoying debug
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* merge language scripts
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* remove obsolete 'VERSION' files
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* update install instructions
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* Add missing translations to gnome
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* Revert "Don't reference count DBus clients, exit core immediately
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when one of them request it"
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* Don't reference count DBus clients, exit core immediately when one
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* [7021] Add contact abstraction support
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* [#7121] Polishing library (over). Indentation, spacing and naming
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* codecs: link to libccrtp, don't use logger
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* [#7038] Fix adding contact
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* * #7037 : stop audio stream after all calls have been hanged up
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* [#7025] Add full support for bookmark
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* SFLPhone KDE do not destroy history anymore
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* Close the daemon once and for all, no more automatic respawning
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* Fix "unregistered account" bug (I hope so)
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* Close SFLPhone at the right place, it still respawn, I don't know
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* Fix regressions introduced in the last commit
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* Dead code elimination 1/3
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* Fix bug, add "add contact" option, fix warning
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* * #7019: Fix IAX codec negociation
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* Remove or comment unnecessary/unhelpful debug output
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* Fix "same as local" account setting, fix IP2IP LED color
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* Add support for some more advanced config options and add missing
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* Fix crash with noise suppressor
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* Alternative can now be selected from the call view context menu
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* Add drag and drop support, initial context menu and fix 3 bugs in
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* Add basic history drag and drop support
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* Complete contact support is back
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* * #6991 : fix IAX problems
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* Fix IAX accounts being disabled by default
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* Revert "deb: forge -g flags for pjsip"
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* * #5884: Disable debug code in pjsip
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* echo suppressor : more assertions
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* Don't let the daemon think crypto is enabled when it's not
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* Some progress on contact support
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* Remove unused getRegistrationCount()
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* remove annoying debug
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* revert SIP bit of e27e5c39bad27bae28f574eb2cba7717e8956229
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* Simplify CallManager::placeCallFirstAccount
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* * #6905 : SIP refactor
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* gnome client: be sure key exchange is set correctly
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* Move code into createSipTransport
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* Fix account registration on start
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* ManagerImpl::registerAccounts(): simplify
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* * #5884: don't mess with pjsip threads in echo suppressor
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* * #6905 : simplify udp/stun/tls pjsip transport creation
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* Restore and improve support for Call history
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* SIPVoIPLink: simplify / refactor
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* Fix libwidget linking
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* gnome: remove some debug
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* AudioRtpFactory::stop() cannot fail
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* * #6905: simplify SIP code
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* pjlib: fix build without SSLv2, fix warnings
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* Port history to the new syntax
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* Test a dock widget based implementation for contact and history
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* Disable SSLv2 support from pjsip and sflphone
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* deb: forge -g flags for pjsip
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* Fix deb packaging to get debug symbols
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* pjproject: update to last stable release (1.10)
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* Require gtk >= 2.20 and glib >= 2.24
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* tlsadvanceddialog: simplify
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* * #6902 : fix errors spotted by -DGSEAL_ENABLE
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* Update daemon dbus XML and port KDE config backend from dbus to
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* Remove unused but set variables
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* * #6929 : fix IM widget, cleanup
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* Unconditionally enable debug symbols
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* Should fix many KDE issues
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* * #6886 : hitting backspace on empty number have no side effects
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* * #6905 : fix AudioCodecFactory access in optimized builds (-O > 0)
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* Remove unsupported and broken jaunty/karmic packages
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* * #6902 : avoid using some gtk deprecated functions
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* Update dbus introspection files
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* * #6904: removed unused contactmanager
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* * #6903 : use correct dbus-cxx package name
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* * #6902: don't use individual gtk headers
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* Fix a segfault when config is not present
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* Merge latest (0.9.13) KDE code. This version is not yet ready for
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git master, but better than the previous one
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* addressbook : simplify
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* * #5659 : sflphone-plugins doesn't depend on libedataserverui
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* * #5659 : addressbook doesn't use libedataserverui
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* gnome client doesn't depend on evolution
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* * #5695: addressbook: simplify
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* * #5695: addressbook : remove AddrBookHandle from plugin
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* * #5695 : addressbook : remove unused stuff in the client
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* * #5695 : addressbook : remove unused stuff, use static mutex
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* gnome client doesn't use evolution
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* gnome: use proper API to set GTK_CAN_FOCUS
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* * #6897: removed unused focus state vars/callbacks
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* gnome: fix calls to sflphone_fill_codec_list_per_account
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* * #6623: gnome: don't leak in mainwindow
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* gnome: mainwindow whitespace cleanup
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* gnome: actions.c parameter doesn't have to be a double pointer
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* * #6895: fix memleaks, cleanup in accountconfigdialog
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* * #6893: fixes segfault in client on clean history
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* * #6894: fix leaks, cleanup in sflnotify
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* daemon: fixed prints in main
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* * #6892: simplify, fix leaks in dialpad
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* * #6887: audiopreference creates audio layer
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* * #6660: use const char * const, not std::string for globally
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* * #6852: Preferences now solely responsible for audiolayer creation.
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* * #6860: refactor uimanager, also fixes #6865
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* * #6853: hangup as soon as all digits have been deleted
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* * #6852: alsa: retry if device is busy
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* * #6852: audiolayer creation depends only on preference.audioApi
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* * #6850: gnome: fix build for gtk < 2.22.0
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* pulse: if we can't peek in audio input, we can't drop samples
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* * #6849: show error window if codecs are missing, instead of dying
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* EchoCancel: unused, remove
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* * #6629 : use number of samples as arguments for audio filters
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* * #6629 : remove unused Algorithm interface
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* * #6629 : use helper to call alsa functions and display error msgs
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* * #6841: fix some error handling
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* * #6629: simplify AlsaLayer::alsa_set_params()
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* Get gdk key definition from header
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* * #6828: Replace raw key codes by gdk defines
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* remove some debug, enhance some other
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* mainbuffer: simplify
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* * #6561 : fix phantom call after transfer
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* Conference Participant set : simplify
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* SIPCall: remove unused functions, make invite session public
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* * #6229 : remove malloc/free from pulse audio loop
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* * #6629 : simplify pulse callbacks
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* * #6629 : keep the correct audio module when frequency changes
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* * #6751: fixed erroneous debug msgs
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* callable_obj.h: removed unneeded pthread header
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* * #6629: Always restart audio driver when changing parameters (ALSA
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* gnome GUI: don't block in DBus signal errorAlert()
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* * #6629 : simplify AudioLayer creation
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* * #6629 : remove unused and unconfigurable frameSize from audiolayer
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* * #6629 : remove unused error message from audio layer
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* Fix logic error when switching audio API
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* Remove unused AudioProcessing class
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* AudioRtpRecordHandler::initNoiseSuppress() : use noiseSuppress
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* * #6629 : use DC blocker directly in audio layers
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* * #6629 : clean AudioLayer
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* * #6629 : don't store mainbuffer inside audiolayer
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* * #6629 : correct AudioLayer::notifyincomingCall()
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* * #6554: cleanup, refactoring in sipvoiplink
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* * #6554: cleanup in iaxvoiplink
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* * #6554: throw exception in getSIPCall if pointer is NULL
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* * #6554: make some methods of sipvoiplink static
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* * #6655: cleanup in managerimpl
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* * #6554: refactoring, fix memleaks in sipvoiplink
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* * #6478: remove throw specs, cleanup in voiplink
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* * #6629 : remove unused AudioDevice
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* * #6655: removed more dependencies from managerimpl
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* * #6744: simplified numbercleaner
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* conference : remove one prototype
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* Don't give glib warnings if icons are not found
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* gnome: fixed includes
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* Codec.h: removed unused function
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* * #6742 : clean dbus & icons
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* * #6699: refactor/cleanup accounts
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* timer : use second precision, not millisecond
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* calltree_update_clock : use correct type, returns something
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* * #6737: fixed typo in dbus call
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* * #6737: removed tests for removed API
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* * #6737: dbus: fixed bug from merge
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* * #6737: cleanup in accountlist
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* * #6737: cleanup in dbus
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* * #6740 : fix history double free
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* * #6740 : remove time updating thread from calls
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* * #6737 : use c99 for client
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* * #6738 : make history loading faster
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* sipvoiplink : don't crash on transfers
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* Don't build networkmanager.cpp at all if NM is disabled
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* * #6554 : simplify sipvoiplink
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* hudson: added -x to git clean command
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* added git clean to hudson script
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* audiocodecfactory: cleanup
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* * #6718: refactored setTlsSettings into SIPAccount
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* * #6718: removed more unused methods
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* * #6718: refactored confmanager code into sipaccount
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* remove unused functions
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* * #6718: confmanager: removed more unused methods
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* AudioCodecFactory : cleanup
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* #6697 : Turn callableElement struct into union
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* * #6718: confmanager: removed more unused methods
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* * #6718: confmanager: removed more unused methods
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* * #6718: removed unused dbus methods, refactoring
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* * #6699: accounts: cleanup/refactoring
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* * #6699: refactoring, cleanup in accounts
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* * #6699: more account cleanup
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* remove unused autoconf variable
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* * #6714: fixed hudson script
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* make distclean in hudson
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* added || exit 1 to run_tests.sh call
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* * #6714: fixed make distcheck for sflphone-plugins
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* * #6714: fixed make distcheck for gnome client
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* * #6714: fixed make distcheck for daemon
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* git: #6698 split the main .gitignore file
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* gnome: gpointer is already a pointer
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* gnome: calltab_init: use calloc instead of malloc
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* * #6699: more account cleanup
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* * #6699: cleanup account
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* * #6554 : more *voiplink cleanup
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* * #6558 : more sipvoiplink simplification
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* * #6558: saner loadSIPLocalIP prototype
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* gnome: #6623 clean calllists
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* * #6692: more audiolayer cleanup
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* * #6692: cleanup/refactoring in audiolayers
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* * #6692: more forward declarations, AudioThread->AlsaThread
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* * #6692: audiolayer cleanup
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* * #6692: alsalayer cleanup
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* * #6558 : remove account creator
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* * #6558 : clean sipvoiplink
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* * #6554 : cleanup sipvoiplink
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* * #6657 : fix launchpad builds for good
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* * #6675 : send RTP dtmf events only once
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* * #6655: more cleanup
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* AudioRtpSession::updateSessionMedia() : simplify
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* * #6655: more cleanup in managerimpl
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* * #6655: removed more code, cleanup
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* * #6655: more cleanup, fixed infinite loop
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* * #6655: removed more unused files
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* * #6655: removed unused mutex
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* * #6655 removed more unused code
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* * #6655: removed unused methods
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* * #6655: cleanup in main
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* * #6663: fixed segfault when off hold from transfer
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* * #6658: user's active codec selection is respected
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* * #6660: static global string should be static const char* const
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* * #6659: use g_strcmp0, not strcmp for vals that may be null
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* callable_obj: fix double free
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* calltree_display_call_info() : simplify
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* * #6657: Fix launchpad builds
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* Logger::log() : simplify
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* AudioRtpSession : privatize members
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* * #6655: more constness, cleaned up/simplified methods
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* * #6654: call DBus::_init_threading so that dbus-c++ to make it
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* set default credentials on account creation
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* AudioCodecFactory::scanCodecDirectory() : simplify and correct
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* * #6623: fixed typos
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* * #6623: fixed more leaks
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* * #6623: fixed more leaks
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* * #6623: fixed more leaks, don't print codec name if null
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* * #6623: more leaks fixed in client
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* * #6623: fix more leaks, fixed some warnings
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* * #6623: fixed leak in history
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* initialize dbus dispatcher correctly
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* Fix tests, hudson doesn't have a dbus daemon running
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* removeCall() : simplify , fix leak
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* stopRtpThread() : simplify
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* *CurrentCall : simplify
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* fix serialization of audio api (pulse / alsa)
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* account map : simplify
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* remove call from callmap before terminating it, avoid use after free
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* * #6630 : don't make DBusManager a singleton
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* call: return confID by value
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* add back history code deleted by error
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* history : reverse logic
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* simplify history serialization and remove some debug
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* remove annoying debug
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* * #6464 : replace cerr with _error
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* * #6464: replace cout with logger macros
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* replace printf() with logger macros
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* remove unused function
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* update eclipse projects
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* uimanager_new() : simplify
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* celt: simplify a bit
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* Fix CELT configure.ac test
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* * #6612 : template speex codecs
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* * #6623: refactored conference obj
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* * #6623: refactored callable object, removed leaks
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* * #6623: more cleanup, fix leaks, make global vars static and rename
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* * #6623: calltree: fixed memleaks, simplified code.
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* audiolayer: init pointer members
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* manager: catch exception on invalid hangup
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* * #6623: don't leak on calls to create_new_call
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* * #6611 : clarify codecs prototypes
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* ringtones : .au and .ul files are both ulaw
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* * #6611 : make sure samplerate converters are called correctly
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* ManagerImpl::switchAudioManager() : simplify
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* * #6623: fixed more leaks
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* * #6623: fixed more leaks
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* * #6623: fixed more leaks
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* * #6623: fixed leak, line-endings in imwidget
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* * #6627: zero-initialize pointers if they're going to be deleted
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* * #6628: don't leak calls on exceptions
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* Revert "audiortp: call join after calling stop on RtpThread"
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* sflphone-client: more constness
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* audiortp: call join after calling stop on RtpThread
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* * #6625: return 0 on successful completion
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* * #6624: fix segfault on servercallfailure
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* * #6621: Fixed double free, unlock mutex in ManagerImpl::terminate
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* * #6220: remove audio stream when peer hangs up
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* * #6596: AudioSymmetricSession shouldn't self-delete
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* resampler: grow internal buffers dynamically
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* merge up and down sampling => resampling
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* Leave test directory unchanged when running make check
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* audio algorithms : remove unused prototype
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* ringtone: detect codec from file extension
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* *AudioFile : simplify
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* * #6596: create local SDP on the stack, not the heap
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* * #6596: don't call Ost::Thread::terminate from dtor
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* audiofile: cleanup (samplerate -> unsigned)
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* samplerateconverter: cleanup
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* RingBuffer::Put() : remove unused return value
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* MainBuffer::putData() : remove unused return argument
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* audiolayer::putMain() : remove unused func
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* AudioLayer::putUrgent() : remove unused return value
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* * #6618: delete any remaining ringbuffers in destructor
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* RingBuffer::availForPut() : remove
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* * #6617: return from main rather than calling exit
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* MainBuffer::availForPut(): remove
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* RingBuffer: simplify
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* alsa : remove write only variable
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* fix memcpy declaration
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* bcopy(src, dst) -> memcpy(dst, src)
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* RingBuffer::Get() : remove constant volume argument
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* return a copy of the call ID, not just a reference.
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* MainBuffer::getDataById() : remove volume argument (always 100)
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* MainBuffer::getData() : remove constant volume argument
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* RingBuffer::Put() : remove constant volume argument
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* MainBuffer::putData() : remove constant (=100) volume argument
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* audiolayer: remove constant _defaultvolume
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* AudioRtpRecordHandler / AudioRtpSession : simplify
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* mainbuffer: fix test
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* iaxvoiplink : simplify
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* sip registration callback: fix a dbus crash
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* MainBuffer: simplify
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* AudioRtpFactory: return cached type of rtp session. The rtp session
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can have disappeared if the call was put on hold
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* AudioRtpFactory: remove unused setters
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* Fix launchpad builds
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* * #6611 : remove unused bandwidth codec information
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* * #6611: AudioCodec: remove useless/unused setters
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* make sure buffer string is initialized correctly
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* * #6596: declare certain destructors virtual
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* audiolayer : cleanup
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* Simplify doc build rules
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* * #6270: don't build dbus-api doc with make, should require make all
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* configure.ac: cleanup
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* Remove copy of dbus-c++ from libs/
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* * #6596: stop clock thread when peer hangs up
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* removed unused Fmtp.h
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* * #6595: more logical initialization order
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* * #6600 : fix account creation
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* * #6601 : fix configure.ac tests
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* remove unused variable
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* Don't mix stack and heap based allocations
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* Fix copyright (2009, 2008, 2009 -> 2008, 2009)
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* Fix warnings found by clang
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* * #6595: fix initialization order for AudioRTP
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* * #6592: removed typedef std::string CallID
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* * #6586: implement local g_slist_free_full for older glib versions
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* * #6579: fix memory leaks in client (there's a lot left)
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* ShortcutPreferences::setShortcuts() : simplify
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* * #6548: remove call to non thread-safe strerror()
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* AudioRtpFactory: each instance is associated to exactly one SipCall
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* create_audiocodecs_configuration() : make static
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* * #6269 : refactor AudioRtpSession
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* Fix AudioSymmetricRtpSession.h inclusion guard (cherry picked from
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commit c3081dce1cc1370d6d3558a4c4ef5cfac0d21caf)
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* * #6269: Rename AudioRtpSession to AudioSymmetricRtpSession
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* * #6574: Don't exit when connection to pulseaudio server fails
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* accountconfigdialog.h : remove some stuff from header
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* * #6560: fix configuration test
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* Fix warning in test
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* * #6560: don't hide password entry in security tab
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* * #6560: set initial password for SIP accounts
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* * #6506: remove useless pointer indirection
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* * 6560: password is now specific to IAX accounts
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* * #6560 : actually use, store, restore, transmit SIP credentials
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* * #6560: YamlEmitter: serialize sequences
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* YamlEmitterException: typo
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* ManagerImpl::computeMd5HashFromCredential() : simplify, fix memleak
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* * #6561: invite_session_state_changed_cb() : simplify
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* * #6561: More useful debug in VoIPLink::removeCall
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* * #6561 : fix ghost call reappearing in GUI after transfer
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* while -> for (make the code smaller)
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* * #6558 : Account::loadConfig() : move IAX code to IAXAccount
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* IAXVoIPLink::getAccountPtr : simplify
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* * #6554 : access the SIPVoIPLink directly, not per account
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* SIPVoIPLink is instanciated only once and is not associated to a
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* yamlnode: use const references when possible (still some left to do)
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* Account::_accountID: constify
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* VoIPLink: simplify, remove unused method
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* hudson test : no need to call run_tests.sh anymore
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* Remove AccountID type and AccountNULL define
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* Make check runs the test (no need to call run_tests.sh manually
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* gnome GUI: Fix tests
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* Revert "Move registration information from SIPAccount to
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* * #6392: pluginmanagertest: fix warnings reported by valgrind
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* * #6547 : remove unused exceptions
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* * #6547: CallManagerException: use runtime exceptions
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* * #6547: InstantMessageException: use runtime exceptions
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* * #6547: do not throw exceptions if some settings are not present in
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* * #6547: YamlParserException: use runtime exceptions
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* * #6547: VoipLinkException: use runtime exceptions
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* * #6547: YamlEmitterException: use runtime exceptions
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* * #6547: DTMFException: use runtime exceptions
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* * #6547: AudioFile: use runtime exceptions
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* * 6547: AudioZRtpSession: remove impossible error case
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* * #6547 : AudioRtpSession: remove impossible error case
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* * #6547: AudioZrtp: use runtime exceptions
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* * #6408 : send authenticationUsername to GUI
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* * #6408 : store/restore authenticationUsername from config file
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* SIPAccount: simplify
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* Move registration information from SIPAccount to SIPVoIPLink
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* SIPAccount::getAccountDetails : simplify
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* * #6540: yaml parser: simplify
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* sdp.cpp : fix a warning
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* * #6540: yaml parser : remove std::string typedefs
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* * #6540: Simplify yaml unserialization
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* * #6540 : add a Conf::ScalarNode constructor for booleans
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* setAccountDetails(): simplify
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* * #6408: store authentication username in daemon
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* * #6408: Be able to set the authentication username in the GUI
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* * #6507 : do not crash if the program is not sflphoned
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* macroify SIPAccount::unserialize()
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* Move all .cpp files from sflphoned target to libsflphone.la, except
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* main() : simplify, return positive error codes
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* * #6507 : find codecs dir in build directory
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* * #6392: Sdp: move clean functions to destructor
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* AlsaLayer::adjustVolume() : simplify
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* alsalayer : reduce indentation
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* malloc/free -> new/delete
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* malloc/free -> new[]/delete[]
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* malloc/free -> new/delete
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* AudioSrtpSession: simplify base64 encoding
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* * #6392: Initialize std::string from pj_str_t correctly
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* * #6392: AudioRtpSession: Initialize remote port
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* Audio settings : Initialize _echoCancelTailLength and
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* Initialize variable
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* YamlParserException : fix use of stack variable after it has been
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* * #6392: fix memory leak in history
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* * #6392 AudioCodec : fix memory leak
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* * #6392 : fix memory leak in sip account
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* * #6408: clean up sipaccount (cosmetics mostly)
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* sipaccount.cpp serialize() : reduce number of lines
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* * #6392: invalid memory access
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* * #6392 : fix invalid memory access
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* * #6479: merged useful code from MimeParameters into Codec interface
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* * #6462: fixed hangup on IP2IP call
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* added run_daemon.sh script
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* test: remove unused variable
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* Remove functions only used by a failing test (cherry picked from
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commit fcf718cb75de7f1882dc61c07bb8d300dfa10f85)
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* * #6360 : make client tests build (cherry picked from commit
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028b2835f040e51ab8ab979b32732b07b8798fce)
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* * #6360 : fix warnings in check_global test (cherry picked from
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commit 9e2bd6a7496dd64f6f48595e385760019aab1193)
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* * 6360: updated API calls in tests, but they're not building yet
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(cherry picked from commit 548f6f0f919b43772a3e9c667e5e292791281795)
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* Fixed include in tests (cherry picked from commit
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aeadc7525c1e31f936670ac8b02f0bcf387c38a8)
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* Remove unused variables and functions
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* IAX: fix warnings (cherry picked from commit
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fd7a113a11cac2cd9a7c36929e88ad28195c4c35)
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* Remove unused DEBUG define which interferes with logger.h (cherry
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picked from commit b2f72b91d0f43cb1dd94d138882a8caa9c841c24)
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* * #6392: no need to check for account NULLity since it is
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* * #6392: fix a memory leak, replace by stack allocation
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* * #6392: remove a variable assignement which confuses cppcheck
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* process_conference_participant_from_serialized() : remove unused
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* * #6392: s/free/g_free/
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* * #6392: fix a memory leak in abookfactory_load_module()
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* * #6392: remove generate_call_id() used only once
528
* * #6392: fix memory leak (opendir() without closedir())
529
* * #6392: AudioRecorder(): ensures mbuffer is set
530
* Remove SFLPHONED_VERSION from global.h, use autoconf PACKAGE_VERSION
532
* #6331: Fix deleting ringtone file after call have been answered
533
* * #6330: merged user_cfg into headers
534
* #6298: Fix conference recording file update at conference end
535
* #6298: Fix record file name serialization for conference
536
* * #6295: cleanup of codec hierarchy
537
* #6298: Fix gtk warnings
538
* * #6300: added script to run tests
539
* #6109: Add recording playback for conference
540
* * #6300: tests do not require an installed sflphone
541
* * #6295: re-removed clone methods
542
* #6109: Fix gtk_critical warnings for incoming calls
543
* #6109: Fix GTK_CRITICAL warning
544
* #6109: Fix icons when history is not activated
545
* #6109: Fix warnings
546
* #6109: Implement stop recorded file playback signal
547
* Revert "* #6295: removed unused clone method"
548
* * #6295: removed unused clone method
549
* * #6296: removed non existant file from Makefile.am
550
* #6109: Stop fileplayback for outgoing call
551
* #6109: Implement stop recording playback button
552
* Fix binding names errors in dbus introspection file
553
* #6109: Implement playback recorded file callback in client
554
* #6109: Store recorded file path on client side
555
* #6109: Add dbus methods for call recording playback
556
* * #6290: remove unused classes from utilspp
557
* * #6288: cleanup sdp
558
* * #6288: fix exception usage
559
* * #6288: simplify SdpException
560
* * #6288: cleanup in sdp.cpp/h
561
* #6109: Only display playback button if record file is set and valid
562
* * 6290: updated configure.ac to remove functor Makefile
563
* * #6290, #6289: removed unused classes from utilspp, fixed make
565
* #6109: Add button for history playback of recorded file
566
* * #6289: removed unused observer class
567
* * #6282: forward declare sdpMedia in sdp.h
568
* * #6281: renamed setCallAudioLocal->setCallMediaLocal
569
* #6183: Handle conference with more tahn two calls
570
* #6183: Fix history icons when calling back a conference from history
571
* #6183: Fix icons inconsistencies in history for conference hang up
572
* #6183: Fix toolbar actions when selecting a conference in history
573
* #6183: Fix conference serialization
574
* #6268: Serialize only calls
575
* * #6269: removed useless type testing
576
* ignore some files in test/
577
* * #6268: Remove dead class AudioSymmetricRtpSession
578
* #6251: Do not had history calls in calllist when loading history
580
* #6251: Fix insertion in history map in before saving history file in
582
* #6251: Fix history unit tests
583
* #6251: Order the list before serailization, get rid of the hashtable
585
* #6251: Implement history serialization using a list wether than a
587
* * #6253: remove external audioport from header, make all members
589
* * #6253: don't store external local audio port (used for NAT) in
591
* #6251: Add start_time timestamp in history serialization
592
* #6251: Fix call insertion in conference items
593
* #6233: Fix serialized account list terminated with a ";" character
594
* #6238: Fix draggable history calls into current calls
595
* #6233: Fix toolbar updates
597
* * #6235: remove pyc files from git tree
598
* #6233: Handle cases when one or manuy calls are unreachable in
599
createConfFomrParticipantList
600
* #6233: Handle wrong numbers in createConferenceFromParticipantList
601
* #6231: Fix drag-n-drop issue
602
* * #6173 : move sippxml in tools
603
* #6231: Fix merging issue
604
* #6183: Implement conference unserialize
605
* * #6212: remove extraneous flags from globals.mak
606
* #6183: Unserialize conference data in conference
607
* #6183: Add account information in request for conference call from
609
* #5755: Add -ldl to liker in sflphone-client-gnome
610
* #5755: Fix fedora 15 compilation issue
611
* #6183: Serialize conference participant phone number and account
612
* #6183: Add conference timestamp in serialization
613
* * #6186: don't include global.h, just logger.h
614
* #6183: Fix saving history to file
615
* #6183: Fix removing call from calllist
616
* * #6184: remove pointers to Manager from AudioRtpSessions
617
* #6183: Calling calltree_add_call explicitely for history
618
* #6183: Ability to store conference inside history tab queue
619
* * 6181: remove unused API from sipcall
620
* #6171: Implment nreCallCreated callback
621
* #6167: Fix participant list NULL ending
622
* #6149: First draft of conference creation from history
623
* #6149: Fix multiple call/conf selection callbacks ...
624
* #6129: Fix place_call function called twice for pressing enter
626
* #6129: Fix double click action for history
627
* #6149: Add dbus call for creating conference from history
628
* #6129: Fix placing call from history and addressbook (still need to
630
* * #6148: removed unused AudioRtpFactory constructor
631
* * #6145: remove unused isAudioStarted
632
* * #6145: remove unused isAudioStarted
633
* #6129: Add conference into history, fix call/conference selection
634
* * #6143: don't use getType outside of serialization methods
635
* * #6132: forward declarations instead of includes
636
* * #6132: add constness, remove redundant "inline" keywords
637
* #6129: Add timestamp to conference object to order history entries
638
* * #6128: remove unused forward declarations from header
639
* * #6127: make noncopyable class actually noncopyable
640
* * #6125: don't include AudioRtpFactory in sipcall.h
641
* #6123: Fix alsa ringback audio file
642
* #6123: Fix raw audio file loading problem
643
* #6109: Fix daemon plugin manager unit test
644
* #6109: Fix history manager unit tests
645
* #6109: Recording filename in daemon and client for history items +
647
* #6109: Refactor AudioFile to play recorded call
648
* * #6104: AudioCodec moved to sfl namespace
649
* * #6099: remove active flags from codec classes
650
* #6095: Add notification-daemon as a runtime dependencies for rpm
652
* #6095: Fix fedora 15 compilation in MineParameters.h
653
* #6095: Declare static variable explicitely for client
654
* #6095: Add logs to build OSC build machine
655
* * #6098: global variables should have file-scope to avoid name
657
* #6095: Fix compilation error for Fedora 15
658
* #6095: Update SFLphone version to 0.9.14
659
* #6095: Add specification file in opensusse build service for
661
* #6073: Fix sflphone-plugins build on launchpad
662
* #6093: Rename CodecDescriptor for AudioCodecFactory
663
* * #6089: fix warnings in make check
664
* * #6086: renamed codecs methods to audio_codecs
665
* * #6085: renamed codec related dbus calls to audio_codec
666
* #6065: Remove g_print from client, use DEBUG instead
667
* #6065: Add actions name for addressbook
668
* * #6085: renamed codecs* widgets/functions audiocodecs*
669
* #6065: Fix Addressbook runtime warnings
670
* #6065: Replace Codecs tab for Audio in account preference dialog
671
* #6065: Fix "transfert" typo
672
* #6065: Fix addressbook action runtime warning in uimanager
673
* * #6082: fixes make check by adding libcrypto libs to test
675
* #6073: Rename plugin/addressbook folders for addressbook/evolution
677
* #6074: Removed AC_SUBST from configure.ac when using
679
* #6073: Fix sflphone-plugins package build
680
* #6073: Fix sflphone-common build
681
* #6065: Fix runtime gtk warning when initializing searchbar without
683
* #6063: Fix mozilla-tellify gitignore
684
* #6063: Remove stream copy file using ifdef macro
685
* * #6012: fix make dist for sflphone-common
686
* #6063: Update .gitignore file
687
* #6058: Fix base64 encoding related warnings
688
* #6056: Fix SdpException handling
689
* #6055: Fix unknown pargma warning for gcc <= 4.5
690
* * #5949: test gcc version before disabling unused-but-set warning
691
* #6054: Fix addressbook plugin compilation warning
692
* #6048: Fix uimanager static initialization
693
* #6046: Fix addressbook factory static initialization of member
695
* #5979: Fix implicit function declaration warning
696
* #6042: Fixed discarding qualifier warnings in client
697
* #6041: Fix instant messaging unhandled case warning
698
* #5994: Implement set current addressbook name and search type in
700
* #5994: add rules for launchpad packaging of addressbook plugin
701
* #5994: Fix addressbook plugin configuration loading
702
* #6027: Fix addressbook enabled test from configuration
703
* #6027: No need of gnomedoc related macros in addressbook plugin
704
* #6027: Add NEWS file required for build
705
* #6027: Add addressbook plugin autogen.sh script
706
* #6027: Remove plugins from client
707
* #6027: Add sflphone-plugins folder at project's root level
708
* #5994: Move addressbook folder from contacts to plugin folder
709
* * #6011: removed unused Makefiles
710
* * #6010: remove unused headers
711
* * #5952: fix "string constant to char*" warnings
712
* * #6009 fixed warnings
713
* * #6003: finished cleanup of account classes
714
* * #6003, #6004: cleanup of account classes, defaultAccount no longer
716
* * #6000: fix memory leak of args object
717
* * #5998: removed using namespace std from networkmanager
718
* * #5998: removed "using namespace std" from ZrtpSessionCallback
719
* * #5998: removed using namespacestd from AudioZrtpSession.h
720
* * #5998: remove "using namespace std" from auriorecord.h and
722
* * #5998: remove using namespace std in main
723
* * #5998: removed "using namespace std" from logger
724
* * #5949: test gcc version before disabling unused-but-set warning
725
* #5994: Installation of addressbook plugin
726
* #5979: Implement codec full addressbook search from plugin
727
* #5979: Implement addressbook factory and plugin
728
* * #5981: unused webwidget removed
729
* #5966: Account config synchronization fix (for stun)
730
* #5954: Handle media name exception
731
* #5954: Fix audio codec name display in client
732
* #5954: Clean up getSessionMedia methods
733
* * #5957: getRecordingSmplRate returns a value
734
* #5954: Clean up getCurrentCodec methods
735
* * #5950: remove "converting to non-pointer type 'int' from NULL"
737
* #5915: Full gain control version
738
* * #5949: remove more unused variable warnings
739
* * #5949: remove unused/unused-but-set variable warnings
740
* * #5949: show_preferences_dialog returns a success value
741
* * #5946: cleanup of include directives, undefined function
742
* * #5515: comment out SSLv2 calls in pjsip
743
* #5915: Implement different slope for attack tme and release time for
745
* #5915: use only one input signal for gain control (removed output
747
* #5921: Fix no audio after holding a conference
748
* #5916: Add gaincontrol files
749
* #5916: Implement FFMPEG/CCRTP video streaming prototype
750
* #5903: Fix call transfer during a conference
751
* #5915: implement rms detector, first order averager, limiter for
753
* #5914: Fix call transfer when no notification request is required
754
* #5899: Fix conference right-click segfault
755
* #5884: temporary fix segfault in pjsip memory pool
756
* #5883: Fix compilation issues on maverick and lucid
757
* #5755: Fix fedora 15 compilation without patching ccrtp
758
* [#5855] Make echo canceller optional
759
* #5855: Fix echo suppression activation/deactivation
760
* #5855: Implement pjsip echo canceller
761
* #5814: Speex initialization function uses samples, not bytes
762
* #5814: Test using more unbalanced signals
763
* #5814: Fix buffer size for long echo length or long echo delay
764
* #5814: Adjust level for echo cancellation at runtime
765
* #5814: Process noise reduction before echo cancelling
766
* #5814: Implement speex post echo canceller processing
767
* #5814: Dump echo cancel file to disk
768
* #5814: Add parameters for echo cancel
769
* #5809: Add configuration parameters
770
* #5809: Implement speex echo canceller in audio rtp session
771
* #5814: Code cleanup
772
* #5814: Fix conf creation with several incomming ringing calls
773
* #5814: Fix conf creation segfault when dragging a call on hold on a
775
* #5809: Added unit test for echo cancellation and implemented
776
"process" virtual method
777
* #5709: Add always recording option in configuration
778
* #5709: Add always recording option in audio conference panel
779
* #5709: Add core functionnality for always recording (missing config
781
* #5769: Fix conference participant handling (detach/attach) and hold
783
* #5747: Fix recording icons and state for conference when adding new
785
* #5769: Code cleanup
786
* #5769: Fix hangup unsent calls
787
* #5769: Fix remove/add additional participant to conference
788
* 5769: Several fixes concerning confererence handling
789
* #5769: Fix compilation error
790
* [#5769] Fix audio streams binding in main buffer
791
* #5769: Removed access to audio mixer from audio layer
792
* #5765: Fix audio crash for illformated wavefiles
793
* #5765: Add maximum iteration for finding fmt and data "chunck"
794
* #5589: Fix compilation of libnotify under
795
* #5757: Fix abort signal when receiving INFO
796
* #5747: Add usersDetached.svg
797
* #5747: Handle offhold action for recording conference
798
* #5747: Fix off hold action for conferences
799
* #5747: Implement update conference in record action in calltree
800
* #5747: Add new icons for recording conferences
801
* #5747: Add recording state for conferences
802
* [#5738] Remove getAudioDriver call from manager (replace by
804
* [#5738] Refactor mutex protecting audiolayer
805
* [#5737] Fix HD conference recording
806
* [#5730] Fix start audio session after changing sampling rate
807
* [#5714] Fix enter keyboard event for addressbbok and history
808
* [5695] Fix addressbook combo box update when no addressbook selected
809
* [#5695] Fix addressbook initialization and search bar update
810
* [#5695] Add mutex for books_data in addressbook to protect async
812
* [#5695] Get back addressbook open from uri
813
* [#5695] Fix absolute addressbook URI for local addressbooks
814
* [#5695] Implement libebook 3.0 interface
815
* [#5571] Better logic for hangup (for case where call have not been
817
* [#5571] Update error handling in voip links
818
* [#5571] Fix compile time warnings
819
* [#5696] Fix installation dependencies for Natty
820
* [#5669] Add mention that sflphone.org is for testing only
821
* [#5693] Add natty in teh dput.conf file
822
* [#5690] Remove not useful logs
823
* [#5670] Use dynamic payload type for rtp dtmf
824
* [#5668] Clean up sflphone configuration logging
825
* [#5668] Fix hook checkbox configuration update
826
* [#5666] Fix unit tests
827
* [#5666] Manage event subscription
828
* [#5666] Emit bye request when subscription is terminated
829
* [#5666] Bye request should be sent after event subscription
830
notification is done on transfer
831
* [#5666] Make reinvite method static (to be called in pjsip
833
* [#5666] Hangup Call in manager for AccountNULL and IP2IP
834
* [#5589] Use PKG_CHECK_MODULE for every client's dependencies
835
* [#5623] Enlarge initial size of pjsip memory pool for calls (16k)
836
* [#5564] Fix audio recording resampling for g722
837
* [#5571] Move attribute handling for onhold/offhold actions in SDP
839
* [#5571] Codec negotiation refactored and unittested
840
* [#5571] Implement tests
841
* [#5571] Implement pjsip negociator
842
* [#5571] Fix unit tests
843
* [#5571] Add Fmtp.h to repository
844
* [#5571] Integrate mime types and codec factory
845
* [#5571] Handle exception when SDP negotiation fails
846
* [#5570] Add sflphoned-sample.yml in repository
847
* [#5564]: Implement stereo to mono mixing for rigntone
848
* [#5342] Update audio stream initialization
849
* [#5514] Restore test ni historytest suite
851
* [#5514] Disable test_create_history_path
852
* [#5514] use pulseaudio in sample config file
853
* [#5514] Fix test: load history from file
854
* [#5514] Do not use X
855
* [#5513] Make unit tests compile successfully
856
* [#3947] Enable unit tests in Jenkins
857
* [#5454] Fix build system to handle new version number
858
* [#5454] Update languages from launchpad
859
* [#5454] Add --without-celt in OpenSuse build service
860
* [#5454] Change version number
861
* [#5331] Added first SDP session tests
862
* [#5273] Update nightly build version tags to conform dpkg rules
863
* [#5211] Refactor send register method for iaxvoiplink and
865
* [#3950] Remove call being transfered from calltree
866
* [#5211] Use appropriate memory pool for transport selector
867
* [#5211] Fix strict aliasing rules warning in pjsip
868
* [#5211] Bring back pjsip shutting down sleep to 1000 ms
869
* [#5211] Fix registration callback segfault when closing the
871
* [#5211] Use the dialog memory pool for Route header in INVITE
873
* [#5211] Add temporary memory pool for findLocalAddressFromUri and
875
* [#5211] Use individual memory pool for dtmfs
876
* [#5211] SipVoipLink refactoring
877
* [#3950] Attended transfer for conference calls
878
* [#5284] Fix DNS resolution for Route with specified port number
879
* [#5284] Some code cleanup
880
* [#3947] Fix typo in hudson script
881
* [#5284] Added sip route to REGISTER, INVITE, BYE request, plus DNS
883
* [#5266] Use RTP dtmf as default
884
* [#5284] Added pjsip_process_route_set after setting routes in regc
886
* [#5286] Fix parsing error due to long configuration file (removed
888
* [#5286] Fix false test in configuration emmiter
889
* [#5286] Code cleanup
890
* [#5286] Updated exception handling in configuration system
891
* [#4969] Fix put SRTP call on hold
892
* [#3950] Add debug messages
893
* [#3950] Ability to perform an attended transfer
894
* [#5276] Fix initialization problem in g722
895
* [#3950] Add replace header in SIPVoIPLink::transferWithReplaces
897
* [#3950] Implemented attended method in SIPVoIPLink
898
* [#3950] Cleanup transaction request received callback
899
* [#3950] Implement dummy attended transfer in gnome-client
900
* [#5249] Fix audio samplerate update algorithm for g722
901
* [#5249] Fix uninitialized variable used in conditional jumps
902
* [#5249] Fix conditional jump error in audiolayer (uninitialized
904
* [#5267] Use autoconf 2.65 as a requirement (instead of 2.67)
905
* [#5267] Restore manual pjsip configuration and compilation
906
* [#5267] Autodetect celt version (0.9.1, 0.7.1)
907
* [#5267] Fix deprecated macros in gnome client configure.ac
908
* [#5267] Update configuration for libcelt-dev
909
* [#5267] Fix build autoconf and automake
910
* [#5227] Deactivate automatic call to astyle after compilation
911
* [#5242] Hangup every calls before leaving
912
* [#5237] Will now nightly-build for natty, Karmic deprecated
913
* [#5229] Use inner class for rtp thread instead of inheritance
914
* [#5211] Move mainbuffer unbind call in rtp final method
915
* [#5211] Initialize sip call memory pool using 16 kb
916
* [#5211] Use call memory pool in session reinvite
917
* [#5211] Add debug messages
918
* [#5211] Use and internal pool for calls
919
* [#5211] Reduce pjsip memory pool usage for stateless error messages
920
* [#5211] Refactor call deletion
922
* [#5208] Refactor codec management for accounts
923
* [#5168] Remove printf from codec's encode & decode method
924
* [#5168] Fix celt compilation on launchpad
925
* [#5168] Fix sflphoned compilation warnings in audiocodec.h
926
* [#[#5168] Must keep the g722 specific RTP rate to avoid incoming
928
* [#5168] Fix static/dynamic payload rtp session update
929
* [#5168] Throw SIPVoipLink Error if codec not instantiated in new
931
* [#5168] Fix dynamic/static codec payload type ambiguity
932
* [#5169] Fix doubled IP2IP profile when no config file
933
* [#4867] Add gtkinfobar in configuration panel
934
* [#4867] Disable input/output/ringtone selection when using default
936
* [#4952] Patches for possible buffer overflows
937
* [$4885] Fix schemas problem
938
* [#4885] sflphone-client-gnome.schemas not present during build
939
* [#4885] Add gconf shemas directories in opensuse build system
940
* [#4885] Add file/folder ownership for opensuse-factory build system
941
* [#4906] Fix opensuse-factory build
942
* [#4885] Update name dependency for libedataserver
943
* [#4885] Fix non-void function without return in dbus-c++
944
* [#4895] Update language translation
945
* [#4896] Update session timestamp when updating media
946
* [#4896] Reapply RTP hack for G722 payload type
947
* [#4896] Update recording sampling rate when updating codec
948
* [#4897] Save codecs in config for each configuration changes
949
* [#4895] Do not save config when sflphone quit
950
* [#4885] Update date for copyright
951
* [#4885] Deactivate siptest that require more than one sipp instance
952
* [#4879] Remove inmcoming call notification from IAX
953
* [#4885] Some cleanup
954
* [#4874] Add setCancel immediate/deffered for ost::Thread
955
* [#4879] Fix incoming call notification
956
* [#4878] Set keyboard focus on searchbar when selecting addressbook
957
* [#4874] Fixed compilation warning
958
* [#4874] Fixed compilation warning in sipvoiplink
959
* [#4874] Fix compile time warning in RTP record handler
960
* [#4874] Fix conditional jump in SDP
961
* [#4874] Fix conditional jump based on uninitialized value
962
* [#4874] Store call id within rtp thread context
963
* [#4874] Fixed conditional jump based on uninitialised value in
965
* [#4871] Fix default account fetching
966
* [#4870] Delete RTP session when Refusing an incoming call
967
* Restore IP to IP call
968
* [#4857] Fix audio codec negotiation problem
969
* [#3947] Adjust ressources allocated to compilation
970
* [#3947] Disable unit tests in Hudson
971
* [#4305] Free mutex only when really quiting SFLphone
972
* [#4859] Update copyright to 2011 in every source file
973
* [#3218] Character '.' stripped by the caller engine
974
* [#4854] Fix typos, desktop entry
975
* [#4847] Apply RTP modification to ZRTP session
976
* [#4852] Update Karmic and Lucid dependencies
977
* [#4852] Add Libedataserver and libedataserverui as gnome client
979
* [#4852] Add authentication mechanism for EDS
980
* [#4851] Fix segfault when closing pulseaudio layer too rapidly
981
* [#4808] Some otehr cleanup
982
* [#4808] Made some cleanup
983
* [#4808] Added mutex in rtp session for codecs and noise process
984
* [#4847] Update audio processing when updating RTP media
985
* [#4842] Add support for linking with gold/ld --no-add-needed
986
* [#4808] Make update g722 related static/dynamic payload logic
987
* [#4827] Upper limit on the number of contacts to import from EDS is
989
* [#4808] Fix put call on/off hold
990
* [#4808] Implement early RTP start for incoming calls
991
* [#4808] Audio stream is no longer start within RTP session.
992
* [#4808] Removed coupling between audio layer and and RTP session
993
* [#4702] Start audio rtp session as soon as it is created
994
* [#4702] Init timestamp to 0
995
* #4702: Send RTP packets immediately, no need of outgoing queue
996
* [#4784] Update dbus-c++ version from gitorious
997
* [#4702] Update RTP timeouts
998
* [#4702] Lengthen RTP timeouts
999
* [PATCH] Fixed compatibility with old libtool versions.
1000
* [PATCH] Accept older libebook (Maemo 5 has 1.4.2)
1001
* [PATCH] Fixed double-free error in preferences dialog
1002
* [PATCH] Fixed building of sflphone-common on Maemo5
1003
* [PATCH] Improved Gnome client initialization error handling. 1. It
1004
no longer segfaults when sflphoned isn't available. 2. User is
1005
provided with GUI error dialog.
1006
* [PATCH] Improved autogen.sh scripts 1. They do not require bash
1007
anymore 2. Added workaround for Debian bug #565663 3. Replaced
1008
manual autotools invocations with single autoreconf call 4. Non-zero
1009
return status on failure
1010
* Revert "[#4468] libtool <= 2.2 doesn't have LT_INIT macro so
1011
AC_PROG_LIBTOOL should be used instead."
1012
* Revert "[#4468] Libebook 1.4 is sufficient"
1013
* Revert "[#4468] Apply big path on dbus communication system"
1014
* [#4468] Apply big path on dbus communication system
1015
* [#4468] Libebook 1.4 is sufficient
1016
* [#4468] libtool <= 2.2 doesn't have LT_INIT macro so AC_PROG_LIBTOOL
1017
should be used instead.
1018
* [#4639] Fix determining default addressbook if this property is not
1020
* [#4639] Fix memory leaks in Addressbook
1021
* [#4637] Fix opening default addressbook at sflphone init
1022
* [#4622] Free yaml events while parsing configuration file
1023
* [#4623] Fix conditional jumps based on uninitialized variable
1024
* [#4622] Fix leaks in yaml serialization engine
1025
* [#4616] Fix addressbook warnings
1026
* [#4514] Adjust RTP timestamp
1027
* #4527: Rename Karmic libyaml and Celt package in debian control file
1028
* #4495: Rework addressbook opening loop
1029
* [#4524] Increment RTP count when sending data
1030
* [#4524] DO NOT start RTP session twice
1031
* [#4367] Use PKG_CHECK_MODULE for celt
1032
* [#4367] Fedora package celt as celt (not libcelt)
1034
* [#4367] Update .po files
1035
* [#4367] Fix segfault in gensin
1036
* [#4354] Make celt a direct dependency on launchpad opensuse build
1038
* [#4367] Make celt a required package, option --without-celt valid
1039
* [#4367] Fix zrtp timestamping error
1040
* [#4367] Fix audio zrtp timing
1041
* [#4367] Dispatch ZRTP packets
1042
* [#4367] Fix segfault when unloading account map
1043
* [#4367] Fix zrtp session
1044
* [#4367] Implement on packet receive
1045
* [#4367] use symetric audio rtp session, not dual
1046
* [#4367] Reduce packet receive/sent timeout
1047
* [#4367] Reduce RTP timeouts
1048
* [#4367] Move speaker data receive
1049
* [#4367] Move speaker data receive
1050
* [#4367] Move receive speaker data method
1051
* [#4367] Remove debug in rtp session
1052
* [#4367] Fix g722 codec clock rate
1053
* [#4367] Fix noise suppression initialization
1054
* [#4367] Fix segfault in RTP mic fadein method
1055
* [#4367] Refactor mic data encoding in rtp session
1056
* [#4367] Implement RTP main loop
1057
* [#4367] Fix compilation problem
1058
* [#4367] Fix AudioRtpclass using TRTPSessionBase
1059
* [#4367] Fix AudioRtpSession putDtmfEvent shadowing
1060
* [#4367] Fix AudioRtpSession putDtmfEvent shadowing
1061
* [#4367] Refactor RTP session (phase 2)
1062
* [#4367] Refactor RTP session (phase 1)
1063
* [#4367] Remove Redeclaration of SymetricAudioRtpSession in
1065
* [#4265] Add continue statement in for loop for invalid addressbook
1066
* [#4261] Makes addressbook initialization more robust
1067
* [#4257] Add maverick in build system
1068
* [#4233] Add sdp related unit tests
1069
* [#4233] Add condition and signal in two incoming call test
1070
* [#4243] Fix segfault in AudioSrtpSession
1071
* [#4243] Fix memory leak in AudioSrtpSession
1072
* [#4243] Make audio srtp optional in for incoming call
1073
* [#4243] Add boolean variable to make sure remote crypto context
1074
initialized only once
1075
* [#4243] Add documentation to AudioSrtpSession
1076
* [#4243] Use 80 bits authentication tags by default
1077
* [#4243] Init audio srtp remote crypto context in
1078
call_on_media_update
1079
* [#4243] Move SDP negotiastion in mod_on_rx_request
1080
* [#4243] Implement initLocalCryptoInfo to be called at different
1082
* [#4243] Init init local crypto context in when initializing audiortp
1083
* [#4243] Change key length according to sdes negociation
1084
* [#4243] Associate callid to accountid for incoming calls
1085
* [#4242] Fix no SDES keys in IP2IP calls
1086
* [#4242] Fix no SDES keys in IP2IP calls
1087
* [#4233] Test for call on/off hold
1088
* [#4233] Add two incoming call test
1090
* [#4233] Add 2 outgoing simultaneous call unit tests
1092
-- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 30 Sep 2011 13:51:04 -0400
1094
sflphone-common (0.9.7~rc1~ppa1~SYSTEM) SYSTEM; urgency=low
1096
** 0.9.7~rc1~ppa1~SYSTEM **
1098
* [#2462] Set explicitly the transport on incoming call too
1100
* [#2462] Use different address for SDP and call IP
1101
* [#2462] Use published address in SIP-SDP
1102
* [#2181] Fixed changelog files
1103
* [#2181] Updated spec file
1104
* [#2402] Fix pointer to int conversion warning (atoi)
1105
* [#2402] Remove daemon warnings, make indent
1106
* [#2459] Make sure the stream is opened when the call is answered
1107
* [#2402] Add conference related picture in documentation
1108
* [#2443] Not much ...
1109
* [#2399] Fix dialing display problem
1110
* [#2450] Fix incoming call already in conference crash
1111
* [#2399] Display peer name on the first line and peer number on the
1113
* [#2450] Handle 403 FORBIDDEN when refused
1114
* [#2447] Bind offHold/onHold actions to button in gtk client
1115
* [#2447] Bind hangup action to button for conference
1116
* [#2447] Add conference action in gtk client's ToolBar
1117
* [#2381] Disable the password hashing in config file
1119
* [#2366] Set callback to null when deleting Pulseaudio streams
1120
* [#1313] Fix main buffer unit test
1121
* [#1313] Fix audio layer unit test
1122
* [#2315] Hide pw in security tab, display when editing, sync with
1124
* [#1313] UnitTest change AudioRtpSession for AudioSymetricRtpSession
1126
* [#2402] Code cleanup
1127
* [#2444] Add debug to catch occasional crash when loading client's
1129
* [#2444] Add debug info to catch occasional crash when loading config
1131
* [#2402] Restore Call menu translations
1132
* [#2403] Use the published address if checked in GUI
1133
* [#2442] Add protection test in sdp
1134
* [#1841] Reapply pjsip patch concerning DNS SRV resolution
1135
* [#2384] Tags incoming call as direct SIP call, if applicable
1136
* [#2402] Change the monkey face
1137
* [#2315] Enable user to display password in clear text
1138
* [#2434] Force optimization level at 2
1139
* [#2284] Fix dbus_get_all_ip_interface compilation warnings
1140
* [#2431] Popup main window on incoming if applicable
1141
* [$2402] Fix simple warnings
1142
* [#2402] Fix implicit variable init order in LibraryManagerException
1143
* [#2402] Fixing implicit variable initialization warnings in
1145
* [#2402] Revert atoi change, fixing codec list doubled entries
1146
* [#2402] Fix gpointer to gint conversion
1147
* [#2402] Fix pointer casting to integer different size warning in
1149
* [#2402] Fix warning discarting qualifiers from pointer target
1150
* [#2402] Fix gtk tree view assignement from incompatible type warning
1151
* [#1669] Fix audio recording folder utf-8 non compatibility issue
1152
* [#2414] Clean up debugs
1153
* [#2414] Use transport set in iptoip Account and update it frm
1155
* [#2348] Use macro N_() to mark ui.xml strings as translatable
1156
* [#2414] Rename getSipAddress/setSipAddress functions
1157
* [#2407] Fix volume controls display
1158
* [#2407] Fixes dialpad
1159
* [#2383] Set ip to ip config when clicking apply button
1160
* [#2404] Update call-to script - Maxime Chambreuil
1161
* [#2405] Client handles unknown call in current state as well
1162
* [#2383] Add DBUS signal to send IPtoIP local address and port as
1164
* [#2383] Add Ip to IP config change apply call back
1166
* [#2402] Code cleanup
1167
* [#2383] Do the same for IPtoIP (init localn ip with first in the
1169
* [#2383] Use first interface in the list if local addresss is not
1171
* [#2403] Clean up unuseful addresses/ports
1172
* [#2403] Use the IP profile SIP port as global SIP port
1173
* [#2383] Fix dbus_get_all_ip_interface warnings
1174
* [#2383] Take into account sameAsLocal when loading published address
1175
* [#2383] Tsake into account sameAsLocal option when saving published
1177
* [#2383] Update local ip address in ip to ip config
1178
* [#2383] Save ip 2 ip local port in config
1179
* [#2406] Update toolbar at startup
1180
* [#2284] Remove redefinition warnings + speex warnings
1181
* [#2383] Fix security table in account config
1182
* [#2383] Save ip 2 ip network interface parameters in config
1183
* [#2403] Restore sip transport selector
1184
* [#2383] Fix filling the Localt IP Address on account creation
1185
* [#2383] Fix Gtk-Critical when checking STUN
1186
* [#2383] Fix reopening account configuration display issue
1187
* [#2383] Load IPtoIP local address and port in preference iptoiptab
1188
* [#2383] Add LocalAddress and Localport in Preference IpToIp tab
1189
* [#2403] Use the address and port associated to the account as often
1191
* [#1753] Removed pjsip generated files
1192
* [#1753] Removed remaining milenage lib references
1193
* [#2383] Add _publishedSameasLocal variable in sipaccount
1194
* [#2383] Add PUBLISHED_SAMEAS_LOCAL variable in config
1195
* [#2383] Fix stun set active or not when opening config
1196
* [#2181] Added RPM 64bits dbus patch
1197
* [#2402] Code indentation
1198
* [#2313] Force $(HOME).cache directory creation at startup
1199
* [#2383] Separate network interface and published address in account
1201
* [#2400] Change dbus service installation path to libdir
1202
* [#2382] Move TLS related published address options in security tab
1203
* [#2382] Indent accountconfigdialog.c
1204
* [#2181] Install libdbus-c++ in $pkglib instead of $lib
1205
* [#1753] Remove ILBC code and disable it by default in the configure
1206
* [#1753] Remove milenage directory
1207
* [#2382] Fix switching interaface instabilities
1208
* [#2396] Save local ip in account creation wizard
1209
* [#2284] Remove warning on hold
1210
* [#2387] Fixes history searching and filtering
1211
* [#1215] Add samplerate display in the GUI
1212
* [#1663] Voicemail icon reflects voice messages
1213
* [#2395] Fix account registration ( specifically with callcentric)
1214
* [#2386] Strip "sip:" on incoming call, fixing history call back
1215
* [#2181] Updated spec files
1216
* [#1215] Display codec name in calltree instead of status bar
1217
* [#2390] Move back nbCalls and stopStream higher in refuseCall
1218
* [#2392] Fix ringtone during call in IAX
1219
* [#2391] Stop audio streams when there is 0 calls only
1220
* [#2391] Add debug when call state is not valid
1221
* [#2390] Clear returns in IAXvoipLink::sendAudioFromMic() method
1222
* [#2380] Fixing IncomingCallNotification not regular
1223
* [#2339] Query conference at client startup
1224
* [#2339] Working conference querying at startup
1225
* [#2339] Add conference in call tree
1226
* [#2339] Primitives to query conferences at client startup
1227
* [#2320] Add account selection in history
1228
* [#2355] Temporary solution: do not delete pointer when removing
1230
* [#2380] Change algorithm in AudioRtp to trigger an
1231
IncomingCallNotification
1232
* [#2274] Comment sdebug in MainBuffer flush method
1233
* [#2274] Add flushMain() in ManagerImpl::addStream
1234
* [#2274] Add getBufferID() method in ring buffer
1235
* [#2274] Fix warning, comment debug in ringbuffer's flush method
1236
* [#2274] Use AudioLayer flushMain() and flushUrgent() in ALSA
1237
* [#2274] Clean up unused variable warning
1238
* [#2274] Protect minbudffer pointer on flushing
1239
* [#2274] Fix playATone method which writing empty buffer in urgent
1241
* [#2274] Use audio layer flushUrgent and flushMain in createStreams
1242
* [#2274] Use flush audio calls from audiolayer
1243
* [#2274] Flush when peer answered call
1244
* [#2375] Flush main buffer in iax when answering a call
1245
* [#2274] Parse displayname using c++ string method
1246
* [#2375] Flush main buffer when off holding calls
1247
* [#2375] Flush main buffer mon RTP startup
1248
* [#2376] Use now Pulseaudio module-cork-music-on-phone
1249
* Updated OSC packaging
1251
-- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 20 Nov 2009 14:00:02 -0500
1253
sflphone-common (0.9.7~beta~ppa1~SYSTEM) SYSTEM; urgency=low
1255
** 0.9.7~beta~ppa1~SYSTEM **
1257
* [#1933] Cleanup debug
1258
* [#1933] Clean up debug
1260
* [#1933] Set the IAx format earlier
1261
* [#1933] Move IAX sendAudioFromMic outside if (call) statement
1262
* [#1933] Fix startstream when offhold in iax and add debug concerning
1264
* [#2371] sflphone_notify_voice_mail: minor gettext message formatting
1266
* [#2371] select_account_cb: properly gettextize status message
1267
* [#2371] show_account_list_config_dialog: properly gettextize status
1269
* INSTALL: Minor tidyup of core install guide
1270
* Add /sflphone-client-gnome/src/icons/Makefile to .gitignore
1271
* [#2181] Updated OpenSUSE files (tmp)
1272
* [#1933] Add debug for codec negociation for iax
1273
* [#1933] Get rid of getMicAvail and getMicData in audiolayer (not
1275
* [#1933] Add "audio codec not determined" error in IAX
1276
* [#1933] Test flush data
1277
* [#1933] Do not need to start audio stream in iax anymore
1278
* [#1933] Protecting pointer
1279
* [#2284] Remove more compilation/execution warnings
1280
* [#2284] Cleanup debug in client, use DEBUG instead of g_print
1281
* [#2284] Clean up uimanager
1282
* [#2370] Remove warnings
1283
* [#2366] Clean up other debug
1284
* [#2366] Clean up debug
1285
* [#2366] Call pa_xfree explicitely in writeToSpeaker
1286
* [#2284] Remove address book warnings
1287
* [#2365] Fixes bad cast
1288
* [#2352] Fix continuous ringing when peer hangup and call not yet
1290
* [#2181] Added version support
1291
* [#2181] Fixed some minor issues
1292
* [#2360] Moved MainBuffer from AudioLayer to ManagerImpl
1293
* [#2352] Makes getMainBuffer() everywhere
1294
* [#2352] Use 50 sec latency on pulseaudio stream creation
1295
* [#2352] Add alsa debug
1296
* [#2359] Update repository documentation
1297
* [#2354] Move pulseaudio disconnectAudioStream after stopping main
1299
* [#2352] Adjust nb byte copied in pulseaudio according to
1301
* [#2352] Specify pulseaudio tlength parameters using pa_usec_to_bytes
1302
* [#2322] Convert italian translation to UTF-8
1303
* [#2357] Fixes window size
1304
* [#2357] Display only actionnable tool item
1305
* [#2333] Update streams parameters
1306
* [#2347] Use GNOME user settings for Menu and Toolbar appareance
1307
* [#2349] Load/Save properly audio params
1308
* [#2322] Update translations from Launchpad
1309
* [#2181] Added Francois Marier script
1310
* [#2350] Remove non-valid test
1311
* [#2181] Updated launchpad packaging
1312
* [#2333] Fix Pulseaudio Capture
1313
* [#2333] Use pulseaudio ADJUST_LATENCY flag and ALSA RT-SCHEDULING
1314
* [#2333] Pulseaudio Interpolate timing
1315
* [#2333] Change (again) Pulseaudio settings to fit logiteck usb hdw
1317
* [#2333] Adjust pulseaudio fragment size to 4096 (max sflphone's
1319
* [#2284] Remove recurrent compilation warning (g++ linker problem)
1320
* [#2333] Safer Audiostream parameters
1321
* [#2333] Fix alsa playback to reduce underrun
1322
* [#2333] Better audiostream parameters
1323
* [#2181] Updated version management
1324
* [#2333] Exclusive test in playback loop
1325
* [#2181] Updated build system
1326
* [#2333] Less underrun with these value
1327
* [#2333] Update playback audiostream parameters
1328
* [#2333] Lengthen the audio buffer reduce number of underrun in
1330
* [#2333] Add ALSA recovery functions for underrun (begin)
1331
* [#2333] Add pa_stream_trigger in pulse audio underrun callabck
1332
* [#2048] Reduce prebuffering in pulseaudio (which affect incomming
1334
* [#2316] Do not display any icons to the right on the history tab
1335
* [#2333] Comment pa_stream_trigger in pulseaudio underrun
1336
* [#2333] Modify pulseaudio streams parameters
1337
* [#2318] Fix transfer tool button double signal
1339
* [#2333] Fix ALSA ringtone
1340
* [#2333] Flush all main buffer before starting audio
1341
* [#2333] Open/Close Alsa thread between calls while there is no audio
1342
* [#2333] Add debug message and test condition on starting playback
1344
* [#2181] Fixed gnome client makefile
1346
* [#2308] Remove getTelephoneTone debug
1347
* [#2308] Change plughw for default in ALSA
1348
* [#2308] Oups, forgot to change function name in audiolayertest.cpp
1349
* [#2308] Cleanup in pulseaudio code (debug, function name)
1350
* [#2308] Fix pulseaudio stream closing assertion failure
1351
* [#2308] Moved pulseaudio mainloop locking from AudioStream
1353
* [2308] Fix latency at the beginning of a call, when playing DTMF and
1355
* [#2181] Updated karmic
1356
* [#2317] [#2319] Fix address book toggle button contextual behaviour
1357
* [#2308] Stop stream when refusing a call
1358
* [#2308] Stop pulseaudio stream when peer hungup
1359
* [#2308] Fix tone and ringtone
1360
* [#2312] Display the STUN entry widget when opening the tab
1361
* [#2308] Implement two different callbacks for capture/playback in
1363
* [#2309] Open/close pulseaudio connections in startStream/stopStream
1364
* [2308] Leave pulseaudio stream running, do not cork/uncork them
1366
* [#2295] Set gtk file chooser to None if nothing is set in
1368
* [#1976] Add codec and conference documentation
1369
* [#2209] Fix recording in regard of resamling
1370
* [#2297] Update .gitignore
1371
* [#2297] Update translation files
1372
* [#2297] Add reference to our coding standards
1373
* [#2297] Remove old docbook code
1374
* [#2296] Reinit tls account settings after modification
1375
* [#2253] Add DcBlocker class to remove capture's dc offset
1376
* [#2034] Fixes for TLS transport to initialize
1377
* [#2284] Add silent build rule + client clean warnings
1378
* [#2274] Fix unserialize history items in cilent at startup
1379
* [#2274] Complete display name parsing and displaying
1380
* [#2274] Parse the Display Name in sip INVITE message
1381
* [#2050] Fix capture volume control in ALSA
1382
* [#1970] Volume controls disable when using pulseaudio
1383
* [#1970] Disable volume controls when using pulseaudio
1384
* [#2277] Fix direct ip2ip ZRTP enabling/disabling in ip2ip
1386
* [#2181] Added launchpad debian files
1387
* [#2181] Added spec files for OSC
1388
* [#2274] Set display name for "Contact" sip header as the hostname
1389
* [#2181] Fixed daemon issues
1390
* [#2181] Fixed gnome client issues
1391
* [#1976] Remove warnings - need to fix the transfer
1392
* [#2006] Add init is_rec variable in ManagerImpl
1393
* [#2006] Update codec display on call selection
1394
* [#2006] Restore double click actions in history and contact calltree
1396
* [#2176] use XDG_CACHE_HOME when initializing sfl.zid file
1397
* [#1976] Fix calltree switching from history
1398
* [#2209] (Re)Fix cache for zid
1399
* [#2209] Clean up debug messages
1400
* [#2209] Clean debug messages
1401
* [#2209] Fix trasnfering a call during a conference
1402
* [#2209] Speex decode must return the number of bytes
1403
* [#2209] Change frameSize speex 32kHz
1404
* [#2209] Fix speex codec framesize
1405
* [#2209] Reinit converterSamplingRate in RTP sessions
1406
* [#2209] Change speex ultra wide band framesize
1407
* [#1747] Add pixmap data
1408
* [#2252] Fix Receiving a server error 488 crashes the callee
1409
* [#2209] Fix iax low rate packate sending
1410
* [#2209] Clean up debug messages
1411
* [#2209] Add resampling changes for IAX
1412
* [#2209] Clean up resampling code
1413
* [#2209] Fix latency introduced by pulseaudio
1414
* [#2209] Fix initialization of mainbuffer's internal sampling rate
1415
* [#2176] Fix upsampling buffer size in audiolayer
1416
* [#2209] Add dynamic converter sampling rate in audiortp sessions
1417
* [#1747] Fixes runtime warnings
1418
* [#1747] Remove from repo
1419
* [#1747] register our icons to be used as stock icons
1420
* [#2209] Fix number of byte in alsa's write to speaker
1421
* [#2209] Fix putting non-resampled data in RTP's mainbuffer
1422
* [#2209] Add alsa resampler
1423
* [#2209] Add a samplerate converter in PulseLayer
1424
* [#2209] Add mainbuffer's internal sampling rate and flushall method
1425
* [#2176] Add mainbuffer stateInfo debug method
1426
* [#2209] Resampling is optimal using SRC_LINEAR not SRC_FASTEST
1427
* [#2176] Remove debug recordings
1428
* [#2176] Fix Holding a conference participant on new calls
1429
* [#2224] Add confID in callable object
1430
* [#2176] Fix putting onhold a call participating to a conference when
1432
* [#2176] Reset auidio buffers when adding streams (rtp, audiolayer)
1433
* [#1976] Use xml to describe toolbars - Add a naviguation toolbar
1434
* [#2176] Remove conference default_id in joinParticipant
1435
* [#2176] Display error message in alsa's snd_pcm_avail_update call
1436
* [#2176] Alsa mic avail data debug
1437
* [#2176] Add some debug message for mic loss problem
1438
* [#2176] Flush mic ring buffer when offholding a call
1439
* [#2176] Reset ringbuffers' readpointer when adding main participant
1440
* [#2176] Fix getAvailData algorithm
1441
* [#2176] Reset ringbuffer's readpointer when adding a new participant
1443
* [#1744] Regex object renamed to Pattern. Previous attempt at
1445
* [#2176] Fix detach main participant problem when adding new one
1446
* [#1976] Use right domain to translate
1447
* [#1976] Add xml menu description
1448
* [#2176] Store a list of confernece participant in client
1449
* [#2176] Fix add participant, joinparticipant methods
1450
* [#2181] Do not install dbus-c++ headers + add return value
1451
* [#2176] Fix minor call handling instabilities
1452
* [#2174] Fix incoming IP call contact address
1453
* [#2211] Add test to protect NULL pointer
1454
* [#1163] Add Advanced account configuration section
1455
* [#2176] Add some usefull comments and debugging info
1456
* [#2176] Add conditions to display security icons in conference
1457
* [#2176] Fix detaching one participant while keeping communication to
1459
* [#2176] Reenable userActive.svg in call tree
1460
* [#2176] Make user active blue (not red)
1461
* [#2176] Fix user active picture
1462
* [#2176] Fix "hidden" merge conflict in sipvoiplink
1463
* [#2176] Remove iax audio stream on peer hungup
1464
* [#2174] Multiple UDP transports functional (TESTED with 2 accounts
1466
* [#2176] Fix fix audio stream binding in iax
1467
* [#2174] Create a default UDP transport + use tp selector for dialogs
1469
* [#2176] Register iax audio stream in mainbuffer
1470
* [#2176] Fix getAudioCodecName in IAXvoipLink
1471
* [#2176] Fix iax account init
1472
* [#2176] Handle multiple account using the same sip transport
1473
* [#2165] Add .png files
1474
* [#2176] Small fixes concerning dtmf
1475
* [#2176] Fix make uninstall in codecs
1476
* [#2174] remove stund makefile generation
1477
* [#2176] Add conference lock
1478
* [#2174] Add transport selector for multiple accounts
1479
* [#2176] Change userActive picture from red to blue
1480
* [#2176] Fix security pixbuff in calltree
1481
* [#2176] Replace sfl.zid in .cache/sflphone instead of .sflphone
1482
* [#2176] Fix add call description
1483
* [#2176] Remove detach button from toolbar
1484
* [#2176] Fix calltree call description state and state code in
1486
* [#2176] Fix pulse audio double free
1487
* [#2176] Fix conference selection
1488
* [#2174] Clean up - remove stun settings in client network
1490
* [#2174] Remove voviva stun code
1491
* [#2174] Rsolve STUN with pjsip - DO NOT WORK
1492
* [#2165] Add user svg
1493
* [#2165] Debugging sip call failed
1494
* [#929] Link against uuid if installed
1496
* Fixed bugs related to libsexy (with GTK < 2.16)
1497
* [#929] Remove uuid-dev dependency in the core
1498
* [#2165] Debugging no negociated codecs at communicatio start
1499
* [#2165] Fix calltree bug (gtktreestore instead of gtkliststore)
1500
* [#2165] Fix several merge problems
1501
* Updated opensuse packaging script
1502
* [#1163] Add missing figures
1503
* [#1163] Update INSTALL file
1505
* [#2165] Add recordabe interface
1506
* [#2165] Finish recording refactoring for call (not for conference)
1507
* [#2165] Enable speaker recording for two different calls
1509
* [#2165] Implement call recording using the Recordable interface
1510
* [#2165] Add get and set to AudioLayer's audio recorder
1511
* [#2165] Add class recordable from which inherit call and conference
1512
* [#2006] Fix G722 and Speex 8khz codec conferencing
1513
* [#2006] add recording of audio buffers
1514
* [#1163] Add general settings section
1515
* [#1163] Fixes makefile error
1516
* [#2006] Fix some minor issues
1517
* [#2006] Drag a conference call on another conference call
1518
(difference conferences)
1519
* [#2006] Fix dragging a conference on itself
1520
* [#1744] Integrating some of the needed regular expression patterns
1522
* COmplete call features
1523
* [#1744] Added support for named subgroup in the Regex object. Also,
1525
* [#1744] Adds thread safety features, compile() and setPattern()
1526
methods to the Regex class.
1527
* [#1744] Fix inconsistency in the finditer method from the last
1529
* [#1744] Added regex pattern object built on top of libpcre. To be
1531
* [#1744] Initial commit towards implementing RFC4568. Unimplemented
1533
* [#2157] Hide "security" and "advanced" tabs for IAX under account
1534
* [#1163] Add call features section
1535
* [#2006] Add joinConference capabilities
1536
* [#2006] Add dbus joinConference signal
1537
* [#2006] Drag a conference call onto a conference to add it
1538
* [#1163] Add addressbook section
1539
* [#2006] Drag a conference call onto a single call to create a
1541
* [#2006] Expand rows automatically
1542
* [#2006] Add minimal multiple conference handling
1543
* [#2006] Add atached/detached conference icons
1544
* [#2006] Add function processRemainingParticipant
1545
* [#2006] Deep refactoring, fix hangup bug
1546
* [#1163] Update documentation - Accounts part
1547
* [#1976] Integrate user doc to gnome client build system
1548
* [#2122] Remove double inclusion in dbus-c++/src/Makefile.am
1549
* Remove pjproject version number
1550
* [#2006] Fix peerHungup
1551
* [#1976] Make Yelp accessible from the GNOME client (need to install
1552
the sflphone.xml first)
1553
* [#2006] Fix multiconferencing hangup
1554
* [#2006] Fix hangup calls in a conference
1555
* [#2150] Make IAx2 reappear
1556
* [#2006] Fix detach participant on multiple call
1557
* [#2006] Can remove rining call from a conference
1558
* [#2006] Reinit confID when removing a participant
1559
* [#2006] Remove get isCurrentCAll in hangup/peerhungup (SipVoipLink)
1560
* [#2006] Fix refuse call
1561
* [#2006] Fix answerring incoming call
1562
* [#2006] Refactor conference's participant list
1563
* [#2101] Re-integrate test compilation in main build system
1564
* [#2101] Make the test directory compile
1565
* [#2136] Restore history functionality
1566
* [#2006] Fix binding main participant to himself
1567
* [#2006] Fix add current/incoming/onHold participant to an existing
1569
* [#2006] Fix add incoming calls to an already created conference
1570
* [#2006] Fix remove stream
1571
* [#2006] Fix detachParticipant/removeParticipant switchCall ids
1572
* [#2006] Fix adding a call in conference having state "CURRENT"
1573
* [#2006] Remove/add main participant from conferences
1574
* [#2006] Hold/unHold conference
1575
* [#2006] Detach a partcipant from drag n drop
1576
* [#2006] Hangup a conference
1577
* [#2006] Add hold/unhold conference dbus messages
1578
* [#2034] gtk-ui fix under the "basic" tab.
1579
* [#2006] Fix dragging calls on conference calls
1580
* [#2006] Fix detach participant from a conference
1581
* [#2034] Added default message is status bar under the account config
1583
* [#2112] Fix a crashed caused when a non-md5 password was sent to
1585
* [#2006] Detach participant by ID
1586
* [#2006] Fix addParticipant method in managerImpl to handle
1587
incoming/answered calls
1588
* [#2006] Add addParticipant method in managerimpl and related dbus
1590
* [#2111] Added the ability to configure zrtp on sip.sflphone.org from
1591
* [#2106] Fixed problem in the account assistant under gtk-ui. Also,
1593
* [#2006] Fix dragging a conference call on another conference call
1595
* [#1904] Small UI fix. Assistant was moved from "Call" to "Edit"
1597
* [#1904] Fix a wrong label under gtk-ui.
1598
* [#2034] Renaming and source code splitting.
1599
* [#2034] Status bar added to account window to better reflect the
1601
* [#2006] Make calltree_remove_call recursive (for GtkTreeStore)
1602
* [#1110] Small gtk-UI fix in the account window (alignment).
1603
* [#2006] Fix remove conference, display children which are still
1605
* [#2006] Recursive function call in calltree_update_call
1606
* [#2006] Add multilayered capabilities to calltree (GtkTreeStore)
1607
* [#2006] Implement remove conference in calltree
1608
* [#2034] Now useless as Direct Ip calls settings moved under
1610
* [#2034] Edit/add buttons were set insensitive all the time under
1612
* [#1887] Information about the state of the current SIP call is
1614
* [#2006] Add call tree remove callback
1615
* [#2006] Fix create_conference function
1616
* [#2006] Update conference_added_cb to add new conference to the list
1617
* [#812] Added new tab under GTK-ui Preferences. Moving Direct Ip
1619
* [#2121] Disable temporarily test compilation
1620
* [#2006] Fix conferencelist to handle conference_obj_t instead of
1622
* [#2006] Add conference_obj structure
1623
* [#2121] Update version
1624
* [#2006] Fix conference selection
1625
* [#2101] Use the new source tree to fetch the right object files
1626
* [#2006] Add conference in calltree
1627
* [#2006] Add Dbus signal conference added/removed/changed
1628
* [#2006] Add getConferenceDetails call on dbus
1629
* [#1904] Registration expire now appears as a spin box under gtk-ui.
1630
* [#812] Fixing a segmentation fault caused by a non-existing account
1632
* [#2006] Add getConfList method over dbus
1633
* [#2006] Add a conferencelist data structure in client-gnome
1634
* [#812] Defaults value are now sent if a non-existing account is
1636
* [#2006] Add sflphone action sflphone_join_participant
1637
* [#2006] Fix buffer read pointer problem deletion
1638
* [pjsip] Attempt at fixing via header incompatibility with
1640
* [#1797] forget something
1641
* [#2006] Add call new state conferencing in deamon
1642
* [#2006] Remove addParticipant method for conference, use
1643
joinParticipant only
1644
* [#1163] Update INSTALL documentation
1645
* [#812] Msec/sec values were not taken into account.
1646
* [#1797] Make pjproject-1.4 compile
1647
* [#2006] Add Detach participant method
1648
* [#2006] Dragndrop fully functional with INCOMING and HOLD call
1649
* [#1797] Add pjproject-1.4
1650
* [#1797] Remove pjproject-1.0.3
1651
* [#2006] Get call state in conference related function
1652
* [#2006] Add joinParticipant (conference) method in ManagerImpl
1653
* [#2006] Add joinConference DBUS message
1654
* [#2006] Store the previously selected call_id on dragndrop
1655
* [#2006] Fix GValue pointer unref in selection callback
1656
* [#2006] Store dragged call_id
1657
* [#2006] Update drag_data_received_cb callback to manipulate CallIDs
1658
* [#2006] Add dragndrop signals
1659
* [#2006] Set calltree reordable
1660
* [#812] Adds the ability to create a TLS listener in case the user
1662
* [#812] Adds the ability to configure local/published address from
1663
* [#1883] Move switchCall in onHoldCall function
1664
* [#812] Deals with the published address/port problem when
1666
* [#1883] Switch call id in managerimpl when peerHungUp
1667
* [#1883] Switch call id before hangup
1668
* [#1883] Add usefull and permanent debug info for conference
1670
* [#812] Fix various segmentation faults related to Direct IP kind of
1672
* [#1883] Fix deletion of std::map elements using iterators
1673
* [#2014] Add libzrtpcpp build dependency
1674
* [#1883] Still some for loop test ambiguity (while loop instead)
1675
* [#1883] Fix for loop initial test ambiguity (use while loop instead)
1676
* [#1883] We must discard data in urgent ring buffer if data is get in
1678
* [#1883] Fix availForGet same id for ringbuffer and readpointer
1679
* [#812] Match "sips" as a Direct IP Call when the user enter a sip
1681
* [#812] Fix segmentation fault related to SIP URI creation.
1682
* [#812] Towards integrating multiple tls listeners at the same time.
1684
* [#1883] Add debug messages in conference and fix mainbufferTest
1685
* [#812] gkt-ui fix. Private key must be fed as a filename and not as-
1687
* [#812] TLS integration within sipvoiplink and pjsip. Also,
1689
* [#1883] Fix Alsa/Pulse mallocation
1690
* [#1883] Fix data corruption in AudioRtp's micData buffer
1691
* [#812] Full dbus integration for all the tls related options under
1693
* [#1883] Fix memory leaks in audiortp session
1694
* [#1883] Fix mem leaks in audio rtp
1695
* [#812] Fix setAccountDetails where TLS_ENABLE was set to the value
1696
* [#812] Small gtk-ui fix.
1697
* [#811][#812] Small gtk-ui fix.
1698
* [#812] Introduced a mechanism for configuration files that makes
1700
* [#812] New dbus bindings added. Also, configuration compliance was
1702
* [#1881] Remove default buffer from MainBuffer (update unit-tests)
1703
* [#1881] Add ring buffer read pointer tests
1704
* [#1883] Fix issues in ringbuffer reader pointers
1705
* [#2034] Implementing a new configuration dialogue for TLS transport
1707
* [#1883] Add some usefull debug and safety checks
1708
* [#2028] Notify the client with libnotify when the zrtp negotiation
1710
* [#811] Harmless no to throw an exception, an makes the application
1712
* [#2028] A minidialog is showed to the user under sflphone-client-
1714
* Removed useless file.
1715
* Ignoring Makefile in src/widget
1716
* [#2027] Fix segmentation fault when showMessage callback is called
1718
* [#2026] keyExchange was set to ZRTP instead of "1"
1719
* [#2024] Fix the wrong summary at the end of the assistant.
1720
* [#1883] Fix mnagerimpl conference map insertion
1721
* [#1883] Add Mutexes in MainBuffer
1722
* [#811] Gtk ui was not presenting the right information about zrtp
1724
* [#2023] security icons were not installed in sflphone-client-gnome.
1725
* [#2021] Fix a mistake in the readme from sflphone-common that gives
1727
* [#811] The current SRTP mode was not properly displayed for the
1729
* [#1743] Re-implementation of the "automatically remove error dialogs
1731
* [#2017] [#2019] Fix the inability to dial a number and place a
1733
* [#811] Final re-integration of ZRTP support in the main branch from
1735
* [#1883] Fix map insertion methods
1736
* [#811] Combo box now is now set to the active key exchange method
1737
* [#811] ZRTP options now configurable back again from the Gtk UI.
1739
* Updated hostname for git clone
1740
* [#1883] Add minimal functionalities to create a conference
1741
* [#811] re-integration of all the methods and signals on dbus.
1743
* [#811] Got out of a precarious position were nothing would compile.
1744
* [#1976] Build documentation squeleton with docbook
1745
* [#1883] Add sflphone-client "addParticipant" button for conference
1746
* [#1994] Better organize the source directory structure. New
1748
* [#1883] Add a simple Conference class
1749
* [#1882] Use static audio buffer in Pulse and ALSA layer (instead of
1751
* [#811] First commit toward re-integration and refactoring of ZRTP
1752
* [#1882] Flush RTP ring buffer before entering mainloop
1753
* [#1882] Fixed MainBuffer::UnBinCallID() in case there is no
1755
* [#1882] Test (and fixe) high level conference and mixing
1757
* [#1772] Apply patch to compile on fedora (sent by Marcin
1758
Zajączkowski <mszpak@wp.pl>)
1759
* [#1882] Update Bind, unBind call_id in MainBuffer
1760
* [#1959] This adds the ability to store password as an MD5 Hash in
1762
* [#1538] Fixes rules compilation
1763
* [#1930][#1931] Fixed a mistake (again) related to index and
1765
* [#1753] Remove ILBC from pjproject - Hacks in pjsip
1766
* [#1930][#1931] Credential was not selected properly using realm
1767
* [#1882] Finilize multiple reading pointer in RingBuffer
1768
* [#1538] Remove configure from autogen.sh to respect debian upstream
1770
* [#1773] Remove generated files from repo
1771
* [#1791] Use XDG_CACHE_HOME to save pid file
1772
* [#1791] Fixes path to save history
1773
* [#1791] Fix debian installation scripts
1774
* [#1930][#1931] Settings are now taken into account in the server.
1775
* [#1882] Add ringbuffer default ring buffer pointer in methods
1777
* [#1882] Add default ringbuffer pointer
1778
* [#1882] Add RingBuffer multiple read pointer basic functionnalities
1779
* [#1882] Fix MainBuffer flushData unit test
1780
* [#1930][#1931] Ability to save and retreive the configuration from
1781
* [#1882] Added Multiple CallID mapping to MainBuffer
1783
* [#1791] If XDG env variables are not null but empty, use default
1785
* [#1791] Make XDG_CONFIG_HOME writable
1786
* [#1930][#1931] Partial commit. Not working yet. Cannot delete
1788
* [#1881] Fixed alsa capture latency problem
1789
* [#1881] Fixed Alsa capture temporarily
1790
* [#1930] [#1931] Partial unbroken commit providing the ability to
1791
* [#1881] MainBuffer implemented in AudioLayer/AudioRTP
1792
* [#1881] Add discard and flush unit-tests
1793
* [#1881] Add discard and flush functionnalites to MainRingBuffer
1794
* [#1881] Add availForGet in MainBuffer
1795
* [#1881] Add availForPut function to MainBuffer
1796
* [#1880] Remove AudioRTP* pointer from SipVoIP (reapered while
1798
* [#1881] Add a map between call id and coresponding ring buffer
1799
* [#1855] Refresh pot file and upload on Launchpad
1800
* [#1881] MainBuffe now robust to false ids on getData and putData
1801
* [#1881] Fix big big big memory leak
1802
* [#1881] Add getData and putData to mainBuffer
1803
* [#1881] Unit-test basic ring buffer functionnaities
1804
* [#1881] Add class MainBuffer and basic buffer creation unit-tests
1805
* [#1880] Fix call transfer (step2) issues
1806
* [#1880] Moved AudioRtp* pointer from SIPVoIPLink to SIPCall class
1807
* [#1791] Add postinst script to keep user data when migrating
1809
* [#1797] Make pjsip compile
1810
* [#1777] Code indentation
1811
* [#1791] Use XDG_DATA_HOME and XDG_CONFIG_HOME for sflphonedrc and
1812
history + unit tests
1813
* [#1746] Useless space does not appear anymore when volume sliders
1815
* [#1643] GtkCheckMenuItem is used instead of icons for elements in
1817
* [#1110] [#1668] STUN parameters are now located in the preferences,
1820
-- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 06 Nov 2009 11:23:15 -0500
1822
sflphone-common (0.9.6-SYSTEM) SYSTEM; urgency=low
1826
* Documentation on echo test
1827
* [redmine_down] codec names not displayed in total
1828
* [redmine_down] crash when hanging up a dialing call because tries to
1829
add it to history whereas no starttime
1830
* [#1927] alternate every time screen changed to call history
1831
* [#1886] clean code
1832
* [#1886] debug messages when loading history removed
1833
* [redmine_down] sflphone-kde icons
1834
* [#1855] Update language files
1835
* [#1502] Update version number
1836
* [redmine_down] setHistory at close
1837
* [#redmine_down] Handle PJ_DECLINE_SC as failure
1838
* [#1923] Fix segmentation fault when adding a new account
1839
* [#1923] Check on iterator before setting the config
1840
* [#1904] Added mnemonic to tabs in sflphone-client-gnome.
1841
* [#1905] The daemon was not sending the currentSelectedCodec signal
1842
on dbus when answering a call.
1843
* [#1922] Default values set to all account details
1844
* [#1886] Spinbox reg expire enables apply, and address book is not
1845
visible when disabled
1846
* [#1905] Bug fix for segmentation fault caused by an empty string,
1847
* [#1910] Warnings in test directory
1848
* [#1919] Error fixed
1849
* [#1855] Update russian translation - Hussein Abdallah
1850
* [#1910] Remove files
1852
* [#1777] Code indentation
1855
* [#1910] Remove warnings compilation in src
1856
* [#1886] removed AccountListModel in configskeleton
1858
* [#1911] check previous and new port
1859
* [#1910] Remove compilation warnings in src/dbus and src/history
1860
* [#1910] Remove compilation warnings in src/audio
1861
* [1855] Update german translation - Sven Werlen
1864
* [#1904] The registration expire value is now configurable from the
1865
* Cleaned up debug messages.
1866
* [#1886] separated initCallItem in two functions
1867
* [#1886] reversed error in commit
1868
* [#1886] clean debug
1869
* [#1886] changed Name of classes and files
1871
* [#1870] In call_state_cb (dbus.c:126), _time_stop was overridden by
1873
* [#1884] Added some new gpg flags to prevent tty warnings
1874
* [#1886] Clean audio config dialog
1875
* [#1886] No more compile warnings. + 1 comm
1876
* [#1872] Check if the user input is smaller than PJ_MAX_HOSTNAME.
1878
* [#1785] Fixed build when no new commit
1879
* [#1852] If chosen by the user, the hostname can now be solved and
1881
* [#1871] * and # inverted back
1882
* [#1869] Conditional compilation that checks if
1883
* [#1309] removed test in main
1884
* [#1425] Put actions in SFLPhone window class instead of ui view,
1885
made a separate toolbar for screens.
1887
-- SFLphone Automatic Build System <team@sflphone.org> Mon, 27 Jul 2009 09:53:00 -0400
1889
sflphone-common (0.9.6~rc2-SYSTEM) SYSTEM; urgency=low
1893
* [#1755] Remove generated file
1894
* [#1753] restore ilbc ...
1895
* [#1866] Methods getSipPort and setSipPort now have an effect on the
1896
* [#1753] make pjsip compile without ilbc. Use ./autogen.sh --disable-
1898
* [#1855] Fix error in russian translation
1899
* [#1805] Remove the old flawed signal mechanism which was failing in
1900
* [#1855] Refresh translation
1901
* Spanish translation finished + po README files updated + echo's in
1903
* [#1850] Yun made the chinese HK-CN translation
1904
* [#1848] Fix transfer interface bug
1905
* [#1862] At install, kde client installs only french translation file
1906
* [#1841] A new fallback mechanism was added to the internal resolver
1908
* Started AccountList model/view
1909
* [#1855] Remove po subdir in Makefile.am
1910
* [#1855] Fix typo error in sflphone-client-gnome
1911
* [#1855] Do not generate Makefile in sflphone-common/po
1912
* [#1855] Copy translation files into both clients dirs
1913
* [#1855] Remove po dir from sflphone-common
1915
* [#1860] mailbox->voicemail...
1916
* make scripts executable
1917
* [#1855] French translation
1918
* [#1855] Chinese zh_HK partially filled...
1919
* [#1859] An unnamed pipe monitored by poll() was added. When we want
1921
* [#1855] Sven completed the first part of the german translation
1922
* [#1855] Cantonese manually filled for already translated, almost
1924
* [#1855] Merge russian translation
1925
* [#1855] Spanish manually filled for already translated, almost equal
1927
* [#1855] Update german translation in ./lang/de
1928
* [#1858] This problem was fixed by removing a useless line in
1929
* [#1855] merged existing translations in lang/ sflphone.po's
1930
* [#1842] [#1843] An attempt at improving the expected behaviour that
1932
* [#1855] added po folder in gnome client and scripts for copying from
1933
common lang folder to clients
1934
* [#1853] Edit before call does nothing on call history
1935
* Put most language entries possible in common. From 300 to 250
1936
entries. Stays underscores problem. Scripts for copy in clients.
1937
* commit to merge master
1938
* [#1825] Changed "Bad authentification" to "Authentication Failed".
1940
* [#1753] Remove ILBC from pjproject
1942
-- SFLphone Automatic Build System <team@sflphone.org> Fri, 17 Jul 2009 19:12:44 -0400
1944
sflphone-common (0.9.6~rc1-SYSTEM) SYSTEM; urgency=low
1948
* Update some version number
1949
* [#1792] Creates .sflphone directory with permission 600. Also,
1951
* [#1810] GUI is now notified that the call failed. Also, a segfault
1953
* [#1816] Address book search disabled when disabled address book and
1954
enabled it back plus button stays triggered
1955
* codeclistmodel + asynchronous loading of address book +
1956
enable/disable address book
1957
* [#1810] Now checking SDP answer after 200 OK. Still need to
1959
* [#1794] Can't use the interface during a call
1960
* Updated translation files
1961
* Russian translation integrated
1962
* Codec list model/view started.
1963
* [#1807] Add configure.ac in pjproject-1.0.3
1964
* [#1787] closeRtpSession added in some places where it should have
1966
* Use Item class for contacts and accounts
1967
* Comments + clean code
1968
* [#1794] Improved debug messages
1969
* [#1805] Replaced the old and unreliable mecanism that was was
1971
* [#1794] Can't use the interface during a call
1972
* [#1787] For those cases where no registered SIP account is
1974
* [#1797] Make pjsip compile
1975
* [#1787] Minor changes. Removed useless commented line. Changed order
1977
* [#1777] Code indentation
1978
* [#1797] Update package generation with new pjsip version
1979
* [#1798] Does not hang up when the call is building up
1980
* [#1797] Update .gitignore with new pjsip version
1981
* [#1797] Remove generated files from repo
1982
* [#1797] Main build system now uses pjproject-1.0.3
1983
* [#1797] Add pjproject-1.0.3
1984
* [#1797] Remove pjproject-1.0.2
1985
* [#1796] Computing time optimization (samplerate conversion)
1986
* [#1787] _audiortp->start() moved away from offhold(),
1988
* [#1312] Added new states for calls initialized by other clients
1989
* [#1795] Crashes when adding a new account, checking it and applying
1990
* [#1782] Missing icons
1991
* [#1793] KDE client compilation problem
1992
* Fake ringtone files can no longer be set.
1994
* [#1312] Able to fetch to differentiate incoming/ringing call state
1995
* [#1784] Use DESTDIR variable in po Makefile - fix language file
1997
* [#1785] Fixed typo
1998
* [#1785] Fixed changelog update
1999
* [#1759] ./autogen.sh --prefix=/usr --with-debug to use optimization
2001
* [#1773] Changed snapshot naming convention
2002
* [#1773] Removed gpg agent use, added repository cache cleaning
2003
* [#1759] Use optimization level 0 for repository, 2 for packages
2004
* [#1777] Code indentation/formatting
2005
* Translated new features in french
2006
* [#1785] Added missing changelog entry
2007
* [#1781] Window title is SFLPhone
2008
* [#1777] Add code indentation/formatting in the buil system
2009
* [#1774] Can't set voicemail number in KDE account creation wizard
2010
* [#1775] Can't modify account information for account created with
2012
* [#1771] Add a "Default" button in context menu to disable chosen
2015
* [#1224] Remove generated file from the repo
2016
* [#1224] Remove generated file from the repo
2017
* [#1762] distclean target should remove kconfig generated files
2018
(settings.h, settings.cpp). Rename them?
2019
* [#1761] clear history button should really clear history
2021
* Implemented Dialpad widget instead of building it in main view.
2022
* Removed last occurence of the old config dialog, that made the build
2024
* [#1755] Do not consider G722 as a dynamic payload elsewhere than in
2026
* [#1753] Remove ilbc Makefile generation
2027
* [#1756] Implement a kde configuration dialog with kconfig xt and
2029
* [#1755] fix audiocodec folder parsing problem
2030
* [#1450] Reinit timestamp comparison in RTP, create session in
2032
* [#1753] Remove milenage third party code from pjsip
2033
* New Config Dialog integrated in GUI.(without codecs)
2034
* [#1753] Remove ILBC codec
2035
* kconfig started, tr2i18n -> i18n, icons folder, accountList changed
2036
* [#1705] Fixed Audio RTP thread creation/start
2037
* [#1714] Fix codec negociation result handling
2038
* [#1678] Fix audiortp payload setting
2039
* [#1678] Put bac putData method in rtp
2040
* [#1669] gtk_file_chooser_get_filename() support UTF-8 by default
2041
* [#1735] Add conditions to sdp update call if call declined
2042
* [#1737] substr of recordings destination folder to remove "file://"
2043
should be done in client rather than in daemon
2044
* [#1731] Enlarge audio stream buffer size
2045
* [#1714] Missing true
2046
* [#1317] Fixed Mandriva timeout
2047
* [#1317] Changed tag convention
2048
* [#1317] Cleaned git-dch
2050
-- SFLphone Automatic Build System <team@sflphone.org> Fri, 10 Jul 2009 15:49:56 -0400
2052
sflphone-common (0.9.6~beta-SYSTEM) SYSTEM; urgency=low
2056
* spec files for mandriva and opensuse updated with buildrequires
2058
* [#1700] Cannot build on ubuntu 8.10 and a few other distribs
2059
* [#1502] Update version number where applicable
2060
* [#1642] Update client icons
2061
* [#1450] Clean up useless debug and comments in sipvoiplink and
2063
* [#1450] Remove Semaphore object in AudioRtp thread deletion
2064
* [#1450] Audio RTP init now synchronized with Sip/SDP
2065
* [#1693] kde client crashes when changing codecs order/activation
2066
* [#1450] Deep refactoring of audiortp
2067
* [#1450] setRtpSessionRemoteIp
2068
* [#1689] getCallList at start
2069
* [#1224] Change path in package files
2070
* [#1450] Audio RTP initialized only once, payload and remote ip set
2072
* [#1450] Add setRtpSessionMedia and setRtpSessionRemoteIp address
2073
* [#1642] Make GNOME GUI fresher and younger ;)
2074
* [#1686] Status bar displaying used account
2075
* added sflphone-kde icon so that it compiles
2076
* [#1659] Ending a call causes the daemon to crash
2077
* corrected introspection XMLs, po files...
2078
* [#1211] g722 media descriptor in codecDescriptor
2079
* [#1310] Install sflphoned in $(prefix)/lib/sflphone
2080
* [#1502] Do not install test binaries and dbus utilitaries
2081
* [#1224] hack for pjsip build system!
2082
* [#1224] Remove pjsip binaries from repo
2083
* [#1224] Upgrade to pjsip 1.0.2
2084
* [#1658] About SFLphone (bugs)
2085
* [#1658] About SFLphone
2086
* [#1660] Displaying all dialed numbers in a call
2087
* Tested status bar.
2088
* [#790] Optimize pulse audio streams parameters
2089
* [#1678] Some usefull debug messages for mutex/semaphore deadlock
2091
* [#1669] Add/remove some usefull/unusefull debug
2092
* [#1665] Fix latency related to pulse audio stream openning/closing
2093
* [#1457] Make the menus and panels accessible in french
2094
* [#1457] Improve broken keyboard accessibility in menus and conf
2096
* [#961] Instanciate only once the searchbar icons
2097
* [#961] Restore transfer fonction
2098
* [#961] Filter on the history type OK
2099
* [#961] Fix compilation problems on hardy/intrepid
2100
* [#1157] Commit missing files
2101
* [#790] Reduce number of start/stop streams call on pulse audio
2102
* [#1639] kde client crashes when no account registered
2103
* [#1620] Fix the searchbar
2104
* [#1620] Get back caltree as it was during gtkcritical area
2105
* [#1620] Add history filter reinit function
2106
* [#1335] Add a missing label in address book preferences
2107
* [#1561] Update russian translation - Hussein Abdallah
2108
* [#1605] Fix edit menu french translation
2109
* [#961] Enable to search in the history according to the call type
2110
* [#1449] Searchbar does not work anymore
2111
* [#961] Add popup menu on the entry primary icon for history
2112
* [#1317] Fixed KDE client package dependency
2113
* [#936] speex 32 khz integration completed
2114
* [#936] Use 320 frame size
2115
* [#936] Test using a frame size at 320 smpls
2116
* [#1214] Enable / Disable history
2117
* [#1607] Fix compilation problem for ubuntu 8.10 (libsexy)
2118
* [#1313] Implement processDataEncode processDataDecode in audiortp
2119
* [#1613] codec list order can't be set
2120
* Better handling of localisation + added languages + corrected
2121
warnings + begginning of new config dialog with kconfig + 14px
2123
* [#1214] Save and load history according to the limit timestamp +
2125
* [1609] Fix call number copy/paste feature
2126
* [1607] Restore clear action icon in searchbar
2127
* [#936] Try to decode using 1280 samples
2128
* [#936] Add some debug
2129
* [#936] Add .cpp file
2130
* [#936] Oops Forgot speex 32 khz
2131
* [#1214] Add configuration panel for history + D-Bus calls
2132
* [#1313] Test rtp thread function, frame size, nbbytes, resampling
2133
* [#790] Flush audio data before closing audio streams
2134
* [#1214] History displays local time
2135
* [#1214] Skip empty field on display
2136
* [#1214] Associate an account to an history entry
2137
* [#1342] Get addressbook options sensitive/non-sensitive
2138
* [#1211] Clean up and comments
2139
* [#1211] Get back to 20 ms framesize
2140
* [#1211] Use sendImmediate instead of putData in RTP
2141
* [#1211] Fix nb byte available in RTP
2142
* [#1211] Clear condition on maxNbSamples in RTP
2143
* [#1211] Fix max byte available in RTP session
2144
* [#1211] G722: Use 160 samples per frame instead of 320
2145
* [#1211] Test using a dynamic payload
2146
* [#1211] Test using a dynamic payload type
2147
* [#1211] Rename size variable (nb_samples, nb_bytes)
2148
* [#1211] Test g722 ip-to-ip sending twice the data lenth
2149
* [#1211] Test g722 ip-to-ip
2150
* [#1214] Do not select an history item by default at startup
2151
* [#1214] Remove some compilation warnings
2152
* [#1214] Handle empty field - remove g_print
2153
* [#1214] Add each history item only once
2154
* [#1214] Handle call timestamps properlier
2155
* [#1214] Do not need timestamp files anymore
2156
* [#1214] Use the saved date for history entry
2158
* [#1214] Client doesn't crash if the D-Bus call fails
2159
* [#1214] Client is able to save its history - still some glitches
2160
* [#1211] Forgot 16000 for g722
2161
* [#1211] G722 initialization
2162
* [#1214] Save name/number, successfully load the history if no fields
2164
* [#1499] Fixed destination directory bug
2165
* [#1214] Restore all the functionalities; peer name/number way more
2167
* [#1214] Add callable_object instead of call_t, refactoring
2168
* [#1211] Test with polycom soundstation 16000
2169
* [#1211] Remove C like inline function in g722 codec
2170
* [#1342] Finalize gnome client preference window formating
2171
* [#1214] Retrieve the history when the gnome client startsup
2172
* [#1306] Implement localization for KDE client
2173
* [#1593] enable accounts apply button when account checked/unchecked
2174
* [#1214] Implement the dbus calls on server side
2175
* [#1214] Add serialized/unserialized functions to pass data on DBUS
2176
* [#1342] Formating gnome client configuration windows
2177
* [#1214] Save sucessfully a map of history items
2178
* [#1499] Removed multiple jobs compilation for KDE client (2)
2179
* [#1214] Load history from file into memory, add unit tests
2180
* [#1534] Throws a length_error exception in case URL exceeds
2181
std::string max_size
2182
* [#1499] Removed multiple jobs compilation for KDE client
2183
* [#1565] make account leds smaller
2184
* [1430] Fix dbus debug
2185
* [#1562] crashes when trying to change item of a call of state "OVER"
2186
* [#1116] Fix compilation bug
2187
* [#1317] Added mandriva and opensuse-11 64 bits
2188
* [#1108] Add messges in main window concerning transfer success
2190
* [#1116] Fix compilation problems
2191
* [#1211] g722 Makefile
2192
* [#1108] Client side transferFailed/trasferSucceded signals handling
2193
* [#1211] G722 mostly completed,
2194
* [#1555] make bigger toolbar (24x24)
2195
* [#1551] remove default mailbox number in wizard and disable mailbox
2196
button when first account doesn't have mailbox number
2197
* [#1342] Re-add sflphone manpages
2198
* [#1116] Fix compilation on non-jaunty distros
2199
* [#1317] Fixed opensuse startup sleep
2200
* [#1108] Add a signal in the client to notify successful or failed
2202
* [#1108] Dbus signals concerning call transfer success/failure
2203
* [#1317] Added opensuse to automatic build system
2204
* [#1223] Fix manpages bug
2205
* [#1060] german translation glitch
2206
* Clean up some gnome client warnings
2207
* [#1547] replace ugly account leds by beautiful icons
2208
* [#1548] add close button that hides windowand just hide on clicking
2210
* [#1549] put introspec XMLs in the client's source
2211
* [#1312] Implement getCallList D-BUS method
2212
* [#1116] Clear text in history and contacts
2213
* [#1499] KDE integration
2214
* [#1469] Modify header linkers in dbus-c++'s Makefile.am's
2215
* [#1469] Remove examples folder from dbus-c++
2216
* [#1214] History integration in build system; unit test squeleton
2218
* [#1469] Remove configure stuff in dbus-c++
2219
* [#1469] Add unofficial mainline dbus-c++
2220
* [#1469] Remove dbus-c++ from freedesktop
2221
* [#1430] Bring account changed signal/callback back to normal
2222
* [#1060] Update german translation - Sven Werlen
2223
* [#1430] Add marshaller one string define
2224
* [#1430] Send account change signal broadcast using account id
2225
* [#1430] Remove condition on setRegistrationState, cause stun to
2227
* [#1317] Centralized version handling
2228
* [#1317] Fixed version number on sfl-git-dch
2229
* [#1317] Refactoring for new distributions
2230
* [#1215] Fix account order at startup if latency
2231
* [#1088] Restore sip dns srv
2232
* [#1214] Add squeleton for history manager
2233
* [#1430] Add accout id to accout changed method
2234
* [#1430] No connectionStatusNotification (account changed) if no
2236
* [#1538] Add COPYING file
2237
* [#1430] Add audio rtp thread tests
2238
* [#1317] Changed version detection
2239
* [#1538] Document license in libs/stund
2240
* [#1317] Added version files
2241
* [#1538] Apply François patches - debian packages
2242
* [#1317] Updated spec files
2244
* [#1538] Apply François patches - debian packages
2245
* [#1535] Change program file structure (directory src...)
2246
* [#1317] Updated build system scripts
2248
* [#1317] Copied introspect files to gnome client
2249
* [#1317] Added opensuse to build-system : first-shot
2250
* [#1317] Remove spec files from configure
2251
* [#1317] Added missing prefix
2252
* removed debug for daemon account fix
2253
* [#1430] Add a connection reference which most likely belong to
2255
* [#1430] Use shared connection instead of private
2256
* make daemon find the account, added userMatch
2257
* Clean code, add comments...
2258
* [#1317] Fixed packaging rules
2259
* [#1317] Updated autogen
2260
* Updated autogen.sh for pjsip
2261
* [#1526] Set accounts order
2262
* [#1317] Fixed pjsip lib dirs
2263
* [#1317] Updated debian packaging for new pjsip configuration script
2264
* [#1317] Switch to autogenerated guess and sub files
2265
* [#1317] Updated pjsip inclusion in build system
2266
* [#1317] Replaced pjsip guess and sub files
2267
* [#1317] Fixed compilation issues on opensuse 11
2268
* [#1505] account list seem to crash the application when clicking
2270
* [#1456] Add a flag to be replaced in the control files
2271
* [#1456] Added version dependancy handling
2272
* put account alias in AccountWidgetItem rather than in the item with
2274
* [#1034] The KDE client should start sflphoned if it is not started
2275
* [#1500] Handle options for notifications and display on incoming
2277
* [#1443] Client should not crash when receive an unexpected
2279
* [#1403] Do not stop the notification anymore
2280
* [#1456] Added version dependancy handling
2281
* [#1426] Daemon crashes when get alsa plugin
2282
* [#1422] Improved error messages
2284
* [#1424] Change logo in tray icon and put a different one when
2286
* [#1425] first part done, window title...
2287
* [#1413] add manpages creating and installing in build system
2288
* [#1417] The client should start the account creation wizard if
2289
started for the first time (if config file doesn't exist)
2290
* [#1421] Make volume bars horizontal when dialpad is hidden.
2291
* Changed main window title and fixed a mistake in sflphone_const.h
2292
* [#1412] make debian package building work
2293
* changelog changed.
2294
* Changed addAccount method in gnome client.
2295
* Debian and man folders added.
2296
* [#1388] Change project name from sflphone_kde to sflphone-client-kde
2297
* Better handle of kabc check.
2298
* [#1351] Automatic generation of dbus interfaces in makefile
2300
* [#1307] Implement "edit before call" in history and address book.
2301
* [#1344] change action_call label in call history from "call" to
2303
* [#1308] Implement Hook feature in kde client
2304
* Improved build system.
2305
* #1219 : Add address book configuration page
2306
* Better handling of registration to the daemon.
2307
* #1039 : Add tray icon in kde.
2308
* Issue no 1216 : Double click on item in history or address book
2310
* display peer name in call list and call history when called from
2312
* Address book functionnal with photo displayed.
2313
* Help menu kde available but actions disappeared. All fonctions in
2315
* Address book functionnal but ugly and making its own sort in the
2316
complete address book.
2317
* Account choice on right click, clean out includes, page address
2319
* Wizard, double click, context menu...
2320
* Removed sflphone_kde.kdevelop.filelist
2321
* Added account creation wizard and translated interface in english.
2322
* Transfer functionnal but ugly.
2323
* transfer not functionnal
2324
* Bug fixed : unholding (UNHOLD_CURRENT, UNHOLD_RECORD)
2325
* Commit functional for push. With install.sh
2327
* Problem with enable accounts. Account display increased.
2328
* Functional with codec order working , playDTMF.
2329
* Commit functional.
2330
* sflphone_kde/build added in .gitignore.
2331
* complete commit for checkout previous.
2332
* Commit before checkout previous version to check the display
2333
bug(little font everywhere...)
2334
* Functionnal client. Rest : history icons, config icons and
2336
* commit before merge asavard for isRecording.
2337
* Call and Automate fusion done and seems to work.
2338
* Commiting before putting Automate class in Call class.
2339
* Functionnal main window without recording, history, voicemail, kio
2341
* client kde avec kdevelop.
2342
* Config Dialog almost finished.
2345
-- SFLphone Automatic Build System <team@sflphone.org> Tue, 23 Jun 2009 11:12:06 -0400
2347
sflphone-common (0.9.5-SYSTEM) SYSTEM; urgency=low
2351
* [#1060] FIx bug in chinese translation
2352
* [#1313] git add rtpTest.cpp rtpTest.h
2353
* [#1313] Add init/close rtp tests
2354
* [#1313] Basic instanciation of the rtp layer
2355
* [#1449] Gtk-Critical concerning history filters and new calls
2356
* [#1400] Make the match with the hostname instead of username
2357
* [#1324] Change status bar label for "Using %s (%s)"
2358
* [#1403] Icon size: 60x60 px
2359
* [#1403] Do not remove notification, improve icon quality
2360
* [#1403] Add smaller icon for gnome notifications
2361
* [#1403] Prevent crash when hangup && no notification
2362
* [#1403] Remove all actions on notifications; code refactoring
2363
* [#1451] Use stun.sflphone.org as default STUN server
2364
* [#1060] New po files - need to be translated
2365
* [#1060] Update french translation - Rebuild template file
2366
* [#1456] Add a flag to be replaced in the control files
2367
* [#1454] Make cppunit optional; remove from build deps in control
2369
* [#1401] Add libexpat1-dev dependency in control files
2370
* [#1448] Take off these ugly debug messages
2371
* [#1448] fixed getTelephoneTone and getTelephoneFile() called
2373
* [#1406] add liblog4c-dev in build-depends
2374
* [#1409] Restore .desktop icon
2376
-- SFLphone Automatic Build System <team@sflphone.org> Mon, 25 May 2009 11:34:40 -0400
2378
sflphone-common (0.9.5-SYSTEM~rc2) SYSTEM; urgency=low
2382
* [#1422] Improved error message
2383
* [#1402] Fix pjsip build
2384
* [#1404] Clear GTK-Critical Bug at client startup
2385
* [#1422] Added automatic VM shutdown when building on more than one
2387
* [#1422] Fixed some issues with new changelog generation script
2388
* [#1422] Moved distribution update to specific file
2389
* [#1422] Dropped git-dch, replace by home made implementation
2390
* [#1402] Fix pjsip build
2391
* [#1404] Clear GTK-Critical Bug at client startup
2392
* Changes for name based dbus connection
2394
* [#1343] Gnome: Implement a callback system to handle focus on
2397
* Refactoring Python code, PEP8
2398
* [#1430] Get back dbus_g_proxy_new_for_name
2399
* [#1430] Get back DBUS_BUS_SESSION type
2400
* [#1430] Dbus fixed owner message binding
2401
* Second test with DBUS owner
2402
* [#1404] Gnome -> Preferences -> Hooks
2403
* [#1404] Gnome -> Preferences -> Recordings
2404
* [#1404] Call History
2405
* [#1404] Gnome -> Preferences -> Address Book
2406
* [#1404] IF the first notification option disable the second
2408
* Dbus with fixed owner does not automatically start the deamon
2409
* Add codec debug tests in pysflphone
2410
* [#1407] Some print info
2411
* [#1407] Add a scenario to pick_up action
2412
* Test client dbus connection to a fixed owner
2413
* Add python dbus test suite
2414
* [#1161] Modified version handling in build system
2415
* [#1314] Test pulse audio and audio streams connect and disconnect
2416
* [#1402] Add info message after configure
2417
* [#1402] Build the daemon with the local pjsip library (vs the
2419
* [#1009] Fix Codec Sampling Rate set to zeros
2420
* [#1314] Add mutex to pulse layer audio streams
2421
* [#1314] Refactoring pulseaudio stream to test connect disconnect
2422
* [#1314] Refactoring of pulselayer to test conect/disconnect
2423
* Add debug messages in debus calls concerning account
2424
* [#1314] Add some return values to audio init functions
2425
* [#1406] add liblog4c-dev in build-depends
2426
* [#1409] Restore .desktop icon
2427
* Bug #1405: Fix strings as requested.
2428
* Bug #1404: Fix strings in preferences panel.
2430
-- SFLphone Automatic Build System <team@sflphone.org> Tue, 19 May 2009 12:08:03 -0400
2432
sflphone-common (0.9.5-0ubuntu1~rc1) SYSTEM; urgency=low
2434
[ SFLphone Project ]
2435
* [#1262] Updated changelogs for version 0.9.5-0ubuntu1 Snapshot 2009-
2439
* Add some python CLI client code; not really functional
2440
* [#1108] Fix peerHungup method for IP to IP call
2442
[ Alexandre Savard ]
2443
* [#1108] Correct setting of SIP contact for direct IP call
2444
* [#1108] SIP user agent handles incoming REFER
2447
* Remove website from repository
2448
* Update translation
2450
[ Alexandre Savard ]
2451
* Sflphone icon's tooltip changed for "configured" instead of
2455
* Update translation
2457
[ Sflphone Project ]
2459
-- Sflphone Project <sflphone@mtl.savoirfairelinux.net> Tue, 05 May 2009 19:16:09 -0400
2461
sflphone-common (0.9.5-0ubuntu1~beta) SYSTEM; urgency=low
2464
* Updated Eclipse stuff
2465
* Improved addressbook config window
2466
* Added sflphone Eclipse stuff
2467
* Implemented addressbook list server side
2468
* Moved dbus stuff in dbus directory
2469
* Updated addressbook configuration
2472
* Remove unuseful installation scripts. Use apt-get build-dep sflphone
2476
[ Alexandre Savard ]
2477
* defining speex 16khz
2480
* Remove unuseful file from build system
2481
* Start dns srv resolver
2483
[ Alexandre Savard ]
2484
* Basic ogg/vorbis initialization
2487
* Handle incoming IP-to-IP invite correctly
2489
[ Alexandre Savard ]
2490
* speex wideband 16000
2493
* Better handling of incoming IP to IP call
2494
* DNS SRV resolution functional
2495
* Implement IAX2 incoming URL
2496
* Allow user to make IP call without any accounts configured
2497
* Add a contextual menu to edit a number from the contacts tab
2498
* Add comments, tooltip and new button to the contextual menu
2499
* add delete event, migrate to GTK 2.16 for sexy icons
2500
* Resolve ticket #1118
2501
* Update suse spec file
2502
* Add phone number cleanup functions, unit tests and panel
2504
* Add pertinent test that fails
2505
* fix dependencies for suse package
2506
* Add contextual edit menu in history - #1120
2508
[ Alexandre Savard ]
2509
* Temporary comit: make speex wideband (16 khz)
2510
* Temporary: shared object for speex narrow band
2511
* Temporary: speex narrowband and wideband coexist
2514
* Fixed bug when no book selected
2515
* Fixed addressbook related compilation warnings
2516
* Fixed GTK client remaining compilation warnings
2517
* Fixed segfault when book removed since last sflphone run
2518
* Fixed bug when book is unreachable (ldap error)
2520
[ Alexandre Savard ]
2521
* Fix codec list in audio config window
2522
* Active/inactive speex codec by payload
2526
* Added some comments
2529
* Add callto: handler script for browsers and al.
2530
* Integrate test compilation in the daemon build-system
2533
* Fixed g_object_unref warning for pixbuf
2534
* Cleaned too verbose output
2535
* Fixed toolbar update warning
2536
* Added support for asynchornous books open (first shot)
2539
* Add a DBus call to fetch the call details from a call ID - Ticket
2543
* Improved async open books
2547
* Add a way to save account order
2548
* commit missing files
2551
* Introduced log4c (ticket #1162)
2554
* Load/save account order functionnal - ticket #813
2556
[ Alexandre Savard ]
2557
* Add CELT codec (#1143)
2558
* Make celt frame size 256 (*1143)
2561
* Switched everything to log4c (ticket #1162)
2562
* Updated eclipse settings
2565
* Restore adding account - ticket #1172
2566
* Add liblog4c dependecy - ticket #1179
2568
[ Alexandre Savard ]
2569
* Double maxAvailByte for frame size in rtp (#1143)
2572
* Add User-Agent SIP header - Ticket #1173
2575
* Fixed autoresize issue (#708)
2578
* Remove libcppuint dependency for the debian packages
2579
* Look for libsexy only if gtk version < 2.16 - Ticket #1116
2580
* Remove libsexy dependency for jaunty. ticket #1116
2583
* Introduced unit tests (#1146)
2585
* Fixed Makefile (#1146)
2588
* [TICKET #1112] Add a test on the voice buffer to send through iax
2590
* Remove doublon in dependencies
2591
* Remove warnings from the client test framework
2592
* Update version number to 0.9.5~beta
2593
* Update build-package script
2594
* Add check dependency in build-deps control file field
2595
* Create debian files for the new sflphone-client-gnome
2596
* [TICKET #1212] Add Replaces field in control files
2597
* [TICKET #1212] Fix manpages installation path
2598
* [TICKET #1212] Add maintainer scripts to create alternatives
2599
* [#1212] Update the manpages generation - edit preinst maintainer
2601
* [#1212] Fix reference error in manpage
2602
* [#1212] Add missing files on the client side
2603
* [#1212] Fix debian docs files - no TODO file
2604
* [1212] Fix manpage creation problem
2605
* [#1220] Generate client-side glue files and marshaller at
2607
* [#1220] Generate server-side glue files at compilation time
2608
* [#1212] Change binary name to sflphone-client-gnome
2609
* [#1212] Update .gitignore to fit the new working tree
2610
* [#1220] Explicitly generate glue files before building the library
2611
* [#1220] Compile dbus directory before audio
2612
* [#1212] Create sflphone-common at the root of the repository
2613
* [#1212] Re-add pjproject
2614
* [#1212] Remove Makefile from repo
2615
* [#1220] Fix Makefile.am
2616
* [#1212] New working directory functional
2617
* [#1212] Update .gitignore
2618
* [#1212] Hack to make pjsip compile..
2619
* [#1220] Use non-installed binary for dbusxx-xml2cpp
2620
* [#1212] Add descriptive files, remove unuseful scripts from tools/
2622
[ Alexandre Savard ]
2623
* Restore speex codecs
2624
* add frame size for celt (#1143)
2625
* add framesize to codec, independant from audiolayer (#1143)
2626
* use codec frame size in rtp (#1143)
2627
* compute fixed_codec_framesize (#1143)
2628
* do not resample if not required (#1143)
2629
* add condition on resampling for decoder (#1143)
2630
* add a condition on bytesAvail == 0 from mic data
2631
* no maximum in rtp decode (#1143)
2632
* compute maximum for decoding (#1143)
2635
* [#1146] Implement unitary tests on the client-side
2637
[ Alexandre Savard ]
2638
* use float instead of int to compute max nb of sample (#1143)
2639
* add nbSampleMax for unresampled data (#1143)
2640
* make thread sleep during 5 ms insead of 20 (#1143)
2641
* use unix usleep (#1143)
2642
* 50 usecond thread!!!!! (#1143)
2643
* try with the smallest compression (#1143)
2644
* use timer set at framesize (#1143)
2647
* [#1161] Restore changelog version
2649
[ Alexandre Savard ]
2653
* [#1161] Update changelog
2654
* [#1220] Add Conflicts: sflphone in debian control files
2655
* [#1179] Add liblog4c3 runtime dependency
2656
* [#1212] FIx typo error in dependency list for itnrepid
2657
* [#1212] FIx .desktop file to point on the right exec
2658
* [#1212] Modify changelog replacing tag
2660
[ Sflphone Project ]
2661
* "[#1262] Updated changelogs for version 0.9.5-0ubuntu1~beta"
2664
* [#1212] restore changelogs
2666
[ Sflphone Project ]
2667
* [#1262] Updated changelogs for version 0.9.5-0ubuntu1 Snapshot 2009-
2671
* [#1212] restore changelogs
2673
[ Sflphone Project ]
2674
* [#1262] Updated changelogs for version 0.9.5-0ubuntu1~beta
2677
* [#1212] restore changelogs
2679
[ Sflphone Project ]
2681
-- Sflphone Project <sflphone@mtl.savoirfairelinux.net> Mon, 27 Apr 2009 16:57:00 -0400
2683
sflphone-common (0.9.4-0ubuntu2) SYSTEM; urgency=low
2685
[ Alexandre Savard ]
2686
* Restore speex and GSM detection
2691
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 8 Apr 2009 11:29:15 -0500
2693
sflphone (0.9.4-0ubuntu1) SYSTEM; urgency=low
2696
* Integrate DBus-c++ and libiax2 in the main build system
2697
* Clean up in the working repository
2698
* Reorder hooks configuration panel
2699
* Protect case when no codecs are active
2700
* Fix some return values
2701
* Add unitary tests for the hook manager (premisces)
2704
* Update chinese translation
2707
* Update german translation
2710
* Update russian translation
2713
* Update spanish translation
2715
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 3 Apr 2009 18:29:15 -0500
2718
sflphone (0.9.4-rc1) SYSTEM; urgency=low
2721
* Fix bug while trying to hold/unhold several simultaneous call
2722
* Improve address book build system
2723
* Implement SIP url popup on incoming call
2724
* Improve GTK+ panel configuration
2726
* GTK+ client refactoring
2728
* Address book improvment
2730
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 27 Mar 2009 18:29:15 -0500
2732
sflphone (0.9.4-0beta1) SYSTEM; urgency=low
2734
[ Alexandre Savard ]
2735
* Display codec used during conversation on the GUI
2736
* Enable/disable STUN parameters at runtime
2737
* Refactor search bar use
2739
* Build system fixes
2740
* Implement SIP re-invite
2741
* Implement IP to IP call
2743
* Integrate GNOME address book based on evolution data server
2745
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 20 Mar 2009 18:29:15 -0500
2748
sflphone (0.9.3-0ubuntu3) SYSTEM; urgency=low
2750
[ Alexandre Savard ]
2751
* Both playback and record streams in PA_STREAM_CORKED (pulseaudio)
2752
* Use PLUGHW device for ALSA capture
2753
* Functional IAX and SIP recording for voicemail
2754
* Use the less CPU-consuming interpolator algorithm for resampling
2755
* Display in GTK GUI the codec used in conversation
2756
* GTK GUI use ASCII instread of utf-8
2757
* Add record menus in GTK GUI
2758
* Put on hold when dialing a new number
2759
* AccountID's are saved in the history
2762
* Integrate DBUS C++, libiax2 in the git repository
2764
* Use libspeexdsp only if available on the system
2765
* Updated .gitignore file
2768
* Account assistant manager improvment
2769
* Add an email request when creating a new account to receive voicemails
2771
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Sat, 14 Feb 2009 13:29:15 -0500
2773
sflphone (0.9.3-0ubuntu2) SYSTEM; urgency=low
2776
* Add compilation note in README
2777
* Use default ALSA plugin for capture
2778
* Fix the ALSA capture problem one more time
2779
* Clean up debug messages in dbus.c
2780
* Add libspeexdsp dependency
2781
* Remove implicit declaration compilation warnings
2782
* Fix links in the website, add release note
2783
* Change capture for the website front page
2784
* Add alsa devel dependency in build-depends control file field
2785
* Clean up, indentation, try to handle latency problems in iax/pulseaudio
2786
* Remove pjsip generated files from the repo
2787
* Use the previous declared curAlias function in accountwindow
2788
* Fix bug in history call duration when the call fails
2789
* Remove runtime warning in the GTK+ client
2790
* Add librsvg2-common dependency to load SVG under KDE
2791
* Refresh .gitignore
2792
* Update locales files + french translation
2793
* Add configuration panel for future noise reduction
2794
* Add configuration panel for audio record module
2795
* Daemon less verbose; accounts don't try to access STUn options anymore
2796
* Fix typo in configwindow
2797
* Add content in the official website
2798
* use a GTK_STOCK icon for the record button
2799
* Complete description text in the assistant manager
2800
* Add libtool flags in client configure.ac
2801
* Remove unuseful dependency (snd)
2802
* Fix SIP transfer problems
2803
* Remove previous version of PJSIP from the repo
2804
* Upgrade PJSIP to version 1.0.1
2805
* Add the new website source in the repository
2806
* Use libspeexdsp for silence detection only if available
2808
[ Loïc Faure-Lacroix ]
2809
* Ajout du logo gpl3
2811
* Ajout de la section screenshot pour le site
2812
* Ajout du favicon dans le header
2813
* Modification des cartes
2815
[ Alexandre Savard ]
2816
* Clean up <speex/libspeexdsp>
2819
* Fix new call button when recording
2821
* Recording: default home folder at startup
2822
* Minor changes to config window
2823
* IAX recording fixed
2824
* Set / get recording path, still need some GTK for client
2825
* AudioRecord file name format
2826
* Now recording in HOME folder
2829
* Fix bug in reqaccount.c
2831
[ Maxime Chambreuil ]
2832
* Update spanish translation
2835
* Update chinese translation
2837
[ Hussein Abdallah ]
2838
* Update russian translation
2840
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Sat, 14 Feb 2009 13:29:15 -0500
2842
sflphone (0.9.3-0ubuntu1) SYSTEM; urgency=low
2845
* Join thread before leaving
2846
* Fix implicit declaration in reqaccount
2847
* Add REST code to build the request to server
2848
* Fix GValue initialization warnings
2849
* Update version number, fix implicit declaration, fix GTK markup
2851
* Apply patch to create custom SIP account from our own server
2853
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 06 Feb 2009 19:17:32 -0500
2855
sflphone (0.9.2-2ubuntu9) SYSTEM; urgency=low
2857
[ Alexandre Savard ]
2858
* Speex audio codec preprocessing initialization
2859
* peer hung up segmentation fault solved
2860
* Stop recording when transfering
2861
* Terminate only one call
2862
* Add isRecording() function
2863
* Fix call_icon GTK client
2864
* Fix SIPCallClose() function, recorded file now close properly
2865
* Function terminateSIPCall added in sipvoiplink and managerimpl
2866
* Fix thread destructor
2867
* setRecordingOption function implement in audiorecord
2868
* Record now implemented in Call class
2869
* Record interface complete (on hold erase previous recording)
2870
* Added recButton in client
2871
* Added: record button related icons
2872
* Record button added
2873
* Overload AudioRecord::recData to get mic and speaker data mixed
2874
* Recording now in audiortp::run() method
2875
* Audio recording working in AudioRTP: receiveSessionForSpeaker
2876
* Open/close a wave file when pulse audio stream start/stop
2879
* Fix path for GTK+ icons; clean up
2881
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Thu, 05 Feb 2009 18:27:53 -0500
2883
sflphone (0.9.2-2ubuntu8) SYSTEM; urgency=low
2887
* Fix bug in merge and in Makefile.am
2888
* Terminate only one call
2889
* Disable PJsip shutdown when changing STUN parameters
2890
* Function terminateSIPCall added in sipvoiplink and managerimpl
2891
* Add a timer to the alsa thread to not jam the CPU load
2892
* Fix bug in sipvoiplink.cpp
2893
* Clean shutdown of pulseaudio on quiting
2894
* Fix DTMF at first start with Pulseaudio
2895
* Remove zeroconf from the build system
2896
* Add a library manager + exception handling
2897
* Clean up in the working directory
2898
* Better handling of capture XRUNs
2899
* Restore mic adjust volume on ALSA layer
2900
* Protect device ALSA operation if not opened
2901
* Fix the switching layer bug
2902
* Use dynamic_cast<> to use audiolayer-specific methods
2903
* Open the audio devices only once at startup
2904
* Refactoring of the ALSA part
2905
* Functional plug-in manager
2906
* Use a C++ thread to handle tones and DTMF in ALSA
2907
* Restore IAXVoIPLink, restore Mutex
2908
* Make the plugins registering against the plugin manager
2909
* Migrate to 1->N relationship between voiplink and accounts
2910
* API plugin for registration
2911
* Use C++ thread in SIP, move everything in sipvoiplink
2912
* Complete singleton pattern for the plugin manager
2913
* Add -Wno-return-type compilation flag to remove warnings; Update
2914
version number in configure.ac
2915
* Add the dynamic loading for the plugin framework; integate unittest
2918
* Update rpm spec file
2919
* modify build package script and spec file for suse
2921
[ Alexandre Savard ]
2922
* Add audiorecorder plugin and testaudiorecorder
2923
* Add audio Recording class, edit global.h
2925
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 04 Feb 2009 14:00:30 -0500
2927
sflphone (0.9.2-2ubuntu7) SYSTEM; urgency=low
2930
* Update changelog to 0.9.2-6
2931
* Fix some dbus-glib implementation details on the client side
2932
* Init history after dbus initialization
2933
* Add error checking in useragent; Clean sipvoiplink
2934
* Prevent crash when trying to call an empty number
2935
* Set the volume of the playback stream to PA_VOLUME_NORM at startup
2936
* Fix GTK+ generic value double initialization
2937
* Fix jaunty control file dependency problems
2938
* Fix jaunty control file dependency problems
2941
* Fix bug ticket # 137
2942
* Tolerant to gsm library of OpenSuse 11
2945
* Update german translation
2947
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 23 Jan 2009 17:48:13 -0500
2949
sflphone (0.9.2-2ubuntu6) SYSTEM; urgency=low
2952
* Migrate STUN configuration to the main config window
2953
* Update french translation
2954
* Other tiny memory leaks
2955
* Fix memory leak in sampleconverter.cpp
2956
* Generate packages from the release branch
2957
* update the build package script
2958
* modify the control files with architecture=any
2959
* Remove valgring uninitialized value
2960
* IAX and SIP use the same global variables to set account
2961
configuration ; fix broken code
2963
[ Maxime Chambreuil ]
2964
* Update spanish translation
2966
[ Hussein Abdallah ]
2967
* Update russian translation
2970
* Update translation files
2971
* Fix the bug when user uncheck the account which fails in the
2972
previous registration
2973
* Add stun error status
2974
* Fix bug ticket #143
2975
* Script for auto-install dependencies
2976
* Fix bug ticket #140
2977
* Fix bug ticket 141
2978
* Fix the reregister process when user change the details of an
2981
-- Emmanuel Milou <manu@sulfur.inside.savoirfairelinux.net> Fri, 16 Jan 2009 18:19:05 -0500
2983
sflphone (0.9.2-2ubuntu5) SYSTEM; urgency=low
2985
* Fix memory leak in the pulseaudio callback
2986
* Update debian package generation script
2987
* Warnings removal in GTK+ client
2988
* Clean adjust volume method in alsalayer
2989
* Plug the sflphone playback volume control to the pulseaudio volume
2991
* Display the date in history according to the current locale
2992
* Generate the changelog according to the git commit messages
2993
* Complete header in chinese translation file
2994
* Use the right gpg key to sign the packages
2995
* add debian jaunty jackalope support
2997
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 14 Jan 2009 21:17:20 -0500
2999
sflphone (0.9.2-2ubuntu4) SYSTEM; urgency=low
3002
* add german translation
3005
* Fix GUI crash in Ubuntu8.10 64bit system
3007
-- Yun Liu <yun.liu@savoirfairelinux.com> Thu, 08 Jan 2009 13:08:51 -0500
3009
sflphone (0.9.2-2ubuntu3) SYSTEM; urgency=low
3012
* The main thread synchronizes the ringtone thread
3013
* disable custom ringtone for the ALSA layer
3014
* Fix the Makefile.am in man directory, add a SEE ALSO section
3017
* Fix daemon crash caused by the previous patch ( for bug ticket #129)
3019
-- Yun Liu <yun.liu@savoirfairelinux.com> Tue, 06 Jan 2009 16:18:38 -0500
3021
sflphone (0.9.2-2ubuntu2) SYSTEM; urgency=low
3023
* Fix bug ticket #129
3025
-- Yun Liu <yun.liu@savoirfairelinux.com> Wed, 5 Jan 2009 15:54:53 -0500
3027
sflphone (0.9.2-2ubuntu1) SYSTEM; urgency=low
3029
* Migrate from eXosip library to pjsip
3030
* Add multiple SIP accounts support
3031
* Fix ringtones problems
3032
* Add a pulseaudio support
3033
* Improve audio quality with ALSA
3034
* Add chinese translation
3035
* Improve spanish translation
3036
* Migrate to a maintained C++ DBus bindings
3037
* Clean and improve the build system
3038
* Add build-dependency on Perl because we need pod2man to generate manpages
3040
-- Yun Liu <yun.liu@savoirfairelinux.com> Wed, 26 Nov 2008 09:47:53 -0500
3042
sflphone (0.9.1) unstable; urgency=low
3043
* Add a search tool in the history
3044
* Migrate some gtk_entry_new to sexy_icon_entry_new
3045
* Bug fix (Ticket #78): The voicemail password isn't displayed anymore in
3047
* Add the SIP registration expire value in the user file.
3049
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Thu, 22 May 2008 11:14:25 -0500
3051
sflphone (0.9.0) unstable; urgency=low
3052
* Add history features
3055
* Mouse events in the history tab
3056
* Smooth switch from the history tab to the calls tab
3057
* Remove most of GTK-Critical warnings
3059
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 13 May 2008 16:58:25 -0500
3061
sflphone (0.9-2008-06-06) unstable; urgency=low
3062
* Audio bug correction: capture stopped after a few minutes of conversation
3063
with USB Plantronics sound card
3065
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Tue, 06 May 2008 16:58:25 -0500
3067
sflphone (0.9-2008-05-06) unstable; urgency=low
3068
* Bug correction: account creation with the assistant
3069
* GTK+ warnings removal
3070
* libnotify warnings removal
3071
* Remove aliasing on the SFLphone logo
3073
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Mon, 05 May 2008 16:58:25 -0500
3075
sflphone (0.9) unstable; urgency=low
3076
* Clean dependencies ( removal of libboost )
3077
* Several GTK improvement and updates
3079
-configuration window
3080
* Migrate from GtkCheckMenuItem to GtkImageMenuItem
3081
* ALSA standard I/O transfers: MMAP instead of R/W
3082
* Fix speex audio quality
3084
-Fix hold/unhold situation
3087
-Ringtone on incoming call
3088
-Fix transfer situation
3089
* Add desktop notification ( libnotify )
3090
* Improve the system tray icon behaviour
3091
* Improve registration error handling
3092
* Register/unregister from the account window takes effect without starting back SFLphone
3093
* Compilation warnings removal
3095
* Add an account configuration wizard
3097
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 30 Apr 2008 16:58:25 -0500
3099
sflphone (0.8.2) unstable; urgency=low
3100
* Internationalization of the GTK GUI
3103
* Slight modifications of the graphical interface ( tooltips, dialpad, ...)
3105
-- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 21 Mar 2008 11:37:53 -0500
added
removed
tools/build-system/launchpad/sflphone-common/debian/changelog
