diff --git a/tools/build-system/launchpad/sflphone-client-gnome/debian/changelog b/tools/build-system/launchpad/sflphone-client-gnome/debian/changelog index 003c33d69cbe10ad5a83b1bebc81c43be2f76df3..c6bec4586d4b6391b32588c2eb5265f2bdbce8c6 100644 --- a/tools/build-system/launchpad/sflphone-client-gnome/debian/changelog +++ b/tools/build-system/launchpad/sflphone-client-gnome/debian/changelog @@ -1,4 +1,573 @@ -sflphone-client-gnome (0.9.6-SYSVER) SYSTEM; urgency=low +sflphone-client-gnome (0.9.7~beta~ppa1~karmic) karmic; urgency=low + + ** 0.9.7~beta~ppa1~karmic ** + + * [#1933] Cleanup debug + * [#1933] Clean up debug + * Fix mic + * [#1933] Set the IAx format earlier + * [#1933] Move IAX sendAudioFromMic outside if (call) statement + * [#1933] Fix startstream when offhold in iax and add debug concerning + codec neg. + * [#2371] sflphone_notify_voice_mail: minor gettext message formatting + cleanup + * [#2371] select_account_cb: properly gettextize status message + * [#2371] show_account_list_config_dialog: properly gettextize status + message + * INSTALL: Minor tidyup of core install guide + * Add /sflphone-client-gnome/src/icons/Makefile to .gitignore + * [#2181] Updated OpenSUSE files (tmp) + * [#1933] Add debug for codec negociation for iax + * [#1933] Get rid of getMicAvail and getMicData in audiolayer (not + used anymore) + * [#1933] Add "audio codec not determined" error in IAX + * [#1933] Test flush data + * [#1933] Do not need to start audio stream in iax anymore + * [#1933] Protecting pointer + * [#2284] Remove more compilation/execution warnings + * [#2284] Cleanup debug in client, use DEBUG instead of g_print + * [#2284] Clean up uimanager + * [#2370] Remove warnings + * [#2366] Clean up other debug + * [#2366] Clean up debug + * [#2366] Call pa_xfree explicitely in writeToSpeaker + * [#2284] Remove address book warnings + * [#2365] Fixes bad cast + * [#2352] Fix continuous ringing when peer hangup and call not yet + answered + * [#2181] Added version support + * [#2181] Fixed some minor issues + * [#2360] Moved MainBuffer from AudioLayer to ManagerImpl + * [#2352] Makes getMainBuffer() everywhere + * [#2352] Use 50 sec latency on pulseaudio stream creation + * [#2352] Add alsa debug + * [#2359] Update repository documentation + * [#2354] Move pulseaudio disconnectAudioStream after stopping main + loop + * [#2352] Adjust nb byte copied in pulseaudio according to + writeableSize + * [#2352] Specify pulseaudio tlength parameters using pa_usec_to_bytes + * [#2322] Convert italian translation to UTF-8 + * [#2357] Fixes window size + * [#2357] Display only actionnable tool item + * [#2333] Update streams parameters + * [#2347] Use GNOME user settings for Menu and Toolbar appareance + * [#2349] Load/Save properly audio params + * [#2322] Update translations from Launchpad + * [#2181] Added Francois Marier script + * [#2350] Remove non-valid test + * [#2181] Updated launchpad packaging + * [#2333] Fix Pulseaudio Capture + * [#2333] Use pulseaudio ADJUST_LATENCY flag and ALSA RT-SCHEDULING + * [#2333] Pulseaudio Interpolate timing + * [#2333] Change (again) Pulseaudio settings to fit logiteck usb hdw + requirement + * [#2333] Adjust pulseaudio fragment size to 4096 (max sflphone's + frames per buffer) + * [#2284] Remove recurrent compilation warning (g++ linker problem) + * [#2333] Safer Audiostream parameters + * [#2333] Fix alsa playback to reduce underrun + * [#2333] Better audiostream parameters + * [#2181] Updated version management + * [#2333] Exclusive test in playback loop + * [#2181] Updated build system + * [#2333] Less underrun with these value + * [#2333] Update playback audiostream parameters + * [#2333] Lengthen the audio buffer reduce number of underrun in + pulseaudio + * [#2333] Add ALSA recovery functions for underrun (begin) + * [#2333] Add pa_stream_trigger in pulse audio underrun callabck + * [#2048] Reduce prebuffering in pulseaudio (which affect incomming + calls' plbck) + * [#2316] Do not display any icons to the right on the history tab + * [#2333] Comment pa_stream_trigger in pulseaudio underrun + * [#2333] Modify pulseaudio streams parameters + * [#2318] Fix transfer tool button double signal + * [#2181] Updated + * [#2333] Fix ALSA ringtone + * [#2333] Flush all main buffer before starting audio + * [#2333] Open/Close Alsa thread between calls while there is no audio + * [#2333] Add debug message and test condition on starting playback + and capture + * [#2181] Fixed gnome client makefile + * [#2181] Updated + * [#2308] Remove getTelephoneTone debug + * [#2308] Change plughw for default in ALSA + * [#2308] Oups, forgot to change function name in audiolayertest.cpp + * [#2308] Cleanup in pulseaudio code (debug, function name) + * [#2308] Fix pulseaudio stream closing assertion failure + * [#2308] Moved pulseaudio mainloop locking from AudioStream + disconnect stream + * [2308] Fix latency at the beginning of a call, when playing DTMF and + wehn starting tone + * [#2181] Updated karmic + * [#2317] [#2319] Fix address book toggle button contextual behaviour + * [#2308] Stop stream when refusing a call + * [#2308] Stop pulseaudio stream when peer hungup + * [#2308] Fix tone and ringtone + * [#2312] Display the STUN entry widget when opening the tab + * [#2308] Implement two different callbacks for capture/playback in + pulseaudio + * [#2309] Open/close pulseaudio connections in startStream/stopStream + * [2308] Leave pulseaudio stream running, do not cork/uncork them + anymore + * [#2295] Set gtk file chooser to None if nothing is set in + configuration + * [#1976] Add codec and conference documentation + * [#2209] Fix recording in regard of resamling + * [#2297] Update .gitignore + * [#2297] Update translation files + * [#2297] Add reference to our coding standards + * [#2297] Remove old docbook code + * [#2296] Reinit tls account settings after modification + * [#2253] Add DcBlocker class to remove capture's dc offset + * [#2034] Fixes for TLS transport to initialize + * [#2284] Add silent build rule + client clean warnings + * [#2274] Fix unserialize history items in cilent at startup + * [#2274] Complete display name parsing and displaying + * [#2274] Parse the Display Name in sip INVITE message + * [#2050] Fix capture volume control in ALSA + * [#1970] Volume controls disable when using pulseaudio + * [#1970] Disable volume controls when using pulseaudio + * [#2277] Fix direct ip2ip ZRTP enabling/disabling in ip2ip + preferences + * [#2181] Added launchpad debian files + * [#2181] Added spec files for OSC + * [#2274] Set display name for "Contact" sip header as the hostname + * [#2181] Fixed daemon issues + * [#2181] Fixed gnome client issues + * [#1976] Remove warnings - need to fix the transfer + * [#2006] Add init is_rec variable in ManagerImpl + * [#2006] Update codec display on call selection + * [#2006] Restore double click actions in history and contact calltree + (GTK) + * [#2176] use XDG_CACHE_HOME when initializing sfl.zid file + * [#1976] Fix calltree switching from history + * [#2209] (Re)Fix cache for zid + * [#2209] Clean up debug messages + * [#2209] Clean debug messages + * [#2209] Fix trasnfering a call during a conference + * [#2209] Speex decode must return the number of bytes + * [#2209] Change frameSize speex 32kHz + * [#2209] Fix speex codec framesize + * [#2209] Reinit converterSamplingRate in RTP sessions + * [#2209] Change speex ultra wide band framesize + * [#1747] Add pixmap data + * [#2252] Fix Receiving a server error 488 crashes the callee + * [#2209] Fix iax low rate packate sending + * [#2209] Clean up debug messages + * [#2209] Add resampling changes for IAX + * [#2209] Clean up resampling code + * [#2209] Fix latency introduced by pulseaudio + * [#2209] Fix initialization of mainbuffer's internal sampling rate + * [#2176] Fix upsampling buffer size in audiolayer + * [#2209] Add dynamic converter sampling rate in audiortp sessions + * [#1747] Fixes runtime warnings + * [#1747] Remove from repo + * [#1747] register our icons to be used as stock icons + * [#2209] Fix number of byte in alsa's write to speaker + * [#2209] Fix putting non-resampled data in RTP's mainbuffer + * [#2209] Add alsa resampler + * [#2209] Add a samplerate converter in PulseLayer + * [#2209] Add mainbuffer's internal sampling rate and flushall method + * [#2176] Add mainbuffer stateInfo debug method + * [#2209] Resampling is optimal using SRC_LINEAR not SRC_FASTEST + * [#2176] Remove debug recordings + * [#2176] Fix Holding a conference participant on new calls + * [#2224] Add confID in callable object + * [#2176] Fix putting onhold a call participating to a conference when + pressing new call + * [#2176] Reset auidio buffers when adding streams (rtp, audiolayer) + * [#1976] Use xml to describe toolbars - Add a naviguation toolbar + * [#2176] Remove conference default_id in joinParticipant + * [#2176] Display error message in alsa's snd_pcm_avail_update call + * [#2176] Alsa mic avail data debug + * [#2176] Add some debug message for mic loss problem + * [#2176] Flush mic ring buffer when offholding a call + * [#2176] Reset ringbuffers' readpointer when adding main participant + * [#2176] Fix getAvailData algorithm + * [#2176] Reset ringbuffer's readpointer when adding a new participant + to a conference + * [#1744] Regex object renamed to Pattern. Previous attempt at + providing + * [#2176] Fix detach main participant problem when adding new one + * [#1976] Use right domain to translate + * [#1976] Add xml menu description + * [#2176] Store a list of confernece participant in client + * [#2176] Fix add participant, joinparticipant methods + * [#2181] Do not install dbus-c++ headers + add return value + * [#2176] Fix minor call handling instabilities + * [#2174] Fix incoming IP call contact address + * [#2211] Add test to protect NULL pointer + * [#1163] Add Advanced account configuration section + * [#2176] Add some usefull comments and debugging info + * [#2176] Add conditions to display security icons in conference + * [#2176] Fix detaching one participant while keeping communication to + others + * [#2176] Reenable userActive.svg in call tree + * [#2176] Make user active blue (not red) + * [#2176] Fix user active picture + * [#2176] Fix "hidden" merge conflict in sipvoiplink + * [#2176] Remove iax audio stream on peer hungup + * [#2174] Multiple UDP transports functional (TESTED with 2 accounts + and 3 calls) + * [#2176] Fix fix audio stream binding in iax + * [#2174] Create a default UDP transport + use tp selector for dialogs + also + * [#2176] Register iax audio stream in mainbuffer + * [#2176] Fix getAudioCodecName in IAXvoipLink + * [#2176] Fix iax account init + * [#2176] Handle multiple account using the same sip transport + * [#2165] Add .png files + * [#2176] Small fixes concerning dtmf + * [#2176] Fix make uninstall in codecs + * [#2174] remove stund makefile generation + * [#2176] Add conference lock + * [#2174] Add transport selector for multiple accounts + * [#2176] Change userActive picture from red to blue + * [#2176] Fix security pixbuff in calltree + * [#2176] Replace sfl.zid in .cache/sflphone instead of .sflphone + * [#2176] Fix add call description + * [#2176] Remove detach button from toolbar + * [#2176] Fix calltree call description state and state code in + conferences + * [#2176] Fix pulse audio double free + * [#2176] Fix conference selection + * [#2174] Clean up - remove stun settings in client network + configuration panel + * [#2174] Remove voviva stun code + * [#2174] Rsolve STUN with pjsip - DO NOT WORK + * [#2165] Add user svg + * [#2165] Debugging sip call failed + * [#929] Link against uuid if installed + * Oops + * Fixed bugs related to libsexy (with GTK < 2.16) + * [#929] Remove uuid-dev dependency in the core + * [#2165] Debugging no negociated codecs at communicatio start + * [#2165] Fix calltree bug (gtktreestore instead of gtkliststore) + * [#2165] Fix several merge problems + * Updated opensuse packaging script + * [#1163] Add missing figures + * [#1163] Update INSTALL file + * [#2165] Fix IAX + * [#2165] Add recordabe interface + * [#2165] Finish recording refactoring for call (not for conference) + * [#2165] Enable speaker recording for two different calls + simultanously + * [#2165] Implement call recording using the Recordable interface + * [#2165] Add get and set to AudioLayer's audio recorder + * [#2165] Add class recordable from which inherit call and conference + * [#2006] Fix G722 and Speex 8khz codec conferencing + * [#2006] add recording of audio buffers + * [#1163] Add general settings section + * [#1163] Fixes makefile error + * [#2006] Fix some minor issues + * [#2006] Drag a conference call on another conference call + (difference conferences) + * [#2006] Fix dragging a conference on itself + * [#1744] Integrating some of the needed regular expression patterns + in order + * COmplete call features + * [#1744] Added support for named subgroup in the Regex object. Also, + new + * [#1744] Adds thread safety features, compile() and setPattern() + methods to the Regex class. + * [#1744] Fix inconsistency in the finditer method from the last + commit. + * [#1744] Added regex pattern object built on top of libpcre. To be + used + * [#1744] Initial commit towards implementing RFC4568. Unimplemented + in the + * [#2157] Hide "security" and "advanced" tabs for IAX under account + * [#1163] Add call features section + * [#2006] Add joinConference capabilities + * [#2006] Add dbus joinConference signal + * [#2006] Drag a conference call onto a conference to add it + * [#1163] Add addressbook section + * [#2006] Drag a conference call onto a single call to create a + conference + * [#2006] Expand rows automatically + * [#2006] Add minimal multiple conference handling + * [#2006] Add atached/detached conference icons + * [#2006] Add function processRemainingParticipant + * [#2006] Deep refactoring, fix hangup bug + * [#1163] Update documentation - Accounts part + * [#1976] Integrate user doc to gnome client build system + * [#2122] Remove double inclusion in dbus-c++/src/Makefile.am + * Remove pjproject version number + * [#2006] Fix peerHungup + * [#1976] Make Yelp accessible from the GNOME client (need to install + the sflphone.xml first) + * [#2006] Fix multiconferencing hangup + * [#2006] Fix hangup calls in a conference + * [#2150] Make IAx2 reappear + * [#2006] Fix detach participant on multiple call + * [#2006] Can remove rining call from a conference + * [#2006] Reinit confID when removing a participant + * [#2006] Remove get isCurrentCAll in hangup/peerhungup (SipVoipLink) + * [#2006] Fix refuse call + * [#2006] Fix answerring incoming call + * [#2006] Refactor conference's participant list + * [#2101] Re-integrate test compilation in main build system + * [#2101] Make the test directory compile + * [#2136] Restore history functionality + * [#2006] Fix binding main participant to himself + * [#2006] Fix add current/incoming/onHold participant to an existing + conference + * [#2006] Fix add incoming calls to an already created conference + * [#2006] Fix remove stream + * [#2006] Fix detachParticipant/removeParticipant switchCall ids + * [#2006] Fix adding a call in conference having state "CURRENT" + * [#2006] Remove/add main participant from conferences + * [#2006] Hold/unHold conference + * [#2006] Detach a partcipant from drag n drop + * [#2006] Hangup a conference + * [#2006] Add hold/unhold conference dbus messages + * [#2034] gtk-ui fix under the "basic" tab. + * [#2006] Fix dragging calls on conference calls + * [#2006] Fix detach participant from a conference + * [#2034] Added default message is status bar under the account config + dialog + * [#2112] Fix a crashed caused when a non-md5 password was sent to + pjsip. + * [#2006] Detach participant by ID + * [#2006] Fix addParticipant method in managerImpl to handle + incoming/answered calls + * [#2006] Add addParticipant method in managerimpl and related dbus + messages + * [#2111] Added the ability to configure zrtp on sip.sflphone.org from + * [#2106] Fixed problem in the account assistant under gtk-ui. Also, + assistant.c + * [#2006] Fix dragging a conference call on another conference call + (same conference) + * [#1904] Small UI fix. Assistant was moved from "Call" to "Edit" + menu. + * [#1904] Fix a wrong label under gtk-ui. + * [#2034] Renaming and source code splitting. + * [#2034] Status bar added to account window to better reflect the + registration + * [#2006] Make calltree_remove_call recursive (for GtkTreeStore) + * [#1110] Small gtk-UI fix in the account window (alignment). + * [#2006] Fix remove conference, display children which are still + active + * [#2006] Recursive function call in calltree_update_call + * [#2006] Add multilayered capabilities to calltree (GtkTreeStore) + * [#2006] Implement remove conference in calltree + * [#2034] Now useless as Direct Ip calls settings moved under + Preferences. + * [#2034] Edit/add buttons were set insensitive all the time under + gtk-ui. + * [#1887] Information about the state of the current SIP call is + displayed + * [#2006] Add call tree remove callback + * [#2006] Fix create_conference function + * [#2006] Update conference_added_cb to add new conference to the list + * [#812] Added new tab under GTK-ui Preferences. Moving Direct Ip + Calls from + * [#2121] Disable temporarily test compilation + * [#2006] Fix conferencelist to handle conference_obj_t instead of + gchar + * [#2006] Add conference_obj structure + * [#2121] Update version + * [#2006] Fix conference selection + * [#2101] Use the new source tree to fetch the right object files + * [#2006] Add conference in calltree + * [#2006] Add Dbus signal conference added/removed/changed + * [#2006] Add getConferenceDetails call on dbus + * [#1904] Registration expire now appears as a spin box under gtk-ui. + * [#812] Fixing a segmentation fault caused by a non-existing account + ID + * [#2006] Add getConfList method over dbus + * [#2006] Add a conferencelist data structure in client-gnome + * [#812] Defaults value are now sent if a non-existing account is + requested + * [#2006] Add sflphone action sflphone_join_participant + * [#2006] Fix buffer read pointer problem deletion + * [pjsip] Attempt at fixing via header incompatibility with + Freeswitch. + * [#1797] forget something + * [#2006] Add call new state conferencing in deamon + * [#2006] Remove addParticipant method for conference, use + joinParticipant only + * [#1163] Update INSTALL documentation + * [#812] Msec/sec values were not taken into account. + * [#1797] Make pjproject-1.4 compile + * [#2006] Add Detach participant method + * [#2006] Dragndrop fully functional with INCOMING and HOLD call + * [#1797] Add pjproject-1.4 + * [#1797] Remove pjproject-1.0.3 + * [#2006] Get call state in conference related function + * [#2006] Add joinParticipant (conference) method in ManagerImpl + * [#2006] Add joinConference DBUS message + * [#2006] Store the previously selected call_id on dragndrop + * [#2006] Fix GValue pointer unref in selection callback + * [#2006] Store dragged call_id + * [#2006] Update drag_data_received_cb callback to manipulate CallIDs + * [#2006] Add dragndrop signals + * [#2006] Set calltree reordable + * [#812] Adds the ability to create a TLS listener in case the user + requests + * [#812] Adds the ability to configure local/published address from + * [#1883] Move switchCall in onHoldCall function + * [#812] Deals with the published address/port problem when + integrating TLS. + * [#1883] Switch call id in managerimpl when peerHungUp + * [#1883] Switch call id before hangup + * [#1883] Add usefull and permanent debug info for conference + cretion/deletion + * [#812] Fix various segmentation faults related to Direct IP kind of + calls. + * [#1883] Fix deletion of std::map elements using iterators + * [#2014] Add libzrtpcpp build dependency + * [#1883] Still some for loop test ambiguity (while loop instead) + * [#1883] Fix for loop initial test ambiguity (use while loop instead) + * [#1883] We must discard data in urgent ring buffer if data is get in + mainbuf + * [#1883] Fix availForGet same id for ringbuffer and readpointer + * [#812] Match "sips" as a Direct IP Call when the user enter a sip + uri + * [#812] Fix segmentation fault related to SIP URI creation. + * [#812] Towards integrating multiple tls listeners at the same time. + This + * [#1883] Add debug messages in conference and fix mainbufferTest + * [#812] gkt-ui fix. Private key must be fed as a filename and not as- + is. + * [#812] TLS integration within sipvoiplink and pjsip. Also, + configure.ac + * [#1883] Fix Alsa/Pulse mallocation + * [#1883] Fix data corruption in AudioRtp's micData buffer + * [#812] Full dbus integration for all the tls related options under + gtk-ui. + * [#1883] Fix memory leaks in audiortp session + * [#1883] Fix mem leaks in audio rtp + * [#812] Fix setAccountDetails where TLS_ENABLE was set to the value + * [#812] Small gtk-ui fix. + * [#811][#812] Small gtk-ui fix. + * [#812] Introduced a mechanism for configuration files that makes + possible + * [#812] New dbus bindings added. Also, configuration compliance was + enforced + * [#1881] Remove default buffer from MainBuffer (update unit-tests) + * [#1881] Add ring buffer read pointer tests + * [#1883] Fix issues in ringbuffer reader pointers + * [#2034] Implementing a new configuration dialogue for TLS transport + settings + * [#1883] Add some usefull debug and safety checks + * [#2028] Notify the client with libnotify when the zrtp negotiation + failed. + * [#811] Harmless no to throw an exception, an makes the application + less + * [#2028] A minidialog is showed to the user under sflphone-client- + gnome + * Removed useless file. + * Ignoring Makefile in src/widget + * [#2027] Fix segmentation fault when showMessage callback is called + after + * [#2026] keyExchange was set to ZRTP instead of "1" + * [#2024] Fix the wrong summary at the end of the assistant. + * [#1883] Fix mnagerimpl conference map insertion + * [#1883] Add Mutexes in MainBuffer + * [#811] Gtk ui was not presenting the right information about zrtp + for + * [#2023] security icons were not installed in sflphone-client-gnome. + * [#2021] Fix a mistake in the readme from sflphone-common that gives + wrong + * [#811] The current SRTP mode was not properly displayed for the + IP2IP + * [#1743] Re-implementation of the "automatically remove error dialogs + [...]" + * [#2017] [#2019] Fix the inability to dial a number and place a + registered + * [#811] Final re-integration of ZRTP support in the main branch from + 0.9.6 + * [#1883] Fix map insertion methods + * [#811] Combo box now is now set to the active key exchange method + * [#811] ZRTP options now configurable back again from the Gtk UI. + IP2IP + * Updated hostname for git clone + * [#1883] Add minimal functionalities to create a conference + * [#811] re-integration of all the methods and signals on dbus. + ManagerImpl + * [#811] Got out of a precarious position were nothing would compile. + * [#1976] Build documentation squeleton with docbook + * [#1883] Add sflphone-client "addParticipant" button for conference + * [#1994] Better organize the source directory structure. New + subdirectories + * [#1883] Add a simple Conference class + * [#1882] Use static audio buffer in Pulse and ALSA layer (instead of + malloc) + * [#811] First commit toward re-integration and refactoring of ZRTP + * [#1882] Flush RTP ring buffer before entering mainloop + * [#1882] Fixed MainBuffer::UnBinCallID() in case there is no + ringbuffer + * [#1882] Test (and fixe) high level conference and mixing + functionalities + * [#1772] Apply patch to compile on fedora (sent by Marcin + Zajączkowski <mszpak@wp.pl>) + * [#1882] Update Bind, unBind call_id in MainBuffer + * [#1959] This adds the ability to store password as an MD5 Hash in + the + * [#1538] Fixes rules compilation + * [#1930][#1931] Fixed a mistake (again) related to index and + credential count + * [#1753] Remove ILBC from pjproject - Hacks in pjsip + * [#1930][#1931] Credential was not selected properly using realm + * [#1882] Finilize multiple reading pointer in RingBuffer + * [#1538] Remove configure from autogen.sh to respect debian upstream + authors policy + * [#1773] Remove generated files from repo + * [#1791] Use XDG_CACHE_HOME to save pid file + * [#1791] Fixes path to save history + * [#1791] Fix debian installation scripts + * [#1930][#1931] Settings are now taken into account in the server. + * [#1882] Add ringbuffer default ring buffer pointer in methods + involving mStart + * [#1882] Add default ringbuffer pointer + * [#1882] Add RingBuffer multiple read pointer basic functionnalities + * [#1882] Fix MainBuffer flushData unit test + * [#1930][#1931] Ability to save and retreive the configuration from + * [#1882] Added Multiple CallID mapping to MainBuffer + * [#1791] Not much + * [#1791] If XDG env variables are not null but empty, use default + ones + * [#1791] Make XDG_CONFIG_HOME writable + * [#1930][#1931] Partial commit. Not working yet. Cannot delete + account + * [#1881] Fixed alsa capture latency problem + * [#1881] Fixed Alsa capture temporarily + * [#1930] [#1931] Partial unbroken commit providing the ability to + * [#1881] MainBuffer implemented in AudioLayer/AudioRTP + * [#1881] Add discard and flush unit-tests + * [#1881] Add discard and flush functionnalites to MainRingBuffer + * [#1881] Add availForGet in MainBuffer + * [#1881] Add availForPut function to MainBuffer + * [#1880] Remove AudioRTP* pointer from SipVoIP (reapered while + merging master) + * [#1881] Add a map between call id and coresponding ring buffer + * [#1855] Refresh pot file and upload on Launchpad + * [#1881] MainBuffe now robust to false ids on getData and putData + * [#1881] Fix big big big memory leak + * [#1881] Add getData and putData to mainBuffer + * [#1881] Unit-test basic ring buffer functionnaities + * [#1881] Add class MainBuffer and basic buffer creation unit-tests + * [#1880] Fix call transfer (step2) issues + * [#1880] Moved AudioRtp* pointer from SIPVoIPLink to SIPCall class + * [#1791] Add postinst script to keep user data when migrating + config/history file + * [#1797] Make pjsip compile + * [#1777] Code indentation + * [#1791] Use XDG_DATA_HOME and XDG_CONFIG_HOME for sflphonedrc and + history + unit tests + * [#1746] Useless space does not appear anymore when volume sliders + and + * [#1643] GtkCheckMenuItem is used instead of icons for elements in + the + * [#1110] [#1668] STUN parameters are now located in the preferences, + under + + -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 06 Nov 2009 11:20:01 -0500 + +sflphone-client-gnome (0.9.6-SYSVER) karmic; urgency=low ** 0.9.6 ** @@ -65,7 +634,7 @@ sflphone-client-gnome (0.9.6-SYSVER) SYSTEM; urgency=low -- SFLphone Automatic Build System <team@sflphone.org> Mon, 27 Jul 2009 09:53:19 -0400 -sflphone-client-gnome (0.9.6~rc2-SYSVER) SYSTEM; urgency=low +sflphone-client-gnome (0.9.6~rc2-SYSVER) karmic; urgency=low ** 0.9.6~rc2 ** @@ -120,7 +689,7 @@ sflphone-client-gnome (0.9.6~rc2-SYSVER) SYSTEM; urgency=low -- SFLphone Automatic Build System <team@sflphone.org> Fri, 17 Jul 2009 19:12:58 -0400 -sflphone-client-gnome (0.9.6~rc1-SYSVER) SYSTEM; urgency=low +sflphone-client-gnome (0.9.6~rc1-SYSVER) karmic; urgency=low ** 0.9.6~rc1 ** @@ -228,7 +797,7 @@ sflphone-client-gnome (0.9.6~rc1-SYSVER) SYSTEM; urgency=low -- SFLphone Automatic Build System <team@sflphone.org> Fri, 10 Jul 2009 15:50:26 -0400 -sflphone-client-gnome (0.9.6~beta-SYSVER) SYSTEM; urgency=low +sflphone-client-gnome (0.9.6~beta-SYSVER) karmic; urgency=low ** 0.9.6~beta ** @@ -523,7 +1092,7 @@ sflphone-client-gnome (0.9.6~beta-SYSVER) SYSTEM; urgency=low -- SFLphone Automatic Build System <team@sflphone.org> Tue, 23 Jun 2009 11:13:42 -0400 -sflphone-client-gnome (0.9.5-SYSVER) SYSTEM; urgency=low +sflphone-client-gnome (0.9.5-SYSVER) karmic; urgency=low ** 0.9.5 release ** @@ -554,7 +1123,7 @@ sflphone-client-gnome (0.9.5-SYSVER) SYSTEM; urgency=low -- SFLphone Automatic Build System <team@sflphone.org> Mon, 25 May 2009 11:34:48 -0400 -sflphone-client-gnome (0.9.5-SYSVER~rc2) SYSTEM; urgency=low +sflphone-client-gnome (0.9.5-SYSVER~rc2) karmic; urgency=low ** 0.9.5 rc2 ** @@ -608,7 +1177,7 @@ sflphone-client-gnome (0.9.5-SYSVER~rc2) SYSTEM; urgency=low -- SFLphone Automatic Build System <team@sflphone.org> Tue, 19 May 2009 12:08:18 -0400 -sflphone-client-gnome (0.9.5-0ubuntu1~rc1) SYSTEM; urgency=low +sflphone-client-gnome (0.9.5-0ubuntu1~rc1) karmic; urgency=low [ SFLphone Project ] * [#1262] Updated changelogs for version 0.9.5-0ubuntu1 Snapshot 2009- @@ -637,7 +1206,7 @@ sflphone-client-gnome (0.9.5-0ubuntu1~rc1) SYSTEM; urgency=low -- Sflphone Project <sflphone@mtl.savoirfairelinux.net> Tue, 05 May 2009 19:16:13 -0400 -sflphone-client-gnome (0.9.5-0ubuntu1~beta) SYSTEM; urgency=low +sflphone-client-gnome (0.9.5-0ubuntu1~beta) karmic; urgency=low [ Julien Bonjean ] * Updated Eclipse stuff @@ -859,7 +1428,7 @@ sflphone-client-gnome (0.9.5-0ubuntu1~beta) SYSTEM; urgency=low -- Sflphone Project <sflphone@mtl.savoirfairelinux.net> Mon, 27 Apr 2009 17:00:03 -0400 -sflphone-client-gnome (0.9.4-0ubuntu2) SYSTEM; urgency=low +sflphone-client-gnome (0.9.4-0ubuntu2) karmic; urgency=low [ Alexandre Savard ] * Restore speex and GSM detection @@ -869,7 +1438,7 @@ sflphone-client-gnome (0.9.4-0ubuntu2) SYSTEM; urgency=low -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 8 Apr 2009 11:29:15 -0500 -sflphone (0.9.4-0ubuntu1) SYSTEM; urgency=low +sflphone (0.9.4-0ubuntu1) karmic; urgency=low [ Emmanuel Milou ] * Integrate DBus-c++ and libiax2 in the main build system @@ -894,7 +1463,7 @@ sflphone (0.9.4-0ubuntu1) SYSTEM; urgency=low -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 3 Apr 2009 18:29:15 -0500 -sflphone (0.9.4-rc1) SYSTEM; urgency=low +sflphone (0.9.4-rc1) karmic; urgency=low [ Emmanuel Milou ] * Fix bug while trying to hold/unhold several simultaneous call @@ -908,7 +1477,7 @@ sflphone (0.9.4-rc1) SYSTEM; urgency=low -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 27 Mar 2009 18:29:15 -0500 -sflphone (0.9.4-0beta1) SYSTEM; urgency=low +sflphone (0.9.4-0beta1) karmic; urgency=low [ Alexandre Savard ] * Display codec used during conversation on the GUI @@ -924,7 +1493,7 @@ sflphone (0.9.4-0beta1) SYSTEM; urgency=low -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 20 Mar 2009 18:29:15 -0500 -sflphone (0.9.3-0ubuntu3) SYSTEM; urgency=low +sflphone (0.9.3-0ubuntu3) karmic; urgency=low [ Alexandre Savard ] * Both playback and record streams in PA_STREAM_CORKED (pulseaudio) @@ -949,7 +1518,7 @@ sflphone (0.9.3-0ubuntu3) SYSTEM; urgency=low -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Sat, 14 Feb 2009 13:29:15 -0500 -sflphone (0.9.3-0ubuntu2) SYSTEM; urgency=low +sflphone (0.9.3-0ubuntu2) karmic; urgency=low [ Emmanuel Milou ] * Add compilation note in README @@ -1018,7 +1587,7 @@ sflphone (0.9.3-0ubuntu2) SYSTEM; urgency=low -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Sat, 14 Feb 2009 13:29:15 -0500 -sflphone (0.9.3-0ubuntu1) SYSTEM; urgency=low +sflphone (0.9.3-0ubuntu1) karmic; urgency=low * Remove debug * Join thread before leaving @@ -1031,7 +1600,7 @@ sflphone (0.9.3-0ubuntu1) SYSTEM; urgency=low -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 06 Feb 2009 19:17:32 -0500 -sflphone (0.9.2-2ubuntu9) SYSTEM; urgency=low +sflphone (0.9.2-2ubuntu9) karmic; urgency=low [ Alexandre Savard ] * Speex audio codec preprocessing initialization @@ -1059,7 +1628,7 @@ sflphone (0.9.2-2ubuntu9) SYSTEM; urgency=low -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Thu, 05 Feb 2009 18:27:53 -0500 -sflphone (0.9.2-2ubuntu8) SYSTEM; urgency=low +sflphone (0.9.2-2ubuntu8) karmic; urgency=low [ Emmanuel Milou ] * Update changelogs @@ -1103,7 +1672,7 @@ sflphone (0.9.2-2ubuntu8) SYSTEM; urgency=low -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 04 Feb 2009 14:00:30 -0500 -sflphone (0.9.2-2ubuntu7) SYSTEM; urgency=low +sflphone (0.9.2-2ubuntu7) karmic; urgency=low [ Emmanuel Milou ] * Update changelog to 0.9.2-6 @@ -1125,7 +1694,7 @@ sflphone (0.9.2-2ubuntu7) SYSTEM; urgency=low -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 23 Jan 2009 17:48:13 -0500 -sflphone (0.9.2-2ubuntu6) SYSTEM; urgency=low +sflphone (0.9.2-2ubuntu6) karmic; urgency=low [ Emmanuel Milou ] * Migrate STUN configuration to the main config window @@ -1159,7 +1728,7 @@ sflphone (0.9.2-2ubuntu6) SYSTEM; urgency=low -- Emmanuel Milou <manu@sulfur.inside.savoirfairelinux.net> Fri, 16 Jan 2009 18:19:05 -0500 -sflphone (0.9.2-2ubuntu5) SYSTEM; urgency=low +sflphone (0.9.2-2ubuntu5) karmic; urgency=low * Fix memory leak in the pulseaudio callback * Update debian package generation script @@ -1175,7 +1744,7 @@ sflphone (0.9.2-2ubuntu5) SYSTEM; urgency=low -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 14 Jan 2009 21:17:20 -0500 -sflphone (0.9.2-2ubuntu4) SYSTEM; urgency=low +sflphone (0.9.2-2ubuntu4) karmic; urgency=low [ Emmanuel Milou ] * add german translation @@ -1185,7 +1754,7 @@ sflphone (0.9.2-2ubuntu4) SYSTEM; urgency=low -- Yun Liu <yun.liu@savoirfairelinux.com> Thu, 08 Jan 2009 13:08:51 -0500 -sflphone (0.9.2-2ubuntu3) SYSTEM; urgency=low +sflphone (0.9.2-2ubuntu3) karmic; urgency=low [ Emmanuel Milou ] * The main thread synchronizes the ringtone thread @@ -1197,13 +1766,13 @@ sflphone (0.9.2-2ubuntu3) SYSTEM; urgency=low -- Yun Liu <yun.liu@savoirfairelinux.com> Tue, 06 Jan 2009 16:18:38 -0500 -sflphone (0.9.2-2ubuntu2) SYSTEM; urgency=low +sflphone (0.9.2-2ubuntu2) karmic; urgency=low * Fix bug ticket #129 -- Yun Liu <yun.liu@savoirfairelinux.com> Wed, 5 Jan 2009 15:54:53 -0500 -sflphone (0.9.2-2ubuntu1) SYSTEM; urgency=low +sflphone (0.9.2-2ubuntu1) karmic; urgency=low * Migrate from eXosip library to pjsip * Add multiple SIP accounts support diff --git a/tools/build-system/launchpad/sflphone-common/debian/changelog b/tools/build-system/launchpad/sflphone-common/debian/changelog index 3d95374e2598dae7692cbf901ef52e4ddb27eed6..98e662737c05b07b4651f4ba6d0edfd830ead1f5 100644 --- a/tools/build-system/launchpad/sflphone-common/debian/changelog +++ b/tools/build-system/launchpad/sflphone-common/debian/changelog @@ -1,4 +1,573 @@ -sflphone-common (0.9.6-SYSVER) SYSTEM; urgency=low +sflphone-common (0.9.7~beta~ppa1~karmic) karmic; urgency=low + + ** 0.9.7~beta~ppa1~karmic ** + + * [#1933] Cleanup debug + * [#1933] Clean up debug + * Fix mic + * [#1933] Set the IAx format earlier + * [#1933] Move IAX sendAudioFromMic outside if (call) statement + * [#1933] Fix startstream when offhold in iax and add debug concerning + codec neg. + * [#2371] sflphone_notify_voice_mail: minor gettext message formatting + cleanup + * [#2371] select_account_cb: properly gettextize status message + * [#2371] show_account_list_config_dialog: properly gettextize status + message + * INSTALL: Minor tidyup of core install guide + * Add /sflphone-client-gnome/src/icons/Makefile to .gitignore + * [#2181] Updated OpenSUSE files (tmp) + * [#1933] Add debug for codec negociation for iax + * [#1933] Get rid of getMicAvail and getMicData in audiolayer (not + used anymore) + * [#1933] Add "audio codec not determined" error in IAX + * [#1933] Test flush data + * [#1933] Do not need to start audio stream in iax anymore + * [#1933] Protecting pointer + * [#2284] Remove more compilation/execution warnings + * [#2284] Cleanup debug in client, use DEBUG instead of g_print + * [#2284] Clean up uimanager + * [#2370] Remove warnings + * [#2366] Clean up other debug + * [#2366] Clean up debug + * [#2366] Call pa_xfree explicitely in writeToSpeaker + * [#2284] Remove address book warnings + * [#2365] Fixes bad cast + * [#2352] Fix continuous ringing when peer hangup and call not yet + answered + * [#2181] Added version support + * [#2181] Fixed some minor issues + * [#2360] Moved MainBuffer from AudioLayer to ManagerImpl + * [#2352] Makes getMainBuffer() everywhere + * [#2352] Use 50 sec latency on pulseaudio stream creation + * [#2352] Add alsa debug + * [#2359] Update repository documentation + * [#2354] Move pulseaudio disconnectAudioStream after stopping main + loop + * [#2352] Adjust nb byte copied in pulseaudio according to + writeableSize + * [#2352] Specify pulseaudio tlength parameters using pa_usec_to_bytes + * [#2322] Convert italian translation to UTF-8 + * [#2357] Fixes window size + * [#2357] Display only actionnable tool item + * [#2333] Update streams parameters + * [#2347] Use GNOME user settings for Menu and Toolbar appareance + * [#2349] Load/Save properly audio params + * [#2322] Update translations from Launchpad + * [#2181] Added Francois Marier script + * [#2350] Remove non-valid test + * [#2181] Updated launchpad packaging + * [#2333] Fix Pulseaudio Capture + * [#2333] Use pulseaudio ADJUST_LATENCY flag and ALSA RT-SCHEDULING + * [#2333] Pulseaudio Interpolate timing + * [#2333] Change (again) Pulseaudio settings to fit logiteck usb hdw + requirement + * [#2333] Adjust pulseaudio fragment size to 4096 (max sflphone's + frames per buffer) + * [#2284] Remove recurrent compilation warning (g++ linker problem) + * [#2333] Safer Audiostream parameters + * [#2333] Fix alsa playback to reduce underrun + * [#2333] Better audiostream parameters + * [#2181] Updated version management + * [#2333] Exclusive test in playback loop + * [#2181] Updated build system + * [#2333] Less underrun with these value + * [#2333] Update playback audiostream parameters + * [#2333] Lengthen the audio buffer reduce number of underrun in + pulseaudio + * [#2333] Add ALSA recovery functions for underrun (begin) + * [#2333] Add pa_stream_trigger in pulse audio underrun callabck + * [#2048] Reduce prebuffering in pulseaudio (which affect incomming + calls' plbck) + * [#2316] Do not display any icons to the right on the history tab + * [#2333] Comment pa_stream_trigger in pulseaudio underrun + * [#2333] Modify pulseaudio streams parameters + * [#2318] Fix transfer tool button double signal + * [#2181] Updated + * [#2333] Fix ALSA ringtone + * [#2333] Flush all main buffer before starting audio + * [#2333] Open/Close Alsa thread between calls while there is no audio + * [#2333] Add debug message and test condition on starting playback + and capture + * [#2181] Fixed gnome client makefile + * [#2181] Updated + * [#2308] Remove getTelephoneTone debug + * [#2308] Change plughw for default in ALSA + * [#2308] Oups, forgot to change function name in audiolayertest.cpp + * [#2308] Cleanup in pulseaudio code (debug, function name) + * [#2308] Fix pulseaudio stream closing assertion failure + * [#2308] Moved pulseaudio mainloop locking from AudioStream + disconnect stream + * [2308] Fix latency at the beginning of a call, when playing DTMF and + wehn starting tone + * [#2181] Updated karmic + * [#2317] [#2319] Fix address book toggle button contextual behaviour + * [#2308] Stop stream when refusing a call + * [#2308] Stop pulseaudio stream when peer hungup + * [#2308] Fix tone and ringtone + * [#2312] Display the STUN entry widget when opening the tab + * [#2308] Implement two different callbacks for capture/playback in + pulseaudio + * [#2309] Open/close pulseaudio connections in startStream/stopStream + * [2308] Leave pulseaudio stream running, do not cork/uncork them + anymore + * [#2295] Set gtk file chooser to None if nothing is set in + configuration + * [#1976] Add codec and conference documentation + * [#2209] Fix recording in regard of resamling + * [#2297] Update .gitignore + * [#2297] Update translation files + * [#2297] Add reference to our coding standards + * [#2297] Remove old docbook code + * [#2296] Reinit tls account settings after modification + * [#2253] Add DcBlocker class to remove capture's dc offset + * [#2034] Fixes for TLS transport to initialize + * [#2284] Add silent build rule + client clean warnings + * [#2274] Fix unserialize history items in cilent at startup + * [#2274] Complete display name parsing and displaying + * [#2274] Parse the Display Name in sip INVITE message + * [#2050] Fix capture volume control in ALSA + * [#1970] Volume controls disable when using pulseaudio + * [#1970] Disable volume controls when using pulseaudio + * [#2277] Fix direct ip2ip ZRTP enabling/disabling in ip2ip + preferences + * [#2181] Added launchpad debian files + * [#2181] Added spec files for OSC + * [#2274] Set display name for "Contact" sip header as the hostname + * [#2181] Fixed daemon issues + * [#2181] Fixed gnome client issues + * [#1976] Remove warnings - need to fix the transfer + * [#2006] Add init is_rec variable in ManagerImpl + * [#2006] Update codec display on call selection + * [#2006] Restore double click actions in history and contact calltree + (GTK) + * [#2176] use XDG_CACHE_HOME when initializing sfl.zid file + * [#1976] Fix calltree switching from history + * [#2209] (Re)Fix cache for zid + * [#2209] Clean up debug messages + * [#2209] Clean debug messages + * [#2209] Fix trasnfering a call during a conference + * [#2209] Speex decode must return the number of bytes + * [#2209] Change frameSize speex 32kHz + * [#2209] Fix speex codec framesize + * [#2209] Reinit converterSamplingRate in RTP sessions + * [#2209] Change speex ultra wide band framesize + * [#1747] Add pixmap data + * [#2252] Fix Receiving a server error 488 crashes the callee + * [#2209] Fix iax low rate packate sending + * [#2209] Clean up debug messages + * [#2209] Add resampling changes for IAX + * [#2209] Clean up resampling code + * [#2209] Fix latency introduced by pulseaudio + * [#2209] Fix initialization of mainbuffer's internal sampling rate + * [#2176] Fix upsampling buffer size in audiolayer + * [#2209] Add dynamic converter sampling rate in audiortp sessions + * [#1747] Fixes runtime warnings + * [#1747] Remove from repo + * [#1747] register our icons to be used as stock icons + * [#2209] Fix number of byte in alsa's write to speaker + * [#2209] Fix putting non-resampled data in RTP's mainbuffer + * [#2209] Add alsa resampler + * [#2209] Add a samplerate converter in PulseLayer + * [#2209] Add mainbuffer's internal sampling rate and flushall method + * [#2176] Add mainbuffer stateInfo debug method + * [#2209] Resampling is optimal using SRC_LINEAR not SRC_FASTEST + * [#2176] Remove debug recordings + * [#2176] Fix Holding a conference participant on new calls + * [#2224] Add confID in callable object + * [#2176] Fix putting onhold a call participating to a conference when + pressing new call + * [#2176] Reset auidio buffers when adding streams (rtp, audiolayer) + * [#1976] Use xml to describe toolbars - Add a naviguation toolbar + * [#2176] Remove conference default_id in joinParticipant + * [#2176] Display error message in alsa's snd_pcm_avail_update call + * [#2176] Alsa mic avail data debug + * [#2176] Add some debug message for mic loss problem + * [#2176] Flush mic ring buffer when offholding a call + * [#2176] Reset ringbuffers' readpointer when adding main participant + * [#2176] Fix getAvailData algorithm + * [#2176] Reset ringbuffer's readpointer when adding a new participant + to a conference + * [#1744] Regex object renamed to Pattern. Previous attempt at + providing + * [#2176] Fix detach main participant problem when adding new one + * [#1976] Use right domain to translate + * [#1976] Add xml menu description + * [#2176] Store a list of confernece participant in client + * [#2176] Fix add participant, joinparticipant methods + * [#2181] Do not install dbus-c++ headers + add return value + * [#2176] Fix minor call handling instabilities + * [#2174] Fix incoming IP call contact address + * [#2211] Add test to protect NULL pointer + * [#1163] Add Advanced account configuration section + * [#2176] Add some usefull comments and debugging info + * [#2176] Add conditions to display security icons in conference + * [#2176] Fix detaching one participant while keeping communication to + others + * [#2176] Reenable userActive.svg in call tree + * [#2176] Make user active blue (not red) + * [#2176] Fix user active picture + * [#2176] Fix "hidden" merge conflict in sipvoiplink + * [#2176] Remove iax audio stream on peer hungup + * [#2174] Multiple UDP transports functional (TESTED with 2 accounts + and 3 calls) + * [#2176] Fix fix audio stream binding in iax + * [#2174] Create a default UDP transport + use tp selector for dialogs + also + * [#2176] Register iax audio stream in mainbuffer + * [#2176] Fix getAudioCodecName in IAXvoipLink + * [#2176] Fix iax account init + * [#2176] Handle multiple account using the same sip transport + * [#2165] Add .png files + * [#2176] Small fixes concerning dtmf + * [#2176] Fix make uninstall in codecs + * [#2174] remove stund makefile generation + * [#2176] Add conference lock + * [#2174] Add transport selector for multiple accounts + * [#2176] Change userActive picture from red to blue + * [#2176] Fix security pixbuff in calltree + * [#2176] Replace sfl.zid in .cache/sflphone instead of .sflphone + * [#2176] Fix add call description + * [#2176] Remove detach button from toolbar + * [#2176] Fix calltree call description state and state code in + conferences + * [#2176] Fix pulse audio double free + * [#2176] Fix conference selection + * [#2174] Clean up - remove stun settings in client network + configuration panel + * [#2174] Remove voviva stun code + * [#2174] Rsolve STUN with pjsip - DO NOT WORK + * [#2165] Add user svg + * [#2165] Debugging sip call failed + * [#929] Link against uuid if installed + * Oops + * Fixed bugs related to libsexy (with GTK < 2.16) + * [#929] Remove uuid-dev dependency in the core + * [#2165] Debugging no negociated codecs at communicatio start + * [#2165] Fix calltree bug (gtktreestore instead of gtkliststore) + * [#2165] Fix several merge problems + * Updated opensuse packaging script + * [#1163] Add missing figures + * [#1163] Update INSTALL file + * [#2165] Fix IAX + * [#2165] Add recordabe interface + * [#2165] Finish recording refactoring for call (not for conference) + * [#2165] Enable speaker recording for two different calls + simultanously + * [#2165] Implement call recording using the Recordable interface + * [#2165] Add get and set to AudioLayer's audio recorder + * [#2165] Add class recordable from which inherit call and conference + * [#2006] Fix G722 and Speex 8khz codec conferencing + * [#2006] add recording of audio buffers + * [#1163] Add general settings section + * [#1163] Fixes makefile error + * [#2006] Fix some minor issues + * [#2006] Drag a conference call on another conference call + (difference conferences) + * [#2006] Fix dragging a conference on itself + * [#1744] Integrating some of the needed regular expression patterns + in order + * COmplete call features + * [#1744] Added support for named subgroup in the Regex object. Also, + new + * [#1744] Adds thread safety features, compile() and setPattern() + methods to the Regex class. + * [#1744] Fix inconsistency in the finditer method from the last + commit. + * [#1744] Added regex pattern object built on top of libpcre. To be + used + * [#1744] Initial commit towards implementing RFC4568. Unimplemented + in the + * [#2157] Hide "security" and "advanced" tabs for IAX under account + * [#1163] Add call features section + * [#2006] Add joinConference capabilities + * [#2006] Add dbus joinConference signal + * [#2006] Drag a conference call onto a conference to add it + * [#1163] Add addressbook section + * [#2006] Drag a conference call onto a single call to create a + conference + * [#2006] Expand rows automatically + * [#2006] Add minimal multiple conference handling + * [#2006] Add atached/detached conference icons + * [#2006] Add function processRemainingParticipant + * [#2006] Deep refactoring, fix hangup bug + * [#1163] Update documentation - Accounts part + * [#1976] Integrate user doc to gnome client build system + * [#2122] Remove double inclusion in dbus-c++/src/Makefile.am + * Remove pjproject version number + * [#2006] Fix peerHungup + * [#1976] Make Yelp accessible from the GNOME client (need to install + the sflphone.xml first) + * [#2006] Fix multiconferencing hangup + * [#2006] Fix hangup calls in a conference + * [#2150] Make IAx2 reappear + * [#2006] Fix detach participant on multiple call + * [#2006] Can remove rining call from a conference + * [#2006] Reinit confID when removing a participant + * [#2006] Remove get isCurrentCAll in hangup/peerhungup (SipVoipLink) + * [#2006] Fix refuse call + * [#2006] Fix answerring incoming call + * [#2006] Refactor conference's participant list + * [#2101] Re-integrate test compilation in main build system + * [#2101] Make the test directory compile + * [#2136] Restore history functionality + * [#2006] Fix binding main participant to himself + * [#2006] Fix add current/incoming/onHold participant to an existing + conference + * [#2006] Fix add incoming calls to an already created conference + * [#2006] Fix remove stream + * [#2006] Fix detachParticipant/removeParticipant switchCall ids + * [#2006] Fix adding a call in conference having state "CURRENT" + * [#2006] Remove/add main participant from conferences + * [#2006] Hold/unHold conference + * [#2006] Detach a partcipant from drag n drop + * [#2006] Hangup a conference + * [#2006] Add hold/unhold conference dbus messages + * [#2034] gtk-ui fix under the "basic" tab. + * [#2006] Fix dragging calls on conference calls + * [#2006] Fix detach participant from a conference + * [#2034] Added default message is status bar under the account config + dialog + * [#2112] Fix a crashed caused when a non-md5 password was sent to + pjsip. + * [#2006] Detach participant by ID + * [#2006] Fix addParticipant method in managerImpl to handle + incoming/answered calls + * [#2006] Add addParticipant method in managerimpl and related dbus + messages + * [#2111] Added the ability to configure zrtp on sip.sflphone.org from + * [#2106] Fixed problem in the account assistant under gtk-ui. Also, + assistant.c + * [#2006] Fix dragging a conference call on another conference call + (same conference) + * [#1904] Small UI fix. Assistant was moved from "Call" to "Edit" + menu. + * [#1904] Fix a wrong label under gtk-ui. + * [#2034] Renaming and source code splitting. + * [#2034] Status bar added to account window to better reflect the + registration + * [#2006] Make calltree_remove_call recursive (for GtkTreeStore) + * [#1110] Small gtk-UI fix in the account window (alignment). + * [#2006] Fix remove conference, display children which are still + active + * [#2006] Recursive function call in calltree_update_call + * [#2006] Add multilayered capabilities to calltree (GtkTreeStore) + * [#2006] Implement remove conference in calltree + * [#2034] Now useless as Direct Ip calls settings moved under + Preferences. + * [#2034] Edit/add buttons were set insensitive all the time under + gtk-ui. + * [#1887] Information about the state of the current SIP call is + displayed + * [#2006] Add call tree remove callback + * [#2006] Fix create_conference function + * [#2006] Update conference_added_cb to add new conference to the list + * [#812] Added new tab under GTK-ui Preferences. Moving Direct Ip + Calls from + * [#2121] Disable temporarily test compilation + * [#2006] Fix conferencelist to handle conference_obj_t instead of + gchar + * [#2006] Add conference_obj structure + * [#2121] Update version + * [#2006] Fix conference selection + * [#2101] Use the new source tree to fetch the right object files + * [#2006] Add conference in calltree + * [#2006] Add Dbus signal conference added/removed/changed + * [#2006] Add getConferenceDetails call on dbus + * [#1904] Registration expire now appears as a spin box under gtk-ui. + * [#812] Fixing a segmentation fault caused by a non-existing account + ID + * [#2006] Add getConfList method over dbus + * [#2006] Add a conferencelist data structure in client-gnome + * [#812] Defaults value are now sent if a non-existing account is + requested + * [#2006] Add sflphone action sflphone_join_participant + * [#2006] Fix buffer read pointer problem deletion + * [pjsip] Attempt at fixing via header incompatibility with + Freeswitch. + * [#1797] forget something + * [#2006] Add call new state conferencing in deamon + * [#2006] Remove addParticipant method for conference, use + joinParticipant only + * [#1163] Update INSTALL documentation + * [#812] Msec/sec values were not taken into account. + * [#1797] Make pjproject-1.4 compile + * [#2006] Add Detach participant method + * [#2006] Dragndrop fully functional with INCOMING and HOLD call + * [#1797] Add pjproject-1.4 + * [#1797] Remove pjproject-1.0.3 + * [#2006] Get call state in conference related function + * [#2006] Add joinParticipant (conference) method in ManagerImpl + * [#2006] Add joinConference DBUS message + * [#2006] Store the previously selected call_id on dragndrop + * [#2006] Fix GValue pointer unref in selection callback + * [#2006] Store dragged call_id + * [#2006] Update drag_data_received_cb callback to manipulate CallIDs + * [#2006] Add dragndrop signals + * [#2006] Set calltree reordable + * [#812] Adds the ability to create a TLS listener in case the user + requests + * [#812] Adds the ability to configure local/published address from + * [#1883] Move switchCall in onHoldCall function + * [#812] Deals with the published address/port problem when + integrating TLS. + * [#1883] Switch call id in managerimpl when peerHungUp + * [#1883] Switch call id before hangup + * [#1883] Add usefull and permanent debug info for conference + cretion/deletion + * [#812] Fix various segmentation faults related to Direct IP kind of + calls. + * [#1883] Fix deletion of std::map elements using iterators + * [#2014] Add libzrtpcpp build dependency + * [#1883] Still some for loop test ambiguity (while loop instead) + * [#1883] Fix for loop initial test ambiguity (use while loop instead) + * [#1883] We must discard data in urgent ring buffer if data is get in + mainbuf + * [#1883] Fix availForGet same id for ringbuffer and readpointer + * [#812] Match "sips" as a Direct IP Call when the user enter a sip + uri + * [#812] Fix segmentation fault related to SIP URI creation. + * [#812] Towards integrating multiple tls listeners at the same time. + This + * [#1883] Add debug messages in conference and fix mainbufferTest + * [#812] gkt-ui fix. Private key must be fed as a filename and not as- + is. + * [#812] TLS integration within sipvoiplink and pjsip. Also, + configure.ac + * [#1883] Fix Alsa/Pulse mallocation + * [#1883] Fix data corruption in AudioRtp's micData buffer + * [#812] Full dbus integration for all the tls related options under + gtk-ui. + * [#1883] Fix memory leaks in audiortp session + * [#1883] Fix mem leaks in audio rtp + * [#812] Fix setAccountDetails where TLS_ENABLE was set to the value + * [#812] Small gtk-ui fix. + * [#811][#812] Small gtk-ui fix. + * [#812] Introduced a mechanism for configuration files that makes + possible + * [#812] New dbus bindings added. Also, configuration compliance was + enforced + * [#1881] Remove default buffer from MainBuffer (update unit-tests) + * [#1881] Add ring buffer read pointer tests + * [#1883] Fix issues in ringbuffer reader pointers + * [#2034] Implementing a new configuration dialogue for TLS transport + settings + * [#1883] Add some usefull debug and safety checks + * [#2028] Notify the client with libnotify when the zrtp negotiation + failed. + * [#811] Harmless no to throw an exception, an makes the application + less + * [#2028] A minidialog is showed to the user under sflphone-client- + gnome + * Removed useless file. + * Ignoring Makefile in src/widget + * [#2027] Fix segmentation fault when showMessage callback is called + after + * [#2026] keyExchange was set to ZRTP instead of "1" + * [#2024] Fix the wrong summary at the end of the assistant. + * [#1883] Fix mnagerimpl conference map insertion + * [#1883] Add Mutexes in MainBuffer + * [#811] Gtk ui was not presenting the right information about zrtp + for + * [#2023] security icons were not installed in sflphone-client-gnome. + * [#2021] Fix a mistake in the readme from sflphone-common that gives + wrong + * [#811] The current SRTP mode was not properly displayed for the + IP2IP + * [#1743] Re-implementation of the "automatically remove error dialogs + [...]" + * [#2017] [#2019] Fix the inability to dial a number and place a + registered + * [#811] Final re-integration of ZRTP support in the main branch from + 0.9.6 + * [#1883] Fix map insertion methods + * [#811] Combo box now is now set to the active key exchange method + * [#811] ZRTP options now configurable back again from the Gtk UI. + IP2IP + * Updated hostname for git clone + * [#1883] Add minimal functionalities to create a conference + * [#811] re-integration of all the methods and signals on dbus. + ManagerImpl + * [#811] Got out of a precarious position were nothing would compile. + * [#1976] Build documentation squeleton with docbook + * [#1883] Add sflphone-client "addParticipant" button for conference + * [#1994] Better organize the source directory structure. New + subdirectories + * [#1883] Add a simple Conference class + * [#1882] Use static audio buffer in Pulse and ALSA layer (instead of + malloc) + * [#811] First commit toward re-integration and refactoring of ZRTP + * [#1882] Flush RTP ring buffer before entering mainloop + * [#1882] Fixed MainBuffer::UnBinCallID() in case there is no + ringbuffer + * [#1882] Test (and fixe) high level conference and mixing + functionalities + * [#1772] Apply patch to compile on fedora (sent by Marcin + Zajączkowski <mszpak@wp.pl>) + * [#1882] Update Bind, unBind call_id in MainBuffer + * [#1959] This adds the ability to store password as an MD5 Hash in + the + * [#1538] Fixes rules compilation + * [#1930][#1931] Fixed a mistake (again) related to index and + credential count + * [#1753] Remove ILBC from pjproject - Hacks in pjsip + * [#1930][#1931] Credential was not selected properly using realm + * [#1882] Finilize multiple reading pointer in RingBuffer + * [#1538] Remove configure from autogen.sh to respect debian upstream + authors policy + * [#1773] Remove generated files from repo + * [#1791] Use XDG_CACHE_HOME to save pid file + * [#1791] Fixes path to save history + * [#1791] Fix debian installation scripts + * [#1930][#1931] Settings are now taken into account in the server. + * [#1882] Add ringbuffer default ring buffer pointer in methods + involving mStart + * [#1882] Add default ringbuffer pointer + * [#1882] Add RingBuffer multiple read pointer basic functionnalities + * [#1882] Fix MainBuffer flushData unit test + * [#1930][#1931] Ability to save and retreive the configuration from + * [#1882] Added Multiple CallID mapping to MainBuffer + * [#1791] Not much + * [#1791] If XDG env variables are not null but empty, use default + ones + * [#1791] Make XDG_CONFIG_HOME writable + * [#1930][#1931] Partial commit. Not working yet. Cannot delete + account + * [#1881] Fixed alsa capture latency problem + * [#1881] Fixed Alsa capture temporarily + * [#1930] [#1931] Partial unbroken commit providing the ability to + * [#1881] MainBuffer implemented in AudioLayer/AudioRTP + * [#1881] Add discard and flush unit-tests + * [#1881] Add discard and flush functionnalites to MainRingBuffer + * [#1881] Add availForGet in MainBuffer + * [#1881] Add availForPut function to MainBuffer + * [#1880] Remove AudioRTP* pointer from SipVoIP (reapered while + merging master) + * [#1881] Add a map between call id and coresponding ring buffer + * [#1855] Refresh pot file and upload on Launchpad + * [#1881] MainBuffe now robust to false ids on getData and putData + * [#1881] Fix big big big memory leak + * [#1881] Add getData and putData to mainBuffer + * [#1881] Unit-test basic ring buffer functionnaities + * [#1881] Add class MainBuffer and basic buffer creation unit-tests + * [#1880] Fix call transfer (step2) issues + * [#1880] Moved AudioRtp* pointer from SIPVoIPLink to SIPCall class + * [#1791] Add postinst script to keep user data when migrating + config/history file + * [#1797] Make pjsip compile + * [#1777] Code indentation + * [#1791] Use XDG_DATA_HOME and XDG_CONFIG_HOME for sflphonedrc and + history + unit tests + * [#1746] Useless space does not appear anymore when volume sliders + and + * [#1643] GtkCheckMenuItem is used instead of icons for elements in + the + * [#1110] [#1668] STUN parameters are now located in the preferences, + under + + -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 06 Nov 2009 11:23:15 -0500 + +sflphone-common (0.9.6-SYSVER) karmic; urgency=low ** 0.9.6 ** @@ -65,7 +634,7 @@ sflphone-common (0.9.6-SYSVER) SYSTEM; urgency=low -- SFLphone Automatic Build System <team@sflphone.org> Mon, 27 Jul 2009 09:53:00 -0400 -sflphone-common (0.9.6~rc2-SYSVER) SYSTEM; urgency=low +sflphone-common (0.9.6~rc2-SYSVER) karmic; urgency=low ** 0.9.6~rc2 ** @@ -120,7 +689,7 @@ sflphone-common (0.9.6~rc2-SYSVER) SYSTEM; urgency=low -- SFLphone Automatic Build System <team@sflphone.org> Fri, 17 Jul 2009 19:12:44 -0400 -sflphone-common (0.9.6~rc1-SYSVER) SYSTEM; urgency=low +sflphone-common (0.9.6~rc1-SYSVER) karmic; urgency=low ** 0.9.6~rc1 ** @@ -228,7 +797,7 @@ sflphone-common (0.9.6~rc1-SYSVER) SYSTEM; urgency=low -- SFLphone Automatic Build System <team@sflphone.org> Fri, 10 Jul 2009 15:49:56 -0400 -sflphone-common (0.9.6~beta-SYSVER) SYSTEM; urgency=low +sflphone-common (0.9.6~beta-SYSVER) karmic; urgency=low ** 0.9.6~beta ** @@ -523,7 +1092,7 @@ sflphone-common (0.9.6~beta-SYSVER) SYSTEM; urgency=low -- SFLphone Automatic Build System <team@sflphone.org> Tue, 23 Jun 2009 11:12:06 -0400 -sflphone-common (0.9.5-SYSVER) SYSTEM; urgency=low +sflphone-common (0.9.5-SYSVER) karmic; urgency=low ** 0.9.5 release ** @@ -554,7 +1123,7 @@ sflphone-common (0.9.5-SYSVER) SYSTEM; urgency=low -- SFLphone Automatic Build System <team@sflphone.org> Mon, 25 May 2009 11:34:40 -0400 -sflphone-common (0.9.5-SYSVER~rc2) SYSTEM; urgency=low +sflphone-common (0.9.5-SYSVER~rc2) karmic; urgency=low ** 0.9.5 rc2 ** @@ -608,7 +1177,7 @@ sflphone-common (0.9.5-SYSVER~rc2) SYSTEM; urgency=low -- SFLphone Automatic Build System <team@sflphone.org> Tue, 19 May 2009 12:08:03 -0400 -sflphone-common (0.9.5-0ubuntu1~rc1) SYSTEM; urgency=low +sflphone-common (0.9.5-0ubuntu1~rc1) karmic; urgency=low [ SFLphone Project ] * [#1262] Updated changelogs for version 0.9.5-0ubuntu1 Snapshot 2009- @@ -637,7 +1206,7 @@ sflphone-common (0.9.5-0ubuntu1~rc1) SYSTEM; urgency=low -- Sflphone Project <sflphone@mtl.savoirfairelinux.net> Tue, 05 May 2009 19:16:09 -0400 -sflphone-common (0.9.5-0ubuntu1~beta) SYSTEM; urgency=low +sflphone-common (0.9.5-0ubuntu1~beta) karmic; urgency=low [ Julien Bonjean ] * Updated Eclipse stuff @@ -859,7 +1428,7 @@ sflphone-common (0.9.5-0ubuntu1~beta) SYSTEM; urgency=low -- Sflphone Project <sflphone@mtl.savoirfairelinux.net> Mon, 27 Apr 2009 16:57:00 -0400 -sflphone-common (0.9.4-0ubuntu2) SYSTEM; urgency=low +sflphone-common (0.9.4-0ubuntu2) karmic; urgency=low [ Alexandre Savard ] * Restore speex and GSM detection @@ -869,7 +1438,7 @@ sflphone-common (0.9.4-0ubuntu2) SYSTEM; urgency=low -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 8 Apr 2009 11:29:15 -0500 -sflphone (0.9.4-0ubuntu1) SYSTEM; urgency=low +sflphone (0.9.4-0ubuntu1) karmic; urgency=low [ Emmanuel Milou ] * Integrate DBus-c++ and libiax2 in the main build system @@ -894,7 +1463,7 @@ sflphone (0.9.4-0ubuntu1) SYSTEM; urgency=low -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 3 Apr 2009 18:29:15 -0500 -sflphone (0.9.4-rc1) SYSTEM; urgency=low +sflphone (0.9.4-rc1) karmic; urgency=low [ Emmanuel Milou ] * Fix bug while trying to hold/unhold several simultaneous call @@ -908,7 +1477,7 @@ sflphone (0.9.4-rc1) SYSTEM; urgency=low -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 27 Mar 2009 18:29:15 -0500 -sflphone (0.9.4-0beta1) SYSTEM; urgency=low +sflphone (0.9.4-0beta1) karmic; urgency=low [ Alexandre Savard ] * Display codec used during conversation on the GUI @@ -924,7 +1493,7 @@ sflphone (0.9.4-0beta1) SYSTEM; urgency=low -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 20 Mar 2009 18:29:15 -0500 -sflphone (0.9.3-0ubuntu3) SYSTEM; urgency=low +sflphone (0.9.3-0ubuntu3) karmic; urgency=low [ Alexandre Savard ] * Both playback and record streams in PA_STREAM_CORKED (pulseaudio) @@ -949,7 +1518,7 @@ sflphone (0.9.3-0ubuntu3) SYSTEM; urgency=low -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Sat, 14 Feb 2009 13:29:15 -0500 -sflphone (0.9.3-0ubuntu2) SYSTEM; urgency=low +sflphone (0.9.3-0ubuntu2) karmic; urgency=low [ Emmanuel Milou ] * Add compilation note in README @@ -1018,7 +1587,7 @@ sflphone (0.9.3-0ubuntu2) SYSTEM; urgency=low -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Sat, 14 Feb 2009 13:29:15 -0500 -sflphone (0.9.3-0ubuntu1) SYSTEM; urgency=low +sflphone (0.9.3-0ubuntu1) karmic; urgency=low * Remove debug * Join thread before leaving @@ -1031,7 +1600,7 @@ sflphone (0.9.3-0ubuntu1) SYSTEM; urgency=low -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 06 Feb 2009 19:17:32 -0500 -sflphone (0.9.2-2ubuntu9) SYSTEM; urgency=low +sflphone (0.9.2-2ubuntu9) karmic; urgency=low [ Alexandre Savard ] * Speex audio codec preprocessing initialization @@ -1059,7 +1628,7 @@ sflphone (0.9.2-2ubuntu9) SYSTEM; urgency=low -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Thu, 05 Feb 2009 18:27:53 -0500 -sflphone (0.9.2-2ubuntu8) SYSTEM; urgency=low +sflphone (0.9.2-2ubuntu8) karmic; urgency=low [ Emmanuel Milou ] * Update changelogs @@ -1103,7 +1672,7 @@ sflphone (0.9.2-2ubuntu8) SYSTEM; urgency=low -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 04 Feb 2009 14:00:30 -0500 -sflphone (0.9.2-2ubuntu7) SYSTEM; urgency=low +sflphone (0.9.2-2ubuntu7) karmic; urgency=low [ Emmanuel Milou ] * Update changelog to 0.9.2-6 @@ -1125,7 +1694,7 @@ sflphone (0.9.2-2ubuntu7) SYSTEM; urgency=low -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Fri, 23 Jan 2009 17:48:13 -0500 -sflphone (0.9.2-2ubuntu6) SYSTEM; urgency=low +sflphone (0.9.2-2ubuntu6) karmic; urgency=low [ Emmanuel Milou ] * Migrate STUN configuration to the main config window @@ -1159,7 +1728,7 @@ sflphone (0.9.2-2ubuntu6) SYSTEM; urgency=low -- Emmanuel Milou <manu@sulfur.inside.savoirfairelinux.net> Fri, 16 Jan 2009 18:19:05 -0500 -sflphone (0.9.2-2ubuntu5) SYSTEM; urgency=low +sflphone (0.9.2-2ubuntu5) karmic; urgency=low * Fix memory leak in the pulseaudio callback * Update debian package generation script @@ -1175,7 +1744,7 @@ sflphone (0.9.2-2ubuntu5) SYSTEM; urgency=low -- Emmanuel Milou <emmanuel.milou@savoirfairelinux.com> Wed, 14 Jan 2009 21:17:20 -0500 -sflphone (0.9.2-2ubuntu4) SYSTEM; urgency=low +sflphone (0.9.2-2ubuntu4) karmic; urgency=low [ Emmanuel Milou ] * add german translation @@ -1185,7 +1754,7 @@ sflphone (0.9.2-2ubuntu4) SYSTEM; urgency=low -- Yun Liu <yun.liu@savoirfairelinux.com> Thu, 08 Jan 2009 13:08:51 -0500 -sflphone (0.9.2-2ubuntu3) SYSTEM; urgency=low +sflphone (0.9.2-2ubuntu3) karmic; urgency=low [ Emmanuel Milou ] * The main thread synchronizes the ringtone thread @@ -1197,13 +1766,13 @@ sflphone (0.9.2-2ubuntu3) SYSTEM; urgency=low -- Yun Liu <yun.liu@savoirfairelinux.com> Tue, 06 Jan 2009 16:18:38 -0500 -sflphone (0.9.2-2ubuntu2) SYSTEM; urgency=low +sflphone (0.9.2-2ubuntu2) karmic; urgency=low * Fix bug ticket #129 -- Yun Liu <yun.liu@savoirfairelinux.com> Wed, 5 Jan 2009 15:54:53 -0500 -sflphone (0.9.2-2ubuntu1) SYSTEM; urgency=low +sflphone (0.9.2-2ubuntu1) karmic; urgency=low * Migrate from eXosip library to pjsip * Add multiple SIP accounts support