SIP2SIP's Updates

[Added sip2sip logs to darcs repo Adrian Georgescu **20131229092741 Ignore-this: 45098a742db57f1ac5d5dcf730c3b095 ] addfile ./ReleaseNotes/SIP2SIPAppcast.xml hunk ./ReleaseNotes/SIP2SIPAppcast.xml 1 + + + + SIP2SIP's Updates + https://blink.sipthor.net/SIP2SIPAppcast.xml + Most recent changes with links to updates + en + + Version 1.4.0 + 10.6.0 + + https://blink.sipthor.net/ReleaseNotes/changelog-sip2sip.html + + Thu Nov 21 11:26:17 BRST 2013 + + + + Version 1.3.3 + 10.6.0 + + https://blink.sipthor.net/ReleaseNotes/changelog-sip2sip.html + + Sun Oct 13 07:23:34 CDT 2013 + + + + Version 1.3.2 + 10.6.0 + + https://blink.sipthor.net/ReleaseNotes/changelog-sip2sip.html + + Mon Oct 7 11:42:29 CEST 2013 + + + + Version 1.3.1 + 10.6.0 + + https://blink.sipthor.net/ReleaseNotes/changelog-sip2sip.html + + Sat Sep 28 08:56:15 CEST 2013 + + + + Version 1.3.0 + 10.6.0 + + https://blink.sipthor.net/ReleaseNotes/changelog-sip2sip.html + + Fri Sep 13 08:11:32 BRT 2013 + + + + Version 1.2.1 + 10.6.0 + + https://blink.sipthor.net/ReleaseNotes/changelog-sip2sip.html + + Mon Sep 2 18:39:35 BRT 2013 + + + + Version 1.2.0 + 10.6.0 + + https://blink.sipthor.net/ReleaseNotes/changelog-sip2sip.html + + Tue Aug 27 17:02:07 CEST 2013 + + + + Version 1.1.1 + 10.6.0 + + https://blink.sipthor.net/ReleaseNotes/changelog-sip2sip.html + + Thu Aug 8 20:33:25 CEST 2013 + + + + Version 1.1.0 + 10.6.0 + + https://blink.sipthor.net/ReleaseNotes/changelog-sip2sip.html + + Mon Aug 5 14:25:36 CEST 2013 + + + + Version 1.0.2 + 10.6.0 + + https://blink.sipthor.net/ReleaseNotes/changelog-sip2sip.html + + WWed Jul 31 11:26:21 CEST 2013 + + + + Version 1.0.1 + 10.6.0 + + https://blink.sipthor.net/ReleaseNotes/changelog-sip2sip.html + + Wed Jul 24 18:41:41 CEST 2013 + + + + Version 1.0.0 + 10.6.0 + + https://blink.sipthor.net/ReleaseNotes/changelog-sip2sip.html + + Mon Jul 22 19:08:51 CEST 2013 + + + + addfile ./ReleaseNotes/changelog-sip2sip.html hunk ./ReleaseNotes/changelog-sip2sip.html 1 + + +SIP2SIP Changelog + + + + + + +

Version 1.4.0

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November 21st, 2013 + +

Added DND until call is ended +
Added copy/paste from Watcher list to Contact list +
Added Jabber and all Email addresses to AB contacts +
Added option to skip contacts sync with server when adding existing account +
Improve Chat handling of HTML payloads +
Added manual sorting for ContactURIs belonging to a Contact +
Enable audio button only if session is not in progress +
Disable presence if contact sync is disabled during enrollment +
Adjusted to latest middleware changes related to stream addition/removal +
Fixed sorting SIP addresses when droping file over contact +
Lookup icons in blink presence contacts only +
Handle multiple URIS when create presence contact from another contact +
Print job title and organization for AB contacts +
Use first SIP URI as name if missing for AB contacts +
Search AB note and job title +
Automatically process , in phone numbers as DTMF delimiter +
Fixed migrating version for session history table +
Fixed outlet alignment +
Print next hop in audio tile +
Allow deletion of contacts in OnlineGroup +
Use ICE by default +
Fixed reloading history enties after database query has executed +
Fixed restoring contacts from backup +
Don't notify subscriptions to myself +
Mark SMS messages as failed if DNS lookup fails +
Always show logs button +
Fixed debug window inconsistencies +
Fixed log line +
Fixed debug window outlets +

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Version 1.3.3

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October 13th, 2013 + +

Fixed debug window inconsistencies +
Fixed path for call transfer xib +
Fixed exception during enrollment +

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Version 1.3.2

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October 7th, 2013 + +

Added chat privacy label +
Added per session/contact chat history storage option +
Added a safety timer for music application muting +
Save use_speech_recognition in blink settings +
Improve settings descriptions +
Refactored logging to work with latest middleware changes +
Allow starting audio call from Chat window +
Honor remote preference for not logging to history database +
Fixed DNS lookup blockiness +
Fixed crash when in-dialog request fails to be sent within a subscription +
Fixed memory leak by initializing the handler after the stream initialized +
Properly handle mutex creation failures +

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Version 1.3.1

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September 27th, 2013 + +

Refactored toggling visibility of collaborative editor +
Update chat window tooltips +
End session when re-Invite fails for last stream +
Improve Chat SMP OTR window behaviour +

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Version 1.3.0

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September 13th, 2013 + +

Added SMP fingerprint verification for OTR chat sessions +
Several OPUS codec related fixes +
Use single global c line when creating SDP +
Don'r render system message when cancelling Chat sessions +
Show loading indicator in Chat view when connecting session +
Don't render OTR raw Chat messages +
Renamed audio encryption menu items for clarity +
Don't duplicate system chat message when stream fails +
Fixed starting when no account is selected +
Fixed marking failed Chat messages +
Display logging to database status for chat messages +
Fixed updating status of OTR encryption lock +

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Version 1.2.1

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September 2nd, 2013 + +

Mark pending outgoing chat messages as failed if the DNS lookup fails +
Fixed reusing audio tile after one sip route fails +
Render encryption lock for each chat message +
Use white font for failed messages +
Fixed processing related messages +
Fixed display of packet loss in audio tile +
Don't atempt using OTR with a remote focus +
Used a closed red lock when chat is encrypted but fingerprint has changed +

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Version 1.2.0

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August 27th, 2013 + +

Added Chat OTR support +
Fixed toggling visibility of all special groups +
Start ringer cleanup timer after application has started +
Improved wording of SRTP menu items +
Update SRTP in info panel +
Refactored audio encryption user interface +
Fixed detecting when music must be paused +
Make Chat drawer audio status consistent with audio tile status +
Fixed position of call transfer contextual menu +
Fixed detecting when music has been paused +
Added pause music setting +
Change chat connection messages +
Print more concise errors in chat window +
Simply LDAP queries +
Handle changes in System Address Book incrementally +
Don't set default DTMF delimiter +
Load AddressBook entries in a dedicated thread +
Fixed resetting chat controller when re-using it +
Print nicely TLS related errors +
Fixed updating audio tile info + +

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Version 1.1.1

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August 8th, 2013 + +

Reload model after deleting policy entry +
Fixed calling logging function +
Skip connecting to chat replication server if chat replication password not set +
Don't add chat to replication journal if chat replication password not set +
Handle hanging ringing sessions with a clean up timer +
Auto select authorized account for session to XMPP contacts +
Increase GUI responsivness when session ends +
Don't try publish after application ended +
Fetch last 10 entries in history menu +
Don't refresh history menu after every call +
Hide end chat delimiter when replaying Chat history +
Don't print chat session delimiters when replaying SMS history +
Disable history scolling for Bonjour accounts +
Update history menu section titles +

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Version 1.1.0

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August 5th, 2013 + +

Added elastic scrollback in time for Chat sessions +
Added elastic scrollback in time for SMS sessions +
Added search text capability within Chat and SMS sessions +
Added setting to enable the Answering Machine in alert panel +
Load history messages from all uris belonging to a contact +
Don't log duplicate database entries error +
Change background colors of failed chat messages +
Log chat errors through its associated session controller +
Fixed sending pending messages after connecting chat stream +
Send own icon only for Bonjour chat +
Improved navigation of history entries by date +
Fixed removing outgoing ringtone when using failure route +
Fixed removing on_hold ringtones for failed sessions +
Fixed initialisation of collaboration editor and display +
Fixed selecting missed calls entries +
Render voice messages recorded by answering machine in contact contextual menu +
Renamed Voicemail to Answering Machine +
Reload only the last ten file transfers by default +
Reworked mute music applications using official Scripting Bridge API +
Use night volume settings for speech synthesizer +
Don't auto open chat window at startup for replicated messages +
Fixed crash on SMS window dealloc +

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Version 1.0.2

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July 31st, 2013 + +

Don't render replicated chat messages older than 2 hours +
Reload last 10 file transfers by default +
Added scrollback in time for chat sessions +
Added scrollback in time for sms sessions +
Rename Voicemail to Answering Machine +
Render voice messages in history contact +
Render voice messages recorded by answering machine in contact contextual menu +

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Version 1.0.1

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July 24th, 2013 + +

Fixed reset of Acoustic Echo Canceller +
Added add/remove Chat to audio session tile +
Close chat stream when closing tab without ending the session +
Finish session when click on disconnect chat button +
Remove jitter buffer from session info panel +
Replace preferred media with pop up button +
Fixed printing duplicate sip code +

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Version 1.0.0

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July 22nd, 2013 + +

Client Features

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SIP2SIP account and Bonjour Discovery +
Acoustic Echo Cancellation +
Ultra-wideband audio (OPUS codec) +
Wideband audio (G.722 and Speex codecs) +
PSTN audio (G711, GSM and iLBC codecs) +
Play/pause iTunes, VLC and Spotify +
Multiparty Conferencing for all supported media +
Keychain password storage +
Voiceover accessibility support +
Address Book contacts integration +
Answering Machine and Auto-Answer +
Dialpad, DTMF, Hold, Transfer and Recording +
Many Call Automation Features +
Presence (RLS and XCAP) +
Instant Messaging (MSRP protocol) +
File Transfers (MSRP protocol) +
Contacts Management (XCAP protocol) +
Directory Services (LDAP protocol) +
Screen Sharing (VNC over MSRP) +
Interoperability with XMPP domains +
History Viewer for all media types +

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Service Features

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Multiparty Conferencing for all supported media +
Contacts sync between multiple devices +
Presence sync across multiple devices +
SMS sync across multiple devices +
MSRP chat sessions sync across multiple devices +
Access to SIP server account settings web page +
PSTN termination (paid option) +

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