Training. Kerio Operator Release History. Legend: + Added feature * Improved/changed feature - Bug fixed! Known issue / missing feature

Transcription

1 Release History Legend: + Added feature * Improved/changed feature - Bug fixed! Known issue / missing feature Version July 17th, 2013 Administration + Packet sniffer can be started on all interfaces at once. - Disable the "Remove" button when no extension is selected. - Fixed editing of IP address groups with more than 50 items. - Fixed NTP daemon restart every Sunday. - Fixed recording of calls to numbers that contain "+" or "*" signs. - Fixed ring indications for Malaysia. - Fixed the missing Date header in messages inserted by the voic / integration. Version May 30th, 2013 Administration + Improved support for IE10. + Added support for several SATA drivers and RAID controllers. + Improved busy tone detection on analog lines in several countries. - Fixed the handling of number rewriting rules for auto-provisioned Softphones. The rules might not be removed from the softphone when deleted in Operator administration. - Fixed syslog server configuration - it could be lost after reboot. - Fixed upgrade process that did not enable paging on Snom phones. - Fixed dial patterns for Snom phones - newer Snom firmwares require that asterisk (*) is escaped. Client Interface - Fixed Javascript error that could appear when removing button rows. Version patch 1 - May 2nd, 2013 Administration - Fixed a Javascript error in the external interface edit dialog. The error influenced customers with the phone language set to British English. L:\Trainings\OPT-TR-0172_ Release History\OPT-TR-0172_ Release History-EN.docx by opt17 Printed on July 18, 2013 Page 1 of 7

2 Version April 22nd, Added support for certificate chains. + Asterisk's sip.conf now explicitly contains the option "alwaysauthreject=yes". (This is the default setting but let us be sure.) + Improved detection of telephony cards when cards are swapped or removed. + SIP usernames containing a slash character are now supported by the SIP interfaces. - The NTP server is restarted once a week to prevent problems when the remote NTP server assigned to Operator by the DNS round robin goes offline during the week. - Log no errors when the computer Operator runs on has no serial port. - The SSL certificate is now parsed correctly even if it does not end with a new line character. - The distinctive ring flags for calls from a queue or group now override the external call flag. - Fixed compatibility issues with Cisco 7940 that could occur after upgrading to Operator Fixed configuration restore that might not set the correct timezone. - Fixed fallback for inbound calls that might not work if the beginning of an internal extension matched the dial-out prefix. - The web server restart from the command line could shorten the trial period. - Fixed crashes that might occur when shutting down the Asterisk process. - Fixed the detection of fax tones on PRI/BRI cards. Administration + It is now possible to enter several NTP servers. + Improved handling of telephony cards when they are swapped or removed. - Fixed update checker notifications, it might report the new version even after an upgrade in some situations. - Fixed the Javascript errors that might occur in Firefox 19 - Fixed the Javascript errors that could be reported by the "Report problem" function in special circumstances (the Administration GUI displays multiple notifications and it runs in the second browser tab) Version March 12th, Fixed the Auto attendant that could play music on hold instead of silence when waiting for user's input. Administration + Added dialog for managing Softphones. + Improved displaying of the Privacy Policy and Legal Notices (a dialog window instead of a new browser window). - Fixed status reporting when restoring a backup file on the box edition - the "server not responding" message could be incorrectly shown. - Fixed Javascript error that could occur in IE9 after logging in. - Fixed displaying of Release Notes during upgrade that could sometimes fail (occurred between RC1 and RC2) - Fixed downloading of very large packet dump files. - Fixed access to UserVoice via the "Suggest Idea" button - it could fail for timezones west of GMT-2. Version September 17th, NFR and Internal licenses could be incorrectly reported as expired. - Some notification s had incorrect date/time information. - Call parking could fail with some Snom phones if the parking position had 3 digits or more. - Fixed call queues with linear ringing strategy so that static agents always ring first. L:\Trainings\OPT-TR-0172_ Release History\OPT-TR-0172_ Release History-EN.docx by opt17 Printed on July 18, 2013 Page 2 of 7

3 - Cisco 7961 did not provision if the extension label was too long. - The ISDN/BRI interface might block if the configuration was re-generated when there was an active call. Administration + Updated the behavior of the button "Contact Technical Support" in the splash screen. MyPhone Interface - Fixed playback of voice mail messages in Safari Version August 23rd, Added auto-provisioning support for Yealink/Well T-32G and T-38G. - The option to use a caller ID different from the phone number for registration only did not work (influenced SIP connections to Megapath). - Some UTF-8 characters were stripped away from SSL certificates. - The Asterisk binary was compiled with SSE2 instructions which could cause crashes on old processors (Pentium 3) - Fixed a "double free" problem in Asterisk's voic module. - The Config log could truncate data in special situations. - Fixed a race condition where two processes could attempt to create an SQLite database at the same time, causing a crash. Administration - Corrected the Italian translation of "Auto Attendant". Version July 24th, The default action for a full call queue was changed to busy signal instead of hanging up. - Corrected call history and active calls data for calls that passed through call parking. - Optimized processing active calls data to improve performance. - A failed upgrade could sometimes stop the web server. Added restart after the failure to ensure clean return to the previous configuration in all situations. - Export of call history to CSV could in some situations contain an "unknown" call type for calls to PBX services and call queues. - Polished sqlite handling so that occasional sqlite warnings do not appear in logs. - Factory reset could fail to delete the existing configuration. - Corrected error introduced in version RC1 - voice mail messages could be left out of backup and could be restored to incorrect location when restoring a backup from a previous version in RC1. Administration - Fixed JavaScript error when editing multiple extensions at once in IE7. - The time in call history was always displayed in GMT. - Some grids still used case sensitive sorting. - The Auditor role was allowed to attempt using some actions in the administration and was refused by the engine. Corrected so that the Auditor cannot even try the actions. - Some dialogs were displayed with scroll bars in Chrome. - The Administration GUI could freeze and had to be reloaded after a long sequence of actions that included opening the context help at a particular place in the sequence. - Show notifications about blocked IP addresses with the warning icon. MyPhone Interface - Fixed call status reporting, the status could randomly disappear for a fraction of a second. L:\Trainings\OPT-TR-0172_ Release History\OPT-TR-0172_ Release History-EN.docx by opt17 Printed on July 18, 2013 Page 3 of 7

4 Version May 9th, Fixed measuring of trial period - the trial could be shortened if the algorithm happened to run at one particular time. - Corrected time zone settings for auto-provisioned Cisco 7961 phones. - Fixed downgrade from version 1.2 to 1.1 that could loose call forwarding configuration in some situations. - The voic access in MyPhone sometimes did not observe the IMAP configuration (port 143 vs. 993), and showed no voice messages as a result. - Updated the Operator/Connect handshake for voic / integration to be compatible with Kerio Connect Fixed a crash in the Asterisk process that could occur when stopping Operator Version March 12th, Fixed IMAP warning about unknown data that could sometimes occur when using the voic / integration + The SIP user ID that differs from telephone number can be now used for registration only (phone number used in calls) - Corrected downgrade from 1.2.x to 1.1.x that could fail if a new phone model unsupported in 1.1.x was autoprovisioned before the downgrade [Workaround: delete the phone entries before downgrading] - Fixed BRI error messages that could occur during boot - Fixed phone firmware upgrade that could fail for auto-provisioned SPA504G, SPA942 and SPA525G + You can now override display name for outbound calls on a SIP interface - The syslog service could freeze in a situation with an extremely high amount of data being written to the debug log - One of the web server's processes could crash when attempting to test LDAP connection with missing configuration data - Caller ID override could still display the original number in some situations - Updgrade by uploading the upgrade image failed if you reloaded the GUI in the browser after having finished the file upload Administration - The mapping of external phone numbers to local extensions could shift after inserting a new number at the beginning of the list Version January 24th, Corrected BRI module reload procedure that could log warnings - Incorrect called number was reported in Status->Calls for calls that went through a Ring group - Fixed a memory leak and/or crash in the phone provisioning TFTP process that could sometimes occur if the TFTP traffic was filtered in one direction by a firewall - Removing voic integration with Kerio Connect could sometimes fail - SIP registration and SIP proxy setup could sometimes work with different IP addresses for the same SIP server when the SIP carrier uses a DNS round robin setup - Fixed several resource leaks and potential deadlocks in Asterisk's voic IMAP module - Did a change in Operator's web server that should prevent Microsoft's KB update from influencing the Admin GUI when used from Internet Explorer Administration - The field that holds external numbers has been extended to allow up to 100 individual phone numbers on a SIP interface - Corrected translations for languages in the Administration GUI - Fixed Javascript error when accessing the provisioned phones screen as a read-only administrator (Auditor) L:\Trainings\OPT-TR-0172_ Release History\OPT-TR-0172_ Release History-EN.docx by opt17 Printed on July 18, 2013 Page 4 of 7

5 Version November 15, Fixed a bug in Auto attendant where an external caller could hear the default music on hold instead of silence while deciding which number to press. - Transferring an external call to another external number could fail on an interface with an empty dial-out prefix. - When integrated with Active Directory, user's phone number in AD could be overwritten with voice mail access number when deleting the user's last extension in Operator. Version October 31, Added new time zone definition for Russia (DST setting is used even during winter) - Voic messages were sent to from the address asterisk@hostname instead of the configured address. - If there was silence on the line (for example when waiting for user input in auto attendant), Operator stopped sending RTP packets. However some SIP providers stop the call if the RTP is not flowing for 20 seconds. To solve this, Operator is now sending RTP packets with silence in this situation. - Voicemai/ integration is now compatible with Connect 7.3. Administration - Fixed a problem in the Administration GUI when editing the PRI/BRI interface that could result in JavaScript errors, displaying the "Web-crash" dialog. Version August 18, Added possibility to change the User-Agent string sent by Operator in the SIP protocol. - The maximum allowed registration interval is now 1 hour (changed because of the SIP provider freephonie.net). + Added support for German SIP provider QSC (the field "To:" is used instead of the number in the INVITE request line). - SIP provider configuration is now correctly generated if the port number differs from Grandstream HT286 is now able to register with Operator. - Removed the repeated short beep that informed about new voic messages on auto-provisioned Polycom phones. - Dial patterns generated for auto-provisioned Polycom phones were incorrect when there was not dial-out prefix in the dial plan. - Fixed directory server errors when attempting to connect to it too soon after booting. - Fixed a race condition in the TFTP server that could lead to a crash. - A call loop created by incorrect fallback configuration could cause 100% CPU utilization. The new implementation prevents loops from overloading the CPU. Administration - Corrected several small translation glitches in Czech, German, and Russian. - The administration GUI did not warn when trying to activate a user from the directory server who collided with an existing local user. Version July 19, Protection against SIP password guessing + Stopping the PBX if an anomalous behavior is detected + Multiple SIP registrations of the same extension + Various PBX voice services (dial-by-extension, dial-by-name, echo,...) * Improved NAT support + Call queue improvements L:\Trainings\OPT-TR-0172_ Release History\OPT-TR-0172_ Release History-EN.docx by opt17 Printed on July 18, 2013 Page 5 of 7

6 + Uploading of custom voice prompt sets * Full implementation of ilbc and G.722 codecs (including transcoding to/from other codecs) + Auto-provisioning support for Polycom phones * Improved Admin GUI notifications * Highlighting had incorrect colors when a log was exported in HTML format. * Operator generates Asterisk configuration after each restart to make sure Asterisk's files are always consistent with the configuration database. The files were sometimes not re-generated when Operator was paired with an Active Directory server. Even though the probability of configuration file corruption is low, the error was fixed and the files are now always re-created after each restart. * A new crash-dump file was sometimes not reported in the Admin GUI. This was caused by a glitch in the implementation of the new notification system in the Admin GUI. The error was fixed. * When using the voic / integration and a user's Full name was empty, the MyPhone interface displayed the subsequent header instead of the name. * DTMF codes were sometimes not sent when calling out using a BRI (EuroISDN) card. * Incoming calls were always sent to the fallback number on a SIP interface that had several external numbers and at the same time its User ID differed from the telephone numbers. Note: It might be necessary to manually restart some auto-provisioned phones after you upgrade from Kerio Operator 1.0.x to The user names provided to the phones in the configuration files have been changed in connection with the support for multiple registrations of the same extension. Some phones (e.g. some Cisco models) are not able to re-sync the configuration automatically after a user name change. Version April 13, The "KSslSocket::accept failed" messages moved from security log to debug log (they appear if the browser displays a dialog to accept the SSL certificate). - Non-configured BRI port produced many unnecessary messages in the warning log. + Improved the algorithm that compiles dial patterns for auto-provisioned phones, resulting in fewer phone restarts when changing the dial plan. - Moving an extension to a user from a directory server was not possible. - Only the first extension could call out on some auto-provisioned Snom models with multiple assigned extensions. - Setting up external IP address for a SIP interface was not working correctly. - Errors when setting up integration with Kerio Connect were not reported correctly. + The administration GUI does not allow empty SIP passwords. - The upper limit for recorded calls' storage was only 8192 MB. + Improved warning in the administration GUI when license expires. + Added 0.5 second or 1 second pauses in some situations (like voic greeting) so that users dont miss the first word of the message. - Snom "Voic " button dialed the string "asterisk" instead of the actual voic extension. - No dial tone was heard when sending a call from auto attendant script to another extension. - Music on hold was not random. + Low-level line parameters are now accessible in the PRI card configuration. Version March 2, Recorded calls can be downloaded - Voic number was not associated with the voic access key on some auto-provisioned phones - Custom voic greetings and recorded user name were ignored - Snom models 320, 370, 821, and 870 were not correctly provisioned when an extension was assigned manually - address could not be used as user name in SMTP authentication - External interfaces always had English as the default language for system messages even if the system default was another language - Backup sometimes returned empty file L:\Trainings\OPT-TR-0172_ Release History\OPT-TR-0172_ Release History-EN.docx by opt17 Printed on July 18, 2013 Page 6 of 7

7 - Codecs were sometimes reset to the default order - Some Czech ordinals were not correctly inflected by the voic interface Training - ilbc and G.722 were not marked as "pass-through only" codecs - DST parameters were not correct in auto-provisioned configuration for Cisco The field for external phone numbers in SIP interface was limited to 67 characters - Session expired when connecting to Control and Operator administration consoles through the same host at the same time - Incoming calls went directly to fallback number with some SIP providers - Incorrect CallerID was used for outgoing calls with some SIP providers Have a Great Day! L:\Trainings\OPT-TR-0172_ Release History\OPT-TR-0172_ Release History-EN.docx by opt17 Printed on July 18, 2013 Page 7 of 7