Kerio Operator (сервер телефонии стандарта SIP\ISDN) - [1] :: В помощь системному администратору

western2
лови.

Добавлено:
Проведём небольшое обновление по поводу версий и их функционала

Легенда:
+ Added feature
* Improved/changed feature
- Bug fixed
! Known issue / missing feature

Version 1.2.2 — May 2nd, 2012
Kerio Operator
- 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 voicemail 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 voicemail/e-mail integration to be compatible with Kerio Connect 7.4
- Fixed a crash in the Asterisk process that could occur when stopping Operator

Version 1.2.1 — March 12th, 2012
Kerio Operator
- Fixed IMAP warning about unknown data that could sometimes occur when using the voicemail/e-mail 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 auto-provisioned 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

Kerio Operator Administration
- Updgrade by uploading the upgrade image failed if you reloaded the GUI in the browser after having finished the file upload
- The mapping of external phone numbers to local extensions could shift after inserting a new number at the beginning of the list

Version 1.2.0 — January 24th, 2012
Kerio Operator
- 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 voicemail 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 voicemail IMAP module
- Did a change in Operator's web server that should prevent Microsoft's KB2585542 update from influencing the Admin GUI when used from Internet Explorer

Kerio Operator 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)

Version 1.1.3 — November 15, 2011
Kerio Operator
- 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 1.1.2 — October 31, 2011
Kerio Operator
+ Added new time zone definition for Russia (DST setting is used even during winter)
- Voicemail messages were sent to e-mail from the address asterisk@hostname instead of the configured e-mail 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.
- Voicemail/e-mail integration is now compatible with Connect 7.3.

Kerio Operator 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 1.1.1 — August 18, 2011
Kerio Operator
+ 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 5060.
- Grandstream HT286 is now able to register with Operator.
- Removed the repeated short beep that informed about new voicemail 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.

Kerio Operator 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 1.1.0 - July 19, 2011
Kerio Operator
+ 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
+ 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
* 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.
* 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.

Kerio Operator Administration
* Improved administration interface notifications
* A new crash dump file was sometimes not reported in the administration interface. This was caused by a glitch in the implementation of the new notification system in the GUI. The error has been fixed.
* Highlighting had incorrect colors when a log was exported in HTML format.

Kerio MyPhone
* When using the voicemail/e-mail integration and a user's Full name was empty, the MyPhone interface displayed the subsequent e-mail header instead of the name.
Note: It might be necessary to manually restart some auto-provisioned phones after you upgrade from Kerio Operator 1.0.x to Kerio Operator 1.1.0. 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-synchronize the configuration automatically after a user name change.

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