diff --git a/sflphone-client-gnome/doc/C/figures/accounts_security.png b/sflphone-client-gnome/doc/C/figures/accounts_security.png
new file mode 100644
index 0000000000000000000000000000000000000000..ca3f76c3d9a899589dc29b299a3117ae8ebe110f
Binary files /dev/null and b/sflphone-client-gnome/doc/C/figures/accounts_security.png differ
diff --git a/sflphone-client-gnome/doc/C/figures/srtp_enabled.png b/sflphone-client-gnome/doc/C/figures/srtp_enabled.png
new file mode 100644
index 0000000000000000000000000000000000000000..68dabf1d368e614c3bf2326a0d8331eea4736154
Binary files /dev/null and b/sflphone-client-gnome/doc/C/figures/srtp_enabled.png differ
diff --git a/sflphone-client-gnome/doc/C/figures/zrtp_options.png b/sflphone-client-gnome/doc/C/figures/zrtp_options.png
new file mode 100644
index 0000000000000000000000000000000000000000..b9229d7b81f31aa6e4a66759bd1856959c82f233
Binary files /dev/null and b/sflphone-client-gnome/doc/C/figures/zrtp_options.png differ
diff --git a/sflphone-client-gnome/doc/C/sflphone.xml b/sflphone-client-gnome/doc/C/sflphone.xml
index b104b8b2f8493a28a289d5a41f861170ddbadcf5..c08d76564af07be3415038c59a46aa44ea01c03e 100644
--- a/sflphone-client-gnome/doc/C/sflphone.xml
+++ b/sflphone-client-gnome/doc/C/sflphone.xml
@@ -270,15 +270,31 @@
</sect2>
<sect2 id="account_security">
<title>Security features</title>
- <para>These features are only available with SIP.</para>
- <sect3 id="accounts_zrtp">
- <title>Secure RTP</title>
+ <para>These features are only available with SIP.</para>
+ <para>Follow the indications to <link linkend='account_edit'>edit an account</link> and choose the <guimenu>Security</guimenu> tab.</para>
+ <!-- ==== Figure ==== -->
+ <figure id="account-security-fig">
+ <title>Security features configuration panel</title>
+ <screenshot>
+ <mediaobject>
+ <imageobject>
+ <imagedata fileref="figures/accounts_security.png" format="PNG"/>
+ </imageobject>
+ </mediaobject>
+ </screenshot>
+ </figure>
+ <!-- ==== End of Figure ==== -->
+
+ <sect3 id="realms">
+ <title>Credentials</title>
+ <para>SFLphone supports multiple realms.</para>
</sect3>
- <sect3 id="accounts_tls">
- <title>TLS</title>
+
+ <sect3 id="security_frame">
+ <title>Security </title>
+ <para>Please refer to the section <link linkend="detailed_security_features">Security features</link> for detailed information about security features.</para>
</sect3>
</sect2>
-
</sect1>
<sect1 id="call_features">
@@ -607,6 +623,109 @@
</sect1>
+<sect1 id="detailed_security_features">
+ <title>Security features</title>
+ <sect2 id="zrtp_srtp">
+ <title>SRTP/ZRTP</title>
+ <sect3 id="zrtp_srtp_definition">
+ <title>SRTP and ZRTP, the big picture</title>
+ <para>RTP is the underlying protocol that is used in pair with the widely used SIP protocol to carry voice data. RTP alone does not provide any security features.</para>
+ <para>Details for implementing Secure RTP (SRTP) were described independently in a separate document (RFC). However, in this paper, one aspect was deliberately left unspecified: how should the encryption keys be exchanged between the two parties involved in a secure RTP session ?</para>
+
+ <para>Mutiple solutions were proposed to fill in that blank. Among them, are SDES (RFC4568) and ZRTP which are probably the most popular today. For the 0.9.7 release, SFLphone integrates support for Secure RTP through the ZRTP protocol, and SDES is expected to be implemented in the very few next releases.</para>
+
+ <para>As of today, blueprints for ZRTP are still laid out and are recognized under the name "zrtp-draftzimmerman" in the RFC machine. The author of ZRTP is Phil Zimmermann, that same person who brought us PGP. Therefore, it is not suprising that he designed ZRTP as an anti-PKI solution for key exchange.</para>
+
+ <para>ZRTP makes possible for two parties to automatically establish a shared secret in a very simple way from the users's point of view. Indeed under SFLphone no special configuration is needed, appart from enabling the option itself.</para>
+
+ <para>If you want to use ZRTP, please take note that if you are connecting to a PBX, this one must have been configured to support ZRTP. Unfortunately, security for VoIP communications is still young and chances are that your PBX software won't support it.</para>
+
+ <para>This does not mean that you want be able to benefit from ZRTP ! In fact, it turns out that you will be able to use it, as long as the server does not need to decode the RTP stream. This is often the case when the person you are calling to uses a codec that you don't support. In that case, the server will need to transcode the RTP packets and obviously need to be able to handle the ZRTP stream.</para>
+
+ <para>Obviously, if you are calling another user (for example by prefixing the number with "sip:") directly, then this one will have to support ZRTP as well if you want to use it.</para>
+ </sect3>
+
+ <sect3 id="enabling_srtp">
+ <title>Enabling SRTP/ZRTP</title>
+ <para>To enable ZRTP per account basis, perform the following steps:</para>
+ <orderedlist>
+ <listitem><para>Choose <menuchoice><guimenu>Edit</guimenu><guimenuitem>Manage accounts</guimenuitem></menuchoice>.</para></listitem>
+ <listitem><para>Select in the list the account you would like to edit, then click on the <guilabel>Edit</guilabel> button.</para></listitem>
+ <listitem><para>Select the <guilabel>Security</guilabel> tab.</para></listitem>
+ <listitem><para>Select <guilabel>ZRTP</guilabel> from the select box named <guilabel>SRTP Key Exchange</guilabel>.</para></listitem>
+ </orderedlist>
+ <!-- ==== Figure ==== -->
+ <figure id="srtp-enabled-fig">
+ <title>Enabling SRTP</title>
+ <screenshot>
+ <mediaobject>
+ <imageobject>
+ <imagedata fileref="figures/srtp_enabled.png" format="PNG"/>
+ </imageobject>
+ </mediaobject>
+ </screenshot>
+ </figure>
+ <!-- ==== End of Figure ==== -->
+ </sect3>
+
+ <sect3 id="account_zrtp">
+ <title>Configuration options</title>
+
+ <para>After enabling SRTP, click the <guilabel>Preferences</guilabel> button.</para>
+ <para>For basic usage, one don't have to worry about that.</para>
+
+ <!-- ==== Figure ==== -->
+ <figure id="zrtp-options-fig">
+ <title>ZRTP configuration panel</title>
+ <screenshot>
+ <mediaobject>
+ <imageobject>
+ <imagedata fileref="figures/zrtp_options.png" format="PNG"/>
+ </imageobject>
+ </mediaobject>
+ </screenshot>
+ </figure>
+ <!-- ==== End of Figure ==== -->
+
+
+ <variablelist>
+ <varlistentry>
+ <term><guilabel>Send Hello Hash in SDP</guilabel></term>
+ <listitem><para>Selecting this option will cause the program to compute an hash function over the "Hello" packet and send it as an SDP field "zrtp-hash:". The remote end might be interested in getting this value to add an additional layer of protection based on another communication channel. Upon receiving this value, the remote point can compute the hash function on the received hello packet and compare it.</para>
+ <para>Take note that for 0.9.7, SFLPhone does not perform the comparasion on its side.</para></listitem>
+ </varlistentry>
+
+ <varlistentry>
+ <term><guilabel>Ask user to confirm SAS</guilabel></term>
+ <listitem><para>The short authentication mechanism is at the heart of the ZRTP protocol. Not requirering the user to manually check the SAS value presents a security risk over Man in the Middle type of attacks.</para>
+
+ <para>Disabling this option will stop the program from prompting the user with the SAS.</para>
+
+ <para>Such an option was motivated to be developped at that time by the the state of the libzrtpcpp library that SFLPhone was making use of. It is only from version x.x that this library can cache results of SAS computation between two peers.</para>
+ </listitem>
+ </varlistentry>
+
+ <varlistentry>
+ <term><guilabel>Display SAS once for hold event</guilabel></term>
+ <listitem><para>When call is put on hold, the RTP stream is stopped and reinitiated later. From the ZRTP point of view, this appears as a "new call". Therefore, the SAS will be redisplayed unless this option is selected.</para></listitem>
+ </varlistentry>
+ <varlistentry>
+ <term><guilabel>ZRTP for direct peer-to-peer calls</guilabel></term>
+ <listitem><para>If you want to use ZRTP for calls that are placed directly to a user (without an intervening PBX), you must enable the option under the "Direct IP Calls" tab in the "configuration" window, available from the "edit" menu.</para>
+
+ <para>Configuration instruction from that point are the same as for configured accounts.</para>
+ </listitem>
+ </varlistentry>
+ </variablelist>
+ </sect3>
+ </sect2>
+ <sect2 id="accounts_tls">
+ <title>TLS</title>
+ </sect2>
+</sect1>
+
+
+
<sect1 id="audio_interfaces">
<title>Audio configuration</title>
<para>
diff --git a/sflphone-client-gnome/doc/Makefile.am b/sflphone-client-gnome/doc/Makefile.am
index 2993072194d2eb793d34df550ab2eb2439f2b3ca..6727048ed8b2627be0c82e25f8555fd0cb40c51f 100644
--- a/sflphone-client-gnome/doc/Makefile.am
+++ b/sflphone-client-gnome/doc/Makefile.am
@@ -24,8 +24,11 @@ DOC_FIGURES = figures/addressbook-button.png \
figures/systemtray-settings.png \
figures/voicemail-notif.png \
figures/account_advanced.png \
+ figures/accounts_security.png \
figures/drag_n_drop.png \
figures/conference.png \
figures/conference_detached.png \
- figures/conference_attached.png
-DOC_LINGUAS = fr de es it zh_TW zh_HK zh_CN ko pl pt_BR pt ru
+ figures/conference_attached.png \
+ figures/srtp_enabled.png \
+ figures/zrtp_options.png
+DOC_LINGUAS = fr es
diff --git a/sflphone-client-gnome/po/Makefile.am b/sflphone-client-gnome/po/Makefile.am
index 6d2d320a9b79675cc62abab8c0a919be3dd0da04..2f8eebd1ccf3f00a642c4b16581e2af010f67dbb 100644
--- a/sflphone-client-gnome/po/Makefile.am
+++ b/sflphone-client-gnome/po/Makefile.am
@@ -38,7 +38,7 @@ SUFFIXES=.po .mo
.po.mo:
$(MSGFMT) -o $@ $<
-install-data-local:
+install-data-local: $(MOFILES)
@catalogs='$(MOFILES)'; \
for i in $$catalogs; do \
destdir=$(locale_installdir); \
@@ -58,4 +58,4 @@ uninstall-local:
done
clean-local:
- rm -rf *.mo
+ rm -f $(MOFILES)
diff --git a/sflphone-common/configure.ac b/sflphone-common/configure.ac
index e3c869ac137efc92d95629e957910492f57c93b6..41bfe9221b850eb4ac988e88fd260feb978e5365 100644
--- a/sflphone-common/configure.ac
+++ b/sflphone-common/configure.ac
@@ -94,7 +94,6 @@ AC_SUBST(PKGADD_VENDOR)
dnl Check for programs
AC_PROG_CC
- SFL_CXX_WITH_DEBUG
AC_PROG_CXX
AC_PROG_CPP
AC_PROG_INSTALL
diff --git a/tools/build-system/build-osc.sh b/tools/build-system/build-osc.sh
index 30fb9ab3c0e319730d6104fbf53f42d62c684fdc..8735ba83dfde2e019d8033e173f3676ac67cc4af 100755
--- a/tools/build-system/build-osc.sh
+++ b/tools/build-system/build-osc.sh
@@ -23,7 +23,7 @@ LAUNCHPAD_PACKAGES=( "sflphone-client-gnome" "sflphone-common" )
REFERENCE_REPOSITORY="${ROOT_DIR}/sflphone-source-repository"
-SOFTWARE_VERSION="0.9.7.beta"
+SOFTWARE_VERSION="0.9.7.rc2"
VERSION_INDEX=1
diff --git a/tools/build-system/launch-build-machine-2.sh b/tools/build-system/launch-build-machine-2.sh
index 46246540a089f1de84f40e24e810d3622bb70566..b6b7638195aa7152ec9585dc24cfd9caed0d023d 100755
--- a/tools/build-system/launch-build-machine-2.sh
+++ b/tools/build-system/launch-build-machine-2.sh
@@ -217,6 +217,7 @@ END
fi
done
+ cp ${DEBIAN_DIR}/changelog.generic ${DEBIAN_DIR}/changelog
done
# if push is activated
diff --git a/tools/build-system/launchpad/sflphone-client-gnome/debian/changelog b/tools/build-system/launchpad/sflphone-client-gnome/debian/changelog
index 0fc72e29660477e57c279ed29a034962687e2f10..8021fb0d7d0b35eb2d8330c906baee884e297e43 100644
--- a/tools/build-system/launchpad/sflphone-client-gnome/debian/changelog
+++ b/tools/build-system/launchpad/sflphone-client-gnome/debian/changelog
@@ -1,3 +1,162 @@
+sflphone-client-gnome (0.9.7~rc1~ppa1~SYSTEM) SYSTEM; urgency=low
+
+ ** 0.9.7~rc1~ppa1~SYSTEM **
+
+ * [#2462] Set explicitly the transport on incoming call too
+ * [#2462] fix typo
+ * [#2462] Use different address for SDP and call IP
+ * [#2462] Use published address in SIP-SDP
+ * [#2181] Fixed changelog files
+ * [#2181] Updated spec file
+ * [#2402] Fix pointer to int conversion warning (atoi)
+ * [#2402] Remove daemon warnings, make indent
+ * [#2459] Make sure the stream is opened when the call is answered
+ * [#2402] Add conference related picture in documentation
+ * [#2443] Not much ...
+ * [#2399] Fix dialing display problem
+ * [#2450] Fix incoming call already in conference crash
+ * [#2399] Display peer name on the first line and peer number on the
+ second
+ * [#2450] Handle 403 FORBIDDEN when refused
+ * [#2447] Bind offHold/onHold actions to button in gtk client
+ * [#2447] Bind hangup action to button for conference
+ * [#2447] Add conference action in gtk client's ToolBar
+ * [#2381] Disable the password hashing in config file
+ * [#2402] Cleanup
+ * [#2366] Set callback to null when deleting Pulseaudio streams
+ * [#1313] Fix main buffer unit test
+ * [#1313] Fix audio layer unit test
+ * [#2315] Hide pw in security tab, display when editing, sync with
+ basic tab
+ * [#1313] UnitTest change AudioRtpSession for AudioSymetricRtpSession
+ instance
+ * [#2402] Code cleanup
+ * [#2444] Add debug to catch occasional crash when loading client's
+ config
+ * [#2444] Add debug info to catch occasional crash when loading config
+ dialog
+ * [#2402] Restore Call menu translations
+ * [#2403] Use the published address if checked in GUI
+ * [#2442] Add protection test in sdp
+ * [#1841] Reapply pjsip patch concerning DNS SRV resolution
+ * [#2384] Tags incoming call as direct SIP call, if applicable
+ * [#2402] Change the monkey face
+ * [#2315] Enable user to display password in clear text
+ * [#2434] Force optimization level at 2
+ * [#2284] Fix dbus_get_all_ip_interface compilation warnings
+ * [#2431] Popup main window on incoming if applicable
+ * [$2402] Fix simple warnings
+ * [#2402] Fix implicit variable init order in LibraryManagerException
+ * [#2402] Fixing implicit variable initialization warnings in
+ AudioRtpSession
+ * [#2402] Revert atoi change, fixing codec list doubled entries
+ * [#2402] Fix gpointer to gint conversion
+ * [#2402] Fix pointer casting to integer different size warning in
+ codec list
+ * [#2402] Fix warning discarting qualifiers from pointer target
+ * [#2402] Fix gtk tree view assignement from incompatible type warning
+ * [#1669] Fix audio recording folder utf-8 non compatibility issue
+ * [#2414] Clean up debugs
+ * [#2414] Use transport set in iptoip Account and update it frm
+ preference
+ * [#2348] Use macro N_() to mark ui.xml strings as translatable
+ * [#2414] Rename getSipAddress/setSipAddress functions
+ * [#2407] Fix volume controls display
+ * [#2407] Fixes dialpad
+ * [#2383] Set ip to ip config when clicking apply button
+ * [#2404] Update call-to script - Maxime Chambreuil
+ * [#2405] Client handles unknown call in current state as well
+ * [#2383] Add DBUS signal to send IPtoIP local address and port as
+ string
+ * [#2383] Add Ip to IP config change apply call back
+ * Clonflict
+ * [#2402] Code cleanup
+ * [#2383] Do the same for IPtoIP (init localn ip with first in the
+ list)
+ * [#2383] Use first interface in the list if local addresss is not
+ defined
+ * [#2403] Clean up unuseful addresses/ports
+ * [#2403] Use the IP profile SIP port as global SIP port
+ * [#2383] Fix dbus_get_all_ip_interface warnings
+ * [#2383] Take into account sameAsLocal when loading published address
+ * [#2383] Tsake into account sameAsLocal option when saving published
+ address
+ * [#2383] Update local ip address in ip to ip config
+ * [#2383] Save ip 2 ip local port in config
+ * [#2406] Update toolbar at startup
+ * [#2284] Remove redefinition warnings + speex warnings
+ * [#2383] Fix security table in account config
+ * [#2383] Save ip 2 ip network interface parameters in config
+ * [#2403] Restore sip transport selector
+ * [#2383] Fix filling the Localt IP Address on account creation
+ * [#2383] Fix Gtk-Critical when checking STUN
+ * [#2383] Fix reopening account configuration display issue
+ * [#2383] Load IPtoIP local address and port in preference iptoiptab
+ * [#2383] Add LocalAddress and Localport in Preference IpToIp tab
+ * [#2403] Use the address and port associated to the account as often
+ as possible
+ * [#1753] Removed pjsip generated files
+ * [#1753] Removed remaining milenage lib references
+ * [#2383] Add _publishedSameasLocal variable in sipaccount
+ * [#2383] Add PUBLISHED_SAMEAS_LOCAL variable in config
+ * [#2383] Fix stun set active or not when opening config
+ * [#2181] Added RPM 64bits dbus patch
+ * [#2402] Code indentation
+ * [#2313] Force $(HOME).cache directory creation at startup
+ * [#2383] Separate network interface and published address in account
+ config
+ * [#2400] Change dbus service installation path to libdir
+ * [#2382] Move TLS related published address options in security tab
+ * [#2382] Indent accountconfigdialog.c
+ * [#2181] Install libdbus-c++ in $pkglib instead of $lib
+ * [#1753] Remove ILBC code and disable it by default in the configure
+ * [#1753] Remove milenage directory
+ * [#2382] Fix switching interaface instabilities
+ * [#2396] Save local ip in account creation wizard
+ * [#2284] Remove warning on hold
+ * [#2387] Fixes history searching and filtering
+ * [#1215] Add samplerate display in the GUI
+ * [#1663] Voicemail icon reflects voice messages
+ * [#2395] Fix account registration ( specifically with callcentric)
+ * [#2386] Strip "sip:" on incoming call, fixing history call back
+ * [#2181] Updated spec files
+ * [#1215] Display codec name in calltree instead of status bar
+ * [#2390] Move back nbCalls and stopStream higher in refuseCall
+ * [#2392] Fix ringtone during call in IAX
+ * [#2391] Stop audio streams when there is 0 calls only
+ * [#2391] Add debug when call state is not valid
+ * [#2390] Clear returns in IAXvoipLink::sendAudioFromMic() method
+ * [#2380] Fixing IncomingCallNotification not regular
+ * [#2339] Query conference at client startup
+ * [#2339] Working conference querying at startup
+ * [#2339] Add conference in call tree
+ * [#2339] Primitives to query conferences at client startup
+ * [#2320] Add account selection in history
+ * [#2355] Temporary solution: do not delete pointer when removing
+ account
+ * [#2380] Change algorithm in AudioRtp to trigger an
+ IncomingCallNotification
+ * [#2274] Comment sdebug in MainBuffer flush method
+ * [#2274] Add flushMain() in ManagerImpl::addStream
+ * [#2274] Add getBufferID() method in ring buffer
+ * [#2274] Fix warning, comment debug in ringbuffer's flush method
+ * [#2274] Use AudioLayer flushMain() and flushUrgent() in ALSA
+ * [#2274] Clean up unused variable warning
+ * [#2274] Protect minbudffer pointer on flushing
+ * [#2274] Fix playATone method which writing empty buffer in urgent
+ ringbuffer
+ * [#2274] Use audio layer flushUrgent and flushMain in createStreams
+ * [#2274] Use flush audio calls from audiolayer
+ * [#2274] Flush when peer answered call
+ * [#2375] Flush main buffer in iax when answering a call
+ * [#2274] Parse displayname using c++ string method
+ * [#2375] Flush main buffer when off holding calls
+ * [#2375] Flush main buffer mon RTP startup
+ * [#2376] Use now Pulseaudio module-cork-music-on-phone
+ * Updated OSC packaging
+
+ -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 20 Nov 2009 13:59:02 -0500
+
sflphone-client-gnome (0.9.7~beta~ppa1~SYSTEM) SYSTEM; urgency=low
** 0.9.7~beta~ppa1~SYSTEM **
diff --git a/tools/build-system/launchpad/sflphone-common/debian/changelog b/tools/build-system/launchpad/sflphone-common/debian/changelog
index 1263d2f8a2315a14efd06b7e9281b93986e4d552..7d5b2fb58422eb4d0d07c06f97e5c612bb380a7e 100644
--- a/tools/build-system/launchpad/sflphone-common/debian/changelog
+++ b/tools/build-system/launchpad/sflphone-common/debian/changelog
@@ -1,3 +1,162 @@
+sflphone-common (0.9.7~rc1~ppa1~SYSTEM) SYSTEM; urgency=low
+
+ ** 0.9.7~rc1~ppa1~SYSTEM **
+
+ * [#2462] Set explicitly the transport on incoming call too
+ * [#2462] fix typo
+ * [#2462] Use different address for SDP and call IP
+ * [#2462] Use published address in SIP-SDP
+ * [#2181] Fixed changelog files
+ * [#2181] Updated spec file
+ * [#2402] Fix pointer to int conversion warning (atoi)
+ * [#2402] Remove daemon warnings, make indent
+ * [#2459] Make sure the stream is opened when the call is answered
+ * [#2402] Add conference related picture in documentation
+ * [#2443] Not much ...
+ * [#2399] Fix dialing display problem
+ * [#2450] Fix incoming call already in conference crash
+ * [#2399] Display peer name on the first line and peer number on the
+ second
+ * [#2450] Handle 403 FORBIDDEN when refused
+ * [#2447] Bind offHold/onHold actions to button in gtk client
+ * [#2447] Bind hangup action to button for conference
+ * [#2447] Add conference action in gtk client's ToolBar
+ * [#2381] Disable the password hashing in config file
+ * [#2402] Cleanup
+ * [#2366] Set callback to null when deleting Pulseaudio streams
+ * [#1313] Fix main buffer unit test
+ * [#1313] Fix audio layer unit test
+ * [#2315] Hide pw in security tab, display when editing, sync with
+ basic tab
+ * [#1313] UnitTest change AudioRtpSession for AudioSymetricRtpSession
+ instance
+ * [#2402] Code cleanup
+ * [#2444] Add debug to catch occasional crash when loading client's
+ config
+ * [#2444] Add debug info to catch occasional crash when loading config
+ dialog
+ * [#2402] Restore Call menu translations
+ * [#2403] Use the published address if checked in GUI
+ * [#2442] Add protection test in sdp
+ * [#1841] Reapply pjsip patch concerning DNS SRV resolution
+ * [#2384] Tags incoming call as direct SIP call, if applicable
+ * [#2402] Change the monkey face
+ * [#2315] Enable user to display password in clear text
+ * [#2434] Force optimization level at 2
+ * [#2284] Fix dbus_get_all_ip_interface compilation warnings
+ * [#2431] Popup main window on incoming if applicable
+ * [$2402] Fix simple warnings
+ * [#2402] Fix implicit variable init order in LibraryManagerException
+ * [#2402] Fixing implicit variable initialization warnings in
+ AudioRtpSession
+ * [#2402] Revert atoi change, fixing codec list doubled entries
+ * [#2402] Fix gpointer to gint conversion
+ * [#2402] Fix pointer casting to integer different size warning in
+ codec list
+ * [#2402] Fix warning discarting qualifiers from pointer target
+ * [#2402] Fix gtk tree view assignement from incompatible type warning
+ * [#1669] Fix audio recording folder utf-8 non compatibility issue
+ * [#2414] Clean up debugs
+ * [#2414] Use transport set in iptoip Account and update it frm
+ preference
+ * [#2348] Use macro N_() to mark ui.xml strings as translatable
+ * [#2414] Rename getSipAddress/setSipAddress functions
+ * [#2407] Fix volume controls display
+ * [#2407] Fixes dialpad
+ * [#2383] Set ip to ip config when clicking apply button
+ * [#2404] Update call-to script - Maxime Chambreuil
+ * [#2405] Client handles unknown call in current state as well
+ * [#2383] Add DBUS signal to send IPtoIP local address and port as
+ string
+ * [#2383] Add Ip to IP config change apply call back
+ * Clonflict
+ * [#2402] Code cleanup
+ * [#2383] Do the same for IPtoIP (init localn ip with first in the
+ list)
+ * [#2383] Use first interface in the list if local addresss is not
+ defined
+ * [#2403] Clean up unuseful addresses/ports
+ * [#2403] Use the IP profile SIP port as global SIP port
+ * [#2383] Fix dbus_get_all_ip_interface warnings
+ * [#2383] Take into account sameAsLocal when loading published address
+ * [#2383] Tsake into account sameAsLocal option when saving published
+ address
+ * [#2383] Update local ip address in ip to ip config
+ * [#2383] Save ip 2 ip local port in config
+ * [#2406] Update toolbar at startup
+ * [#2284] Remove redefinition warnings + speex warnings
+ * [#2383] Fix security table in account config
+ * [#2383] Save ip 2 ip network interface parameters in config
+ * [#2403] Restore sip transport selector
+ * [#2383] Fix filling the Localt IP Address on account creation
+ * [#2383] Fix Gtk-Critical when checking STUN
+ * [#2383] Fix reopening account configuration display issue
+ * [#2383] Load IPtoIP local address and port in preference iptoiptab
+ * [#2383] Add LocalAddress and Localport in Preference IpToIp tab
+ * [#2403] Use the address and port associated to the account as often
+ as possible
+ * [#1753] Removed pjsip generated files
+ * [#1753] Removed remaining milenage lib references
+ * [#2383] Add _publishedSameasLocal variable in sipaccount
+ * [#2383] Add PUBLISHED_SAMEAS_LOCAL variable in config
+ * [#2383] Fix stun set active or not when opening config
+ * [#2181] Added RPM 64bits dbus patch
+ * [#2402] Code indentation
+ * [#2313] Force $(HOME).cache directory creation at startup
+ * [#2383] Separate network interface and published address in account
+ config
+ * [#2400] Change dbus service installation path to libdir
+ * [#2382] Move TLS related published address options in security tab
+ * [#2382] Indent accountconfigdialog.c
+ * [#2181] Install libdbus-c++ in $pkglib instead of $lib
+ * [#1753] Remove ILBC code and disable it by default in the configure
+ * [#1753] Remove milenage directory
+ * [#2382] Fix switching interaface instabilities
+ * [#2396] Save local ip in account creation wizard
+ * [#2284] Remove warning on hold
+ * [#2387] Fixes history searching and filtering
+ * [#1215] Add samplerate display in the GUI
+ * [#1663] Voicemail icon reflects voice messages
+ * [#2395] Fix account registration ( specifically with callcentric)
+ * [#2386] Strip "sip:" on incoming call, fixing history call back
+ * [#2181] Updated spec files
+ * [#1215] Display codec name in calltree instead of status bar
+ * [#2390] Move back nbCalls and stopStream higher in refuseCall
+ * [#2392] Fix ringtone during call in IAX
+ * [#2391] Stop audio streams when there is 0 calls only
+ * [#2391] Add debug when call state is not valid
+ * [#2390] Clear returns in IAXvoipLink::sendAudioFromMic() method
+ * [#2380] Fixing IncomingCallNotification not regular
+ * [#2339] Query conference at client startup
+ * [#2339] Working conference querying at startup
+ * [#2339] Add conference in call tree
+ * [#2339] Primitives to query conferences at client startup
+ * [#2320] Add account selection in history
+ * [#2355] Temporary solution: do not delete pointer when removing
+ account
+ * [#2380] Change algorithm in AudioRtp to trigger an
+ IncomingCallNotification
+ * [#2274] Comment sdebug in MainBuffer flush method
+ * [#2274] Add flushMain() in ManagerImpl::addStream
+ * [#2274] Add getBufferID() method in ring buffer
+ * [#2274] Fix warning, comment debug in ringbuffer's flush method
+ * [#2274] Use AudioLayer flushMain() and flushUrgent() in ALSA
+ * [#2274] Clean up unused variable warning
+ * [#2274] Protect minbudffer pointer on flushing
+ * [#2274] Fix playATone method which writing empty buffer in urgent
+ ringbuffer
+ * [#2274] Use audio layer flushUrgent and flushMain in createStreams
+ * [#2274] Use flush audio calls from audiolayer
+ * [#2274] Flush when peer answered call
+ * [#2375] Flush main buffer in iax when answering a call
+ * [#2274] Parse displayname using c++ string method
+ * [#2375] Flush main buffer when off holding calls
+ * [#2375] Flush main buffer mon RTP startup
+ * [#2376] Use now Pulseaudio module-cork-music-on-phone
+ * Updated OSC packaging
+
+ -- Julien Bonjean <julien.bonjean@savoirfairelinux.com> Fri, 20 Nov 2009 14:00:02 -0500
+
sflphone-common (0.9.7~beta~ppa1~SYSTEM) SYSTEM; urgency=low
** 0.9.7~beta~ppa1~SYSTEM **
diff --git a/tools/build-system/osc/sflphone-client-gnome.spec b/tools/build-system/osc/sflphone-client-gnome.spec
index 794dc7784e837ebecbf98a18f77336f8a4850701..dba4cd8ebbdf090931c142a83b961ed2299b0805 100644
--- a/tools/build-system/osc/sflphone-client-gnome.spec
+++ b/tools/build-system/osc/sflphone-client-gnome.spec
@@ -125,6 +125,7 @@ make clean
%lang(zh_HK) %{_prefix}/share/locale/zh_HK/LC_MESSAGES/*.mo
%lang(it) %{_prefix}/share/locale/it/LC_MESSAGES/*.mo
%lang(pt_BR) %{_prefix}/share/locale/pt_BR/LC_MESSAGES/*.mo
+%lang(da) %{_prefix}/share/locale/da/LC_MESSAGES/*.mo
%doc AUTHORS COPYING README
%doc %{_prefix}/share/man/man1/sflphone-client-gnome.1.gz
%doc %{_prefix}/share/man/man1/sflphone.1.gz
diff --git a/tools/build-system/osc/sflphone-common-dbus-service-in-libdir.patch b/tools/build-system/osc/sflphone-common-dbus-service-in-libdir.patch
deleted file mode 100644
index 72d23454fb73ad3a87f1b83cdf9cc809a844d50a..0000000000000000000000000000000000000000
--- a/tools/build-system/osc/sflphone-common-dbus-service-in-libdir.patch
+++ /dev/null
@@ -1,14 +0,0 @@
-diff --git sflphone-common/src/dbus/Makefile.am sflphone-common/src/dbus/Makefile.am
-index 5a2745e..5ca2e42 100644
---- sflphone-common/src/dbus/Makefile.am
-+++ sflphone-common/src/dbus/Makefile.am
-@@ -38,7 +38,7 @@ service_DATA = $(service_in_files:.service.in=.service)
-
- # Rule to make the service file with bindir expanded
- $(service_DATA): $(service_in_files) Makefile
-- sed -e "s|libexec|$(prefix)/lib/sflphone|" $<> $@
-+ sed -e "s|libexec|$(libdir)/sflphone|" $<> $@
-
- EXTRA_DIST = *.xml README
-
-
diff --git a/tools/build-system/osc/sflphone-common.spec b/tools/build-system/osc/sflphone-common.spec
index 0bc1a2b5735f438c0efef7d25ae32d5d64feee21..d44f94103e8e4e61fefc2be0a03d0e2b4a9f58c4 100644
--- a/tools/build-system/osc/sflphone-common.spec
+++ b/tools/build-system/osc/sflphone-common.spec
@@ -21,7 +21,6 @@ Packager: Julien Bonjean <julien.bonjean@savoirfairelinux.com>
BuildRoot: %{_tmppath}/%{name}
Source0: sflphone-common-%{version}.tar.gz
-Patch0: sflphone-common-dbus-service-in-libdir.patch
BuildRequires: speex-devel
BuildRequires: gcc-c++
BuildRequires: expat
@@ -95,7 +94,6 @@ Authors:
%prep
%setup -q
-%patch0 -p1
%build
cd libs/pjproject
@@ -127,7 +125,7 @@ make clean
%dir %{_libdir}/sflphone/plugins
%dir %{_prefix}/share/sflphone
%dir %{_prefix}/share/sflphone/ringtones
-%{_libdir}/libdbus-*
+%{_libdir}/sflphone/libdbus-*
%{_libdir}/sflphone/codecs/*
%{_libdir}/sflphone/plugins/*
%{_prefix}/share/dbus-1/services/org.sflphone.*