Version History
3.16.4.2 (April 10, 2024) – Update Release
For PBX
– Added PAL WebSocket/REST method for deleting prompts (deletePrompt)
– Fixed the bug where simultaneous ringing did not function correctly
– Updated the embedded Web Phone to version 2.13.3.
– Added additional access control options for System Administrators
– Improved the interop issue with outbound calls from Brekeke Phone on iOS
– Fixed minor bugs
For SIP Server
– Improved Block List handling
– Improved certificate validation for TLS and STIR/SHAKEN.
– Improved STIR/SHAKEN’s key management.
– Fixed the bug where multi-byte characters were not output correctly in Push Notification for FCM.
– Fixed the bug where an exception happened under the bind setup if REGISTER’s Contact URI is *.
– Fixed minor bugs
3.16.3.1 (March 10, 2024) – Standard Release
For PBX
– Fixed the bug where Auto Sync did not work properly in version 3.16.1.5
– Improved where Audio Player on Safari did not work properly
– Fixed minor bugs
For SIP Server
– Fixed the bug where Terminating character for user-info did not work properly
3.16.1.5 (Feburary 9, 2024) – Beta Release
For PBX
– Added PAL WebSocket/REST methods (email, userdir, alias, registered, searchExtensions)
– Enhanced email sending capabilities through PAL WebSocket/REST API for HTML emails
– Improved push notification for Brekeke Phone
– Improved the PAL WebSocket connection method for better reliability
– Enhanced the Auto Sync feature (now supports re-sync from the admin tool).
– Added phone_idx to the notify_status evevnt of PAL WebSocket
– Added API for MFA for UC/CCS
– Added c_recinfo column for the log database
– Fixed the bug where the SQL Query module did not work
– Fixed the bug where encryption module for the provisionig feature did not work on Linux
– Added post call feature for CCS
– Updated Apache Tomcat version to v9.0.83 for Windows
– Fixed minor bugs
For SIP Server
– Improved WebPush interoperability
– Improved STIR/SHAKEN interoperability
– Improved IPv6 interoperability
– Improved certificate file management.
– Improved Reverse Proxy handling for WebSocket
– Improved SIP Registrar performance
– Improved the Mirroring function
– Fixed the bug where an error occurred if the “tag” value was not matched under Redundancy and Failover deployment
– Fixed the bug where a response packet sent from subsequent failover destination might be unaccepted under Redundancy and Failover deployment
– Fixed the bug where an error occurred if the “tag” value was not matched under Redundancy and Failover deployment
– Fixed the bug where the monitoring timeout in Mirroring was increased even if there was no delayed monitoring response
– Fixed the bug where FCM in Push Notification didn’t handle multibyte characters correctly
– Fixed minor bugs
3.14.7.4 (June 22, 2023) – Update Release
For PBX
– Support REST API for userdir
– Support am API key for PAL REST API
– Increase the buffer size to 64k for the PAL WebSocket
– Fixed the bug where a session was disconnected even though when it failed to put on call park (Panasonic, Nakayo)
– Fixed some bugs for the line key features for the Nakayo phone
– Fixed the encoding bug where communicating with CIM
– Fixed the bug where PBX freezes on infrequent occasions when very high traffic on the connection with CIM
– Added the getNoteNames to PAL WebSocket
– Updated Apache Tomcat version to v9.0.71 for Windows
– Fixed minor bugs
For SIP Server
– Fixed minor bugs
3.14.5.18 (Feburary 23, 2023) – Update Release
For PBX
– Fixed the bug where Brekeke Phone on iOS could not receive Push Notifications
– Fixed minor bugs
3.14.5.17 (Feburary 16, 2023) – Update Release
For PBX
– Improve the line key feature using PAL methods
– Fixed the bug where the Phonebook display name cache was not cleared properly
– Improve the Avatar feature
– Adding a feature to sync call parameters with CIM
– Optimize Push Notification for Brekeke Phone
– Fixed the memory leak with PAL WebSocket when specifying the user/park/queue
– Fixed minor bugs
For SIP Server
– Improved internal timer management
– Adding a field for the listening port to the Apple LPC settings
– Fixed the bug where the Application List does not show LPC properly.
3.14.4.12 (December 27, 2022) – Update Release
For PBX
– Support sending ping-pong on PAL WebSocket connections
– Fixed the bug where RTP packets for RFC2833 could not be routed when SDP has multiple lines for telephone-event
– Added an interface for Brekeke Phone v2.11.x LPC
– Fixed the bug where Auto Sync does not work with some PAL WebSocket methods(copyRouteTemplate, createExtension)
– Fixed the bug where Auto Sync does not work when renaming an ARS template
– Fixed the bug where, on rare occasions, a NOTIFY request is not sent for Event talking
– Enhanced music on hold function to be able to specify a music file using a script
– Added multiple jump fields to the Send/Recv module template for UC Chatbot in Flow Designer
– Fixed minor bugs
For SIP Server
– Added STI-CA List management
– Improved Mirroring stability
– Fixed the bug where a Mirroring’s virtual interface couldn’t be paired with a physical service interface under the binding deployment.
– Fixed the bug where some failure in blocking malicious activities.
– Fixed minor bugs
3.14.3.5 (October 4, 2022) – Update Release
For PBX
– Fixed the bug where a call disconnected after recording or detecting DTMF on infrequent occasions
– Update log4j to v2.18.0
– Fixed the bug where importing module templates did not work properly when the same group already existed
– Fixed minor bugs
For SIP Server
– Fixed the bug where the SIP server failed to send 100 OK on infrequent occasions
– Improved STIR/SHAKEN connectivity
– Fixed bug where the SIP server failed to process an initial INVITE request with a Route header
– Fixed minor bugs
3.14.2.5 (September 6, 2022) – Update Release
For PBX
– Fixed the bug where stereo audio files of an email attachment were corrupted
– For auto answer function, added x_autoanswer: true on a Push Notification payload
– Enhanced music on hold function where multiple music files can be rotated
– Improved early media handling when whisper/confirm call function is used.
– Added x_time on Push Notification payload for Brekeke Phone
– Added deleteContact(for Brekeke Phone), getFlowNames(for UC) methods to Brekeke PAL WebSocket
– Fixed the bug where Jump of the previous module is deleted when a module is deleted in Flow Designer
– Updated the design of Flow Designer
– Improved the processing of PRACK
– Added service IP for redundancy to the BindToDevice mapping
– Updated the version of Web Phone to v2.10.9
– Added encrypted plug-in for SAXA brand’s IP telephone handsets
– Fixed minor bugs
3.14.1.2 (August 1, 2022) – Standard Release
For PBX
– Fixed the bug where the incorrect agent status was shown when logging into CIM using BLF
– Fixed the bug where SIP-INFO (dtmf-relay) did not work for the Call Hunting queue
– Improved Flow Designer
– Fixed the bug where module properties could not be set from the extension settings
– Added default language settings
– Improved shutdown process
– Added the Ring Timeout settings to the Tenant options page
– Added the Carecom integration
– Added Domain field in the PBX > Options page
– Optimized product performance
– Update Web Phone to v2.9.13
– Fixed the bug where Remote Support did not work under some network conditions
– Improved handling of the recording file for the Zoho integration
– Added module templates for UC chatbot to Flow Designer
– Fixed the bug where a branch ID will conflict under certain conditions
– Fixed minor bugs
For SIP Server
– Mirroring: Send an E-Mail Alert from the event when the role is changed from standalone
– Mirroring: Keep the designated role if both servers are in standalone role
– Added Apple Local Push Connectivity support
– Improved the interface binding at the RTP relay
– Added reverse proxy settings
– Fixed minor bugs
3.12.4.1 (March 25, 2022) – Update Release
– Fixed the bug where existing IVR flows cannot be loaded when the PBX version is updated from v3.10 or older
– Added minor improvements
3.12.3.1 (March 7, 2022) – Update Release
For PBX
– Improved the CPS adjustment allowing as short as 0.1 millisecond
– Added/Improved Brekeke PAL methods (getNote/setNote/email)
– Updated Apache Tomcat version to v9.0.58 for Windows
– Change the default behavior to add the userinfo part of the SIP URI in the Contact header for incoming sessions
– Support new functions available in the Brekeke Phone v2.9.x
– Fixed the bug where flow settings could not be saved in the Flow Designer
– Added the function to play a prompt file after answering an incmoing session with sound.answer property
– Fixed minor bugs
For SIP server
– Fixed the bug where the RTP relay did not work correctly with IPv6
– Fixed the bug where IPv6 handling caused a parse error under certain conditions
– Fixed the bug where Push Notification caused a database error under certain conditions
– Fixed minor bugs
3.12.2.2 (January 11, 2022) – Update Release
For PBX
– Improved the getContactList method of PAL WebSocket interface
– Improved Push Notification function for Brekeke Phone
– Improved display name encoding for 3pcc/PAL WebSocket
– Improved the file lock method while reading/writing of a Note
– Fixed minor bugs
For SIP server
– Fixed the bug where a virtual interface with non-5060 did not accept a SIP response when there is no default listening port binding on the same virtual interface
– Fixed the bug where the SIP server did not check a packet’s eligibility during the startup when the binding is enabled
– Fixed the bug where the virtual interface IP address could be added in the BlockList
– Fixed the bug where interface pattern order was not prioritized correctly
– Fixed minor bugs
3.12.1.5 (December 10, 2021) – Standard Release
For PBX
– Added the Console page
– Added the File Manager page
– Added the Remote Support page
– Improved the Flow Designer (called IVR Designer in previous versions)
– Added tenant administrators access to the Flow Designer
– Added the Flow Settings page in Multi-tenant (Flow Selection, Extension Action) Edition
– Improved push notification functions
– Returns Q.850 code (205) when a timeout occurs at CPA
– Added any system administrator access (previously only allowed for sa) to Brekeke PAL WebSocket settings
– Fixed the bug where hold status was not released when Brekeke PBX received INVITE w/o SDP in a mid-session
– Support CPS adjustment setting
– Added capability to use SCA under multiple UA for the same PBX user
– Added a parameter to specify the SIP UA index number at 3PCC/Brekeke PAL WebSocket
– Improved an error handling when a user has been deleted
– Added system administrator’s access rights (required ADV setting)
– This version is compatible with Web Phone (v2.7.5)
– This version is compatible with CCS v2.7.8
For SIP Server
– Added improved interface and transport binding
– Added STIR/SHAKEN functions, including STI-AS and STI-VS
– Added Alias database cache
– Added support for FCM HTTPv1
– Added support for EC (Elliptic Curve) key file
– Added the feature to work behind an HTTP reverse proxy such as NGINX for WebSocket
– Added the feature to send a client certificate if it is requested over TLS/WSS handshake
– Improved transport connection connectivity
– Improved Push Notification interoperability
– Improved NAT Traversal
– Improved DNS handling
– Improved Mirroring stability
– Added the Local IP Address setting at the Mirroring menu
– Improved WebSocket connectivity by expanding the buffer size and managing fragmented frames
– Added the sending of PING over WebSocket
– Improved Failover handling
– Improved the digital certificate management
– Improved TAP connection persistence
– Improved relay function at ICE packets
– Fixed the bug where SIP proxy refused to accept received SIP packet under certain conditions if [Keep address/port mapping] is used.
– Fixed the bug where PushNotification didn’t work correctly with Multiple-Domain mode.
– Fixed the bug where the CDR function might freeze under certain conditions when the Failover plugin is used
– Fixed the bug where $ifdst was not applied in the Upper/Thru register
– Changed the export limit of the number of concurrent secure transport connections from 1000 to 2500
– Fixed minor bugs
3.10.6.5 (June 16, 2021) – Update Release
– Fixed the bug where the 3rd party database is terminated on some occasions.
3.10.6.5 (June 16, 2021) – Update Release
For SIP Server
– Fixed the bug where the 3rd party database is terminated on some occasions.
3.10.6.4 (May 28, 2021) – Update Release
For PBX
– Fixed a bug where PBX could not keep refreshing an auth token for Zoho integration
– Added an option to restrict using 3pcc only from the same IP address that is used for a SIP UA.
– Added a parameter to specify the phone index number for 3pcc / PAL WebSocket API (makecall method)
– Fixed a bug in RecordingFileHttpUploader plug-in where it failed to continue uploading files when an error occurred
– Added PAL WebSocket methods to copy/rename ARS route template (copyRouteTemplate/renameRouteTemplate)
– Added REST API interface for PAL methods
For SIP Server
– Allowing larger header size for ws/wss transport
– Fixed a bug where it reproduces deadlock under rare circumstances
3.10.5.6 (March 25, 2021) – Update Release
– Updated Apache Tomcat version to 9.0.44 for Windows
– Updated Web Phone version to 2.7.5
– Fixed the bug where RTP packets for telephone-events(RFC2833) are not correctly routed on some occasions
– Fixed minor bugs
3.10.5.4 (March 4, 2021) – Update Release
For PBX
– Fixed the bug where unhold the call when receiving a re-INVITE request without SDP
– Fixed the bug where one-way audio occurs when a queued call in call hunting is answered using pick up call function.
– Fixed the bug that occurs during version update from v3.5.2.x or older to a newer version when Java 11 or later is used.
– Fixed minor bugs
For SIP Server
– Fixed minor bugs
3.10.4.3 (December 11, 2020) – Update Release
For PBX
– Fixed the bug where the [Call Recording patterns] in the user settings did not work properly
– Improved the installer and now it does not require MSVCR90.DLL
For SIP Server
– Fixed the bug where the SIP-REGISTER handling in Standard Edition does not release thread resources under a certain conditions when it is integrated with RGS (Ametek’s Responder Gateway Server)
3.10.4.1 (November 13, 2020) – Update Release
For PBX
– Enhanced BLF functions allowing display and pick up calls located in the call queue
– Enhanced BLF functions allowing agents to log in to Brekeke CIM using BLF (a button of the phone)
– In Brekeke PAL WebSocket, call parameter information (ex1, ex2) will be included in the notify_status event
– Fixed the bug where Brekeke PBX crashes when a certain type of packets is received during ICE process occur
– Unused device information for Push Notification will be removed from the list.
– Fixed minor bugs
For SIP Server
– Fixed the bug where exporting Alias DB fails when third party database is used
3.10.3.0 (August 21, 2020) – Update Release
– Improved WebRTC protocol (DTLS-SRTP) connectivity with Chrome M85 or later
3.10.2.8 (August 7, 2020) – Standard Release
For PBX
– Added an option to recording in stereo (PBX user is recorded on R and the other party is recording on L)
– Added an option to send emails for Call recording files to another address
– Added an option to send call recording files to an email address different from an email notification for voicemail.
– Fixed the bug where timestamp of the outgoing RTP packets is not correctly set on some occasions
– Fixed the bug where RTP packets for telephone-events(RFC2833) are not correctly routed on some occasions
– Fixed the bug where there was a 3-second delay when keypad command is used
– Added the Topic field for APNS (Push Notifications)
– Added some additional information in the Call Log files
– Added the “Background Script” module template to the IVR designer
– Added the following IVR functions – getQueuePosition(), stop(), stopPlaying(), stopRecording()
– Fixed the bug where Zoho integration did not work properly when a call is manually made from a phone (not using Zoho’s screen).
– Fixed the bug where RINGING event is sent after ENDTALKING with PAL WebSocket.
– Added syntax check when the regular impression is used at the setting
– Added an option to replace the logo/favicon for each tenant
– Added search methods in contains ARS plug-in
– Added function to send an alert email when stored voicemail messages reaching full capacity
– Fixed minor bugs
3.9.6.0 (May 19, 2020) – Update Release
For PBX
– Fixed a bug where BLF/SCA does not work properly (issue existed v3.9.5.8 only)
– Fixed minor bugs
3.9.5.8 (April 1, 2020) – Update Release
For PBX
– Improved call tutoring feature allowing multiple tutors to be able to communicate with an agent
– Added the http plug-in (ARS plug-in)
– Updated Web Phone version to 2.1.2
– Updated Apache Tomcat version to 9.0.33 for Windows
– Support AdoptOpenJDK (Windows installer)
– Fixed minor bugs
For SIP Server
– Fixed the bug where $replaceuri.from and $replaceuri.to don’t work
– Fixed the bug where AutoSync for the user database does not work on some occasions (Removed the UNIQUE attribute from the idx_userdir_uid index for the t_userdir table)
– Fixed minor bugs
3.9.5.5 (March 6, 2020) – Update Release
For PBX
– Made improvement of the handling of sending emails
– Fixed the bug where call transfer fails in a certain occasion
– Fixed the bug where music on hold did not work properly after using 3PCC call
– Fixed the bug where voice quality issues occur when there is a high disk usage
– Added the option to specify minimum file size when RecordingFileHTTPUploader plugin is used
– Fixed minor busg
For SIP Server
– Added “apns-push-type” header support in the APNS handling
– Fixed the bug where the Mirroring function didn’t remove the Service IP address during the shutdown
– Fixed the bug where the $wait4reg method didn’t work under certain conditions
– Fixed minor bugs
3.9.4.9 (February 3, 2020) – Update Release
For PBX
– Improved timeout handling of the RecordingFileHttpUploader class (Audio File Plug-in)
– Support continuous monitoring by prefix 91*(supervise mode) and 92*(tutor mode)
– Support continuous monitoring by adding a parameter continuous:true for the barge function of Brekeke PAL webSocket
– Added a scheduling feature for selecting voicemail greetings to play
– Zoho CRM integration
– Modified to unhold the call when receiving a re-INVITE request without SDP
– Added support of DTLS 1.2
– Upgrade user interface of Web Phone allowing better usability
– Fixed minor bugs
For SIP Server
– Fixed the bug where the FCM plugin didn’t point to the latest endpoint URL
– Fixed the bug where the Failover plugin didn’t handle DNS-SRV failover with non-UDP transport
– Fixed the bug where DNS-SRV Load Balancing counter was increased unexpectedly
– Fixed the bug where SIP proxy returned “603 Decline” when non-UDP transport couldn’t be made under certain conditions
– Fixed the bug where WebSocket server returned HTTP “400 Bad Request” under certain conditions
– Fixed the bug where the Mirroring function didn’t use the same “realm” value for Authentication in the default
– Added the new shutdown option in Mirroring function which doesn’t trigger the switching of the role
– Added the option to accept free-formatted text for TAP
– Improved license framework.
– Fixed minor bugs
3.9.4.3 (August 13, 2019) – Update Release
For SIP Server
– Fixed the bug where Brekeke SIP Server (Advanced Edition) may reject INVITE with “500 Server Internal Error” under certain condition
– Fixed the bug where, under a rare case, duplicated records are made in the registered database
– Fixed minor bugs
3.9.4.1 (July 15, 2019) – Update Release
For PBX
– Added copy function for the module template in IVR designer
For SIP Server
– Fixed the bug where embedded database stops working when Java version 9 or later version is used
– Fixed the bug where unnecessary records remained in registered database
– Fixed the bug where Registrar for Advanced Edition will consume resources when there is no response from the database.
– Added new variable “net.rtp.openflow.matching.in_port” for adding “in_port” matching conditions in the OpenFlow function. The default value is false.
– Fixed minor bugs
3.9.2.7 (April 26, 2019) – Update Release
For PBX
– Text to Speech (Google Cloud Text to Speech, IBM Watson Text to Speech) added to IVR
– Fixed minor bugs
For SIP Server
– Fixed the bug where the database connection did not recover after disconnection when 3rd party database was used
– Fixed the bug, on a rare occasion, where some configurations on SIP server are not applied
– Fixed the bug where SIP server did not handle certain SDP content
– Improved OpenFlow monitoring
– Fixed minor bugs
3.9.2.4 (April 12, 2019) – Update Release
For PBX
– Fixed the bug where the HTTP module template did not work
– Fixed minor bugs
For SIP Server
– Improved the SDN and Mirroring integration
– Fixed minor bugs
3.9.2.1 (March 28, 2019) – Update Release
For PBX
– Call Log is added to Auto Sync
– Shows outbound sessions which have not connected in the Call Logs
– Fixed minor bugs
For SIP Server
– Improved interface address handling of mirroring function
– Fixed the bug where API Key/Server Key could not be stored in the Edit Application (GCM/FCM) page
– Fixed minor bugs
3.9.1.8 (March 20, 2019) – Update Release
For PBX
– Web Phone has been updated
– Support STARTTLS to send emails
– Changed the mirroring request options
– Updated Apache Tomcat version to 9.0.16 for Windows
– Fixed minor bugs
For SIP Server
– Improved the interface address handling at Mirroring function.
– Fixed the bug where the DNS resolver might block the starting of the SIP server.
– Fixed minor bugs
3.9.1.4 (December 27, 2018) – Update Release
For PBX
– Fixed the bug where a play list cannot correctly be played in IVR script/flow.
3.9.1.3 (December 21, 2018) – Standard Release
For PBX
– Improved redundancy feature
– Improve WebRTC connectiviry
– Fixed minor bugs
For SIP Server
– Improved PRACK handling
– Fixed the bug where ACK was sent after BYE under certain conditions
– Added a script action for Heartbeat
– Fixed minor bugs
3.9.1.0 (December 4, 2018) – Beta Release
For PBX
– Improved redundancy function by updating AutoSync function to achieve active-active redundancy
– Fixed the bug where RTCP was failed to send to the caller using ARS IN pattern
– Added “session.response.timeout” support at the Phone Type properties settings.
– Fixed the bug where DTMF was not detected using RFC2833 when the payload type is 98
– Added supported character (i.e, < >) in IVR playlist
– Fixed the bug where the replacement string in ARS can be null
– Fixed the bug where an issue occurs when RFC2833 and SIP-INFO are used at the same time
– Added a method “recording” and “separate” at the interface for Brekeke CCS
– Fixed minor bugs
For SIP Server
– Improved mirroring function to minimize downtime and increase the efficiency of the system administrative tasks
– Improved transport connectivity
– Fixed the bug where FCM/GCM handling didn’t generate JSON data correctly under certain conditions.
– Fixed minor bugs
3.8.6.4 (August 30, 2018) – Update Release
For PBX
– Added a secure transfer feature (for Brekeke CCS)
– Modified IVR extension allowing double-bite character set in the property field
– Fixed the bug where script couldn’t be directly written at the post call script in IVR Flow
– Added “HTTP”, “Set Callback in Queue”(for Brekeke CCS) as module templates in IVR Flow
– Fixed the bug where replacing a display name by the phonebook did not work when the port number was changed for the embedded database
– Fixed the bug where, in some rare cases, the connection from Brekeke CIM was disconnected.
– Fixed the bug where, in some rare cases, some information to cdr (Brekeke SIP Server) were missing such as pbx_tenant.
For SIP Server
– Fixed the bug where the secondary server of Mirroring reported internal errors
– Fixed the bug where Mirroring Request Pattern didn’t work.
– Fixed the bug where the SIP proxy kept consuming memory under certain conditions.
– Improved Apple Push Notification Service (APNS) interoperability
– Improved transport connection connectivity
– Improved RADIUS failover
– Fixed minor bugs
3.8.5.2 (May 31, 2018) – Update Release
For PBX
– Added “Call Park” page in product admintool to set up timeout/callback
– Fix the bug where a call was disconnected when forwarding destination has% prefix
– Fix the bug where exported user logs (CSV file) enter incorrect time
– Improve Brekeke PAL WebSocket to work with a newer version of Tomcat
– Fixed the bug where the connection with Brekeke CIM gets disconnected on very rare occasions.
– Added Java 10 support
– Fixed minor bugs
For SIP Server
– Added Apple Push Notification Service (APNS) VoIPPush and Token-based provider trust support
– Fixed minor bugs
3.8.4.9 (April 20, 2018) – Update Release
For PBX
– Fixed the bug where the auto-provisioning device page did not work properly.
– Fixed minor bugs
3.8.4.4 (April 18, 2018) – Update Release
For PBX
– Fixed the bug where failed to detect DTMF signals with RFC2833 format.
– Fixed minor bugs
3.8.4.2 (April 6, 2018) – Update Release
For PBX
– Fixed the bug where failed to save IVR flow when the module is placed at acertain location
– Fixed the bug where a processed error occurred when PBX receives 10ms packetwhile G.729a codec is used
– Fixed minor bugs
3.8.3.4 (February 2, 2018) – Update Release
For PBX
– Fix the bug where deadlock occurs between phonebook and database connection
– Fix the bug where ARS session counter does not reset when a value larger than 0 set at Session interval under ARS settings
– Fix the bug where jump destination of IVR Designer’s module is not correctly saved
3.8.3.0 (January 16, 2018) – Update Release
For PBX
– Fixed the display errors at the edit screen of script module in IVR designer setting
– Modified IVR flow settings to allow writing scripts in postscript field
– Fixed the bug where linking issue with stand-alone video client
– Fixed the bug where multibyte character gets garbled when Tomcat 8 has been used
– Fixed minor bugs
For SIP Server
– Use TIMESTAMP/DATATIME as the default data type for CDR time stamps.
– Fixed minor bugs
3.8.2.4 (December 29, 2017) – Standard Release
For PBX
– Added automatic upload function where an error occurs at RecordingFileHttpUploader
– Added an option at Phone Type settings to choose phonebook for the display name of incoming calls
– Fixed minor bugs
For SIP Server
– Added CREATE TABLE statement editor in CDR
– Added SQL error log in CDR
– Fixed minor bugs
3.8.1.1 (December 1, 2017) – Beta Release
For PBX
– Cleaned and updated default Dial Plan of the bundled SIP server
– Added timer function to stop provisioning process
– Fixed a bug where filter settings were not applied to CSV download in the
– Call Logs page
– Added import/export function to user settings
– Added import/export function to ARS route settings
– Added copy function to extension
– Added/improved Brekeke PAL WebSocket methods for Push Notifications and Phone book
– Fixed minor bugs
For SIP Server
– Added Phone book function
– Added new CDR function
– Improved DNS performance
– Improved Push Notification interoperability
– Fixed minor bugs
3.7.7.8 (October 4, 2017) – Update Release
For PBX
– Fixed the bug where, in a rare occasion, uploading of audio file from RecordingFileHttpUploader stop working
– Changed where any type of extensions will be evaluated by the User fields of ARS OUT-Pattern
– Fixed minor bugs.
For SIP Server
– Fixed the bug where SRTP/RTP conversion did not start correctly when subsequent INVITE is received during the closing phase after the initial INVITE was rejected
– Fixed the bug where the routing problem occurs when a first failover destination returns 18x response with Record-Route header, and the next destination is located on localhost
– Fixed minor bugs.
3.7.7.3 (August 3, 2017) – Update Release
For PBX
– Improved a security issue.
– Added a function to be able to copy “Phone type”
– Fixed minor bugs
For SIP Server
– Improved RTP relay when ICE is used
– Added live logging update in the [Diagnostics]>[Debug Logs] page.
– Added HTTPS support in $webget()
3.7.6.6 (May 26, 2017) – Update Release
For PBX
– Added Brekeke PAL WebSocket method for Web Push (pnmanage)
– Modified access permission level to Brekeke PAL WebSocket
– Improved WebPhone function
– Fixed minor bugs
For SIP Server
– Fixed the bug where TLS and WSS packet mirroring for redundancy caused a request timeout when the secondary server is not running.
– Added Multiple Dispatcher mode to execute DialPlan rules concurrently.
– Added support for Web Push for Chrome browser.
– Fixed the bug where variables to bind IP address for RTP relay did not work properly.
– Fixed minor bugs
3.7.5.0 (April 21, 2017) – Update Release
For PBX
– Fixed minor bugs
For SIP server
– Fixed the performance issue of RTP relay
– Fixed minor bugs
3.7.4.8 (April 4, 2017) – Standard Release
For PBX
– Added a setting to manage blocked IP address that has been added multiple failed try to log in
– Modified access permission level to Brekeke PAL WebSocket
– Improved WebPhone feature
– Fixed the bug where timeout setting is not working properly while uploading files when using “com.brekeke.pbx.vm.VmIVRService.RecordingFileHttpUploader” plug-in.
– Fixed minor bugs
For SIP server
– Added heartbeat option to continue after a failure
– Fixed minor bugs
3.7.4.3 (March 21, 2017) – Beta Release
For PBX
– Improved 3pcc/call back function
– CPA integration for Aculab ProsodyS
– Removed codec G.729 option (allowing unlimited G.729 codec sessions)
– Added to support for binding RTP port with ARS (&net.rtp.bind)
– Added some IVR module templates for Brekeke CIM
– Added some functions to Brekeke PAL WebSocket
– Added support for Tomcat 8 (Brekeke PAL WebSocket did not properly work.)
– Fixed minor bugs
For SIP server
– Added SRTP conversion
– Added provisioning feature
– Added dial plan deploy patterns plugin API
– Added option to disable TLS 1.0 or older
– Added function to send out alert emails for various alerts
– Added to support for binding RTP port with dial plan. (&net.rtp.bindsrc/&net.rtp.binddst)
– Fixed minor bugs
3.6.3.0 (November 17, 2016) – Update Release
For PBX
– Fixed the bug where email notification settings were occasionally lost after a version update (v3.5 or later).
– Fixed the bug where entered value at DID settings was not properly saved.
– Fixed the bug where audio was enabled during confirm call process on very rare occasions.
– Fixed minor bugs.
For SIP server
– Fixed the bug where the B2B-UA mode didn’t set correct branch-ID in re-INVITE’s ACK.
– Added timer for DNS resolver to clear suspension of failed resolved IP address. The default length is 10 minutes.
– Fixed minor bugs.
3.6.2.6 (October 20, 2016) – Update Release
For PBX
– Improved connection time to establish a session for call pickup/park pickup.
– Improved connection method to Brekeke CIM
– Fixed minor bugs
3.6.2.5 (September 26, 2016) – Update Release
For PBX
– Fixed the bug where a call could not be received when a semicolon is included in a display name.
– Fixed the bug where, in a rare circumstances, media server failed to response when a recording session ended abruptly.
– Fixed the bug where default user class settings were not enabled at User settings.
– Fixed the bug where template variable “&c<n>” could not use properly.
– Improved PAL WebSocket access permission for user level users.
– Fixed minor bugs.
For SIP server
– Fixed the bug where the SIP server failed to process some characters in display name.
– Fixed the bug where, under certain conditions, the failover plugin could not close a session until timeout.
– Fixed minor bugs.
3.6.2.2 (August 1, 2016) – Update Release
For PBX
– Fixed the bug where multiple users cannot be assigned to Brekeke PAL WebSocket.
– Fixed the bug where the database module template of IVR designer did not work on Java 8.
– Fixed the bug where post event script of IVR flow did not work properly.
– Fixed the bug where SRTP was not used when it was set to “Mandatory” protocol.
– Made changes how the session is handled where re-INVITE will be sent after ACK to callee.
– Add alert email feature when the maximum allowed number of call recording is reached.
– Fixed minor bugs.
For SIP Server
– Fixed the bug where TLS/WSS handling stops accepting incoming connections under certain conditions.
– Fixed bug where Failover plugin could not generate ACK/BYE packet correctly when there are high CPS.
– Fixed bug where SIP proxy did not pick “transport=” parameter from Path: header.
– Fixed minor bugs.
3.6.2.0 (June 20, 2016) – Update Release
For PBX
– Fixed the bug where default user class settings were not enabled at User settings
– Fixed the bug where calls could not be received with a particular SIP UAs and settings
– Added a capability to send an alert email to system administrators when concurrent session capacity is reached
– Fixed minor bugs
For SIP Server
– Fixed the bug where ‘reason’ cannot be stored when an IP address is added through product admintool ([Blocked IP address]).
– Fixed the bug where the registrar stops accepting REGISTER’s responses replied from Upper/Thru server.
– Fixed the bug where TLS listening became slower when there are delays in TLS-handshake
– Added SDN Diagnostic
– Fixed minor bugs
3.6.1.8 (May 31, 2016) – Standard Release
For PBX
– Modified the timing of sending NOTIFY while using blind transfer
– Fixed minor bugs
For SIP Server
– Improved performance in handling of OpenFlow
– Improved interoperability in radius
– Fixed minor bugs
3.6.1.2 (May 9, 2016) – Beta Release
For PBX
– Added a capability to transfer calls using DTMF entry while in queue
– Added a capability to update Caller ID on SIP UAs using re-INVITE request.
– Improved connection method to Brekeke CIM
– Added fields in Log Database (c_tag, c_instanceid)
– Added support of stand-alone video client.
– Added web phone support
– When SCA is used, ~line will be used as suffix whereas line- was used as prefix in previous versions
– Added some methods for Brekeke PAL Websocket
– Added “System Administrators” page to manage multiple administrator accounts
– Added capability to send alert emails to administrators
– Fixed minor bugs
For SIP Server
– Added SDN feature that controls OpenFlow switch for relying RTP, blocking malicious IP addresses and applying QoS policies. With this feature, RTP packets are relayed over the switch.
– Fixed the bug where the SIP proxy didn’t handle MESSAGE with multibyte-characters correctly when it is sent over TCP/TLS
– Fixed minor bugs
3.5.5.2 (February 12, 2016) – Update Release
For PBX
– Improved early media handling when ARS failover occurred.
– Improved session handling to prevent sending unnecessary re-INVITE requests
– Fixed the bug where CSV file for IVR log was garbled
– Fixed minor bugs
For SIP Server
– Added “Exact Match” in Block List Policy that is used to block or allow single IP address.
– Use WSS (WebSocket over TLS) for reaching a SIP client even when it points WS (WebSocket) if REGISTER was sent over WSS
– Fixed minor bugs
3.5.3.3 (November 30, 2015) – Update Release
For PBX
– Fixed the bug where cancel request was not processed properly (i.e., ACK request were not able to send.)
– Improved the memory use of Brekeke PAL WebSocket process.
– Fixed the bug where Brekeke PAL Websocket’s unhold method did not work properly.
– Fixed the bug where session counter did not work correctly for some ARS patterns
– Fixed minor bugs
3.5.3.0 (November 10, 2015) – Update Release
For PBX
– Fixed the bug in IVR script where ivr.connected() returned ‘true’ when the session is not connected.
– Script module of IVR Flow now works with java 8.
– Various voicemail setting options have been added to both user and tenant level accesses.
– Fixed minor bugs
For SIP Server
– Fixed the bug where log has been deleted prior to the date set to delete
– Added a setting “Configuration > RTP > RTP exchanger > Send before receiving (behind NAT)”
– Fixed minor bugs
3.5.2.8 (September 18, 2015) – Standard Release
– Fixed minor bugs
3.5.2.2 (September 11, 2015) – Beta Release
For PBX
– Improved user interface for ARS Field Settings
– Added configuration option of Target User/Schedule in Switch Plan(IVR)
– Fixed the bug where call transfer using Refer failed when parameter other than Replaces are included in Refer-To parse
– Added Debug Logs feature
– Fixed minor bugs
For SIP Server
– Updated the embedded DB HSQLDB version to v2.3.3
– Improved DialPlan editor
– Fixed minor bugs
3.5.0.6 (August 17, 2015) – Beta Release
For PBX
– Added call transfer function (using DTMF dial tone) while caller is in voicemail
– Improved call status display
– Fixed minor bugs
For SIP Server
– Added support for Google Cloud Messaging (GCM) for Android and Chrome
– Added [Command] page under [Tools] menu which allows to execute management commands
– Improved interoperability in Radius
– Fixed minor bugs
3.4.7.5 (July 17, 2015) – Update Release
– Fixed the bug where missed calls were ocasionally not stored in the log database
– Fixed minor bugs
3.4.7.2 (July 15, 2015) – Update Release
– Fixed the bug where product admintool were not displayed correctly when Internet Explorer or Safari
– Fixed minor bugs
3.4.7.0 (July 14, 2015) – Update Release
For PBX
– the bug where some sessions were blocked when there are a large number of REGISTER requests via ARS settings
– the bug where call hold status has not been cleared
– Added capability in Recording File plug-in to process recording even when the corresponding user has been deleted
– Added interface to voicemail feature for third-party application
– Fixed minor bugs
For SIP Server
– Fixed the bug where the Failover plug-in occasionally failed to handle late responses sent from failed destinations
– Fixed minor bugs
3.4.5.1 (May 8, 2015) – Update Release
For PBX
– Added capability to set customized DTMF timeout for both initial entry and consecutive entries
– Added SIP ;cause=200 ;text=\”Call completed elsewhere\” to Reason header when CANCEL is sent after a ring group call has been answered by a SIP UA
– Fixed the bug where the SIP listener may lose an ACK packet.
– Fixed minor bugs
For SIP Server
– Fixed the bug where restore/back up did not work for Push Notification settings and Key files
– Added Push Notification Information in [Server Status] page
– Fixed the bug where the SIP proxy didn’t accept CANCEL under a certain condition
– Keep Reason header of CANCEL if it exists
– Support Push Notification for iOS 8.2 and Apple Watch
– Fixed minor bugs
3.4.4.3 (April 22, 2015) – Standard Release
For PBX
– Increased buffer size for call recording
– Added an option to send email notification when call recording file created as a voice message file via IVR
– Added several methods for Brekeke PAL WebSocket
– Added ‘media before answer’ option to allow connection before a call answered when a call are transferred via Auto Attendant or IVR
– Fixed minor bugs
For SIP Server
– Added support for Apple Push Notification Service (APNS) for iOS
– Improved the method to detect interface IP addresses
– Strengthen TCP/TLS stability when a SIP UA closes connection unexpectedly
– Improved TLS/WSS connectivity
– Improved Block-List feature to detect malicious connections over TCP/TLS/WebSocket
– Improved Block-List feature to detect invalid SIP packets further
– Allow to bind a transport protocol (TCP/TLS/WS/WSS) to a certain interface
– Added PKCS#12 file support for TLS/WSS
– Added new Dial Plan method $rule.sps to calculate the Session-per-Second per Dial Plan rule
– Added new Dial Plan method $route.sps to calculate the Session-per-Second per route
– Fixed the bug where the $action=block didn’t work under a certain condition
– Fixed the bug where the [Interface address] definition is applied unexpectedly
– Fixed minor bugs
3.4.3.5 (February 23, 2015) – Beta Release
For PBX
– Added tag feature for received calls
– Removed WebRTC option and add the function to all edition
– Improved connection process (ICE, DTLS) on WebRTC
– Ability to apply Phone Type changes without restarting Brekeke PBX
– Fixed the display issue with IE8
– Fixed the bug where call log is failed to be created
– Fixed minor bugs
For SIP Server
– Improved TLS connectivity
– Improved network interface addresses discovery
– Added WebSocket in Standard Edition
– Fixed minor bugs
3.4.2.3 (November 24, 2014) – Beta Release
For PBX
– Updated look & feel of the Admintool
– Added WebRTC connectivity
– Improved RTP packet handling for video
– Fixed minor bugs
For SIP Server
– Improved WebSocket connectability
– [Evaluation Edition] 2 WebSocket connenctions added
– [Evaluation Edition] 2 TLS connections added
– Improved look & feel of the Admintool
– Fixed minor bugs
3.3.9.3 (January 26, 2015) – Update Release
For PBX
– Fixed the bug where timer and schedule did not work correctly when the system time switches to and from summer time
– Fixed a rare bug where media server get disconnected
– Fixed the bug where failed to record calls after a call transfer
– Fixed the bug where the configuration settings of RFC2833 in [Options] did not set correctly
– Fixed minor bugs
For SIP Server
– Added “pdd-time” attribute in Radius Accounting
– Fixed the bug where Radius Accounting didn’t put actual remote IP address in “h323-remote-address” attribute under certain conditions
3.3.8.1 (October 16, 2014) – Update Release
For PBX
– Fixed the bug where initiating call recording or IVR
– Fixed minor bugs
For SIP Server
– Improve the Failover plugin for handling response packets received with delays
– Fixed minor bugs
3.3.7.4 (September 19, 2014) – Update Release
For PBX
– Ability to accept 60msec of codec G.729
– Fixed the bug where handling 40msec of codec G.729.
– Fixed the bug where codec handling of incoming sessions
– Fixed the bug where call parking placed with the keypad command did not work properly when pick up was tried before the call disconnected by the person who was placing the parked call
– Fixed minor bugs
For SIP Server
– Fixed the bug where the Radius Accounting fails under heavy load
– Fixed the bug where the TCP handling occasionally freezes when the TCP connection is disconnected unexpectedly
– Fixed minor bugs
3.3.5.8 (July 1, 2014) – Update Release
For PBX
– Fixed the bug that call park placed using dial keypad command does not function correctly when picked up before hanging up
– Fixed the bug in call conference feature (v3.3.5.4 only)
– Changed the default setting length to determine deletion of the recorded file to 2 seconds from 0.2 seconds
– Fixed the bug that connection was allowed to use Brekeke PAL WebSocket when incorrect password was used
– Fixed minor bugs
For SIP Server
– Fixed the bug that the details of Dial Plan History cannot be displayed
– Fixed the bug where re-INVITE after 407 might not be resend correctly
– Fixed the bug where keep-alive SIP packet doesn’t have the valid SIP-URI in To header
– Fixed the bug where the SIP Server didn’t parse OS’s network configuration file correctly in certain Linux distributions
3.3.5.4 (April 28, 2014) – Update Release
For PBX
– Fixed the bug that occurs rarely, which the same user name cannot be created after a deleting a user
– Fixed the bug that occurs rarely, which call transfer failed when it is transferred from an auto attendant
– Fixed the bug that user class was not applied correctly
– Fixed minor bugs
3.3.4.4 (March 3, 2014) – Standard Release
For PBX
– Added option to specify FROM as conference number when conference is called automatically.
– Fixed the bug that multiple callee are sporadically connected when simultaneous ring is used.
– Fixed the bug that processing SDP on 18x
– Ability to set result in script module of IVR Flow
– Fiexed minor bugs
For SIP Server
– Return 603 response when the unprocessed time of the request packet passed predefined length
– Improved performance in Failover plugin
– Improved the server to perform properly releasing resources
– Fixed the bug where the [Active Sessions] page couldn’t disconnect certain sessions
– Fixed minor bugs
3.3.3.1 (January 31, 2014) – Standard Release
For PBX
– Added password encryption when Brekeke PAL WebSocket is used
– Added recording status on Brekeke PAL WebSocket
– Fixed minor bugs
For SIP Server
– Support certificate chaining for TLS
– Support PEM formatted certificate and key files for TLS
– Fixed the bug where the TCP/TLS connection might not accept REGISTRER if it indicates “Contact: *”.
– Fixed the bug where the [Active Sessions] page didn’t show sessions made by the previous day.
– Fixed the bug where the [Active Sessions] page might not show sessions correctly if there are SUBSCRIBE sessions.
– Fixed the bug where the [Remote Address Pattern] didn’t work if it is for the exiting interface address.
3.3.1.2 (December 11, 2013) – Beta Release
For PBX
– Added video stream support
– Improved SDP handling
– Fixed minor bugs
For SIP Server
– Added RFC4574 (SDP Label Attribute) support
– Added RFC4796 (SDP Content Attribute) support
– Improved NAT Keep-Alive handling
– Improved DNS configuration
– Improved Block-List feature to detect war dialing
– Fixed minor bugs
3.2.4.3 (September 30, 2013) – Update Release
For PBX
– Added @<registered phone ID> as one of the value option at Target filed of ARS Out pattern.
– Fixed minor bugs
For SIP Server
– Fixed the bug where the SIP proxy didn’t accept certain kinds of Via header.
– Changed the default value of [Block List] > [Settings] > [Attempt Tracking] > [Multiple Accesses].
– Fixed minor bugs.
3.2.4.1 (September 20, 2013) – Update Release
For PBX
– Call parameters could not be set correctly at ARS Custom field.
– Fixed minor bugs
For SIP Server
– Fixed the bug where the SIP proxy kept sending 407 to SUBSCRIBE
– Fixed the bug where the Mirroring function didn’t handle TLS
correctly after failover happened.
– Fixed the bug where the Mirroring function didn’t work when both TLS and TCP are used simultaneously.
– Added the [Do not Block Local IP Address] in the Block List function for excluding SIP packets sent from local network
– Fixed minor bugs
3.2.3.6 (Aug 29, 2013) – Standard Release
For PBX
– Fixed the bug where Brekeke PBX cannot get the correct status with Brekeke PAL option
– Fixed minor bugs
For SIP Server
– Add [Block List Status] in the [Server Status] page
– Add User-Agent in the [Call Logs] page
– Added the [Pattern] in [Interface address] settings which used for selecting a matched interface address
– Added new field [Adjusted Expires] in the [Configuration]->[SIP] page which adjusts REGISTER expiration period
– Added searchable filter by User-Agent at [Registered Clients] page
– Fixed minor bugs
3.2.2.9 (Aug 9, 2013) – Beta Release
For PBX
– Ability to change Busy Forwarding Destination according to response code
– Ability to accept 10msec/40msec of codec G.729
– Changed connection method for PAL WebSocket
– Fixed minor bugs
For SIP Server
– Fixed minor bugs
3.2.2.3 (July 26, 2013) – Beta Release
For PBX
– Added SRTP support
– Improved log database
– Added MWI (NOTIFY without SUBSCRIBE) at Phone Type settings
– Added Paging at Inbound settings
– Added Tenant initial value field in Field settings (Route Local Settings) in ARS menu
– Enabled order changes for ARS Route (Variables)
– Added &template and &name as ARS Variables
– Added PAL WebSocket
– Improved initial startup time when there are large number of users
– Fixed minor bugs
For SIP Server
– Improved performance in DNS resolver
– Improved performance in Radius client
– Improved TLS handling
– Improved DialPlan syntaxes
– Added Error Log feature
– Added Block List feature
– Fixed the bug where the SIP proxy didn’t allow to relay RTP between AVP and SAVP
– Added Auto Sync for User Authentication
– Added drag and drop feature to change priorities for DialPlan rules
– Fixed minor bugs
3.1.9.1 (Aug 6, 2013) – Update Release
For SIP Server
– Fixed the bug where CANCEL request might be rejected with 481 under Advanced Edition.
– Fixed the bug where SUBSCRIBE session doesn’t timeout in “active” status.
3.1.9.0 (July 15, 2013) – Update Release
For PBX
– Fixed the bug where RTP relay did not work when PBX tries to use the port that is already in used by other application (Windows OS)
– Fixed the bug where call recording did not work when PBX tries to use the port that is already in used by other application (Windows OS)
– Fixed the bug in Japanese language display where user type can be changed even when one is logged in as a user
For SIP Server
– Fixed the bug where RTP relay did not work when SIP server tries to use the port that is already in used by other application (Windows OS)
– Fixed the bug where SIP server reject CANCEL with a response, “481 Call Leg/Transaction Does Not Exist”, when CANCEL was received before INVITE session was made.
3.1.8.2 (June 04, 2013) – Update Release
For PBX
– Fixed the bug where KEE_COPY property didn’t work in RecordingFileHttpUploader plug-in
– Fixed the bug where ARS settings were not saved correctly when Backup/Restore is used
– Fixed the bug where restriction of IP addresses didn’t work with 3PCC
– Fixed minor bugs
For SIP Server
– Fixed the bug where value set at “Timeout” and “Interval” were not applied at Heatbeat setting
– Fixed the bug where the TLS listener didn’t handle the case correctly when connection timeout happen during the handshaking process
– Fixed minor bugs
3.1.7.8 (March 26, 2013) – Update Release
For SIP Server
– Fixed the bug where SIP server could not handle RTP relay when the size of the RTP packet is large
3.1.7.7 (March 14, 2013) – Update Release
For PBX
– Fixed the bug where media server occasionally stops when call recording had ended and restarted.
– Added broadcast option to conference feature
– Fixed the bug where ARS feature mulfucntion when pbx file from a previous verison is restored at v3.1.x.x
– Fixed minor bugs
3.1.7.0 (March 1, 2013) – Update Release
For PBX
– Fixed the bug where RTP session timeout did not work under certain conditions
– Fixed the bug where timeout for transfer from IVR was not handled accurately
– Fixed the bug where the log parameters of recording file in the property file was incorrectly entered
– Improved call recording performance
– Fixed minor bugs
For SIP Server
– Fixed the bug where the SIP proxy did not handle CANCEL correctly when the destination requests INVITE authentication
– Fixed the bug where RTP session timeout did not work under certain conditions
– Fixed minor bugs
3.1.5.8 (January 24, 2013) – Update Release
For PBX
– Fixed the bug where system deletes recorded file in next day when [Conversation recording file in voicemail inbox] is set to “no”.
– Fixed the bug where call cannot be made when display name of “To” header includes “&” character.
– Fixed the bug where secondary server settings are not reflected when ARS route has been deleted on the primary server.
– Fixed minor bugs
For SIP Server
– Fixed the bug where the SIP proxy might not release session resources when a subsequent SIP request contains an invalid From-tag.
3.1.5.4 (January 16, 2013) – Update Release
For PBX
– Fixed the bug where voice prompt’s audio quality depleted when converting PCM16 to ULAW format
– Fixed the bug where call forwarding to call queue fails
– Corrected header display at user call log
– Fixed minor bugs
For SIP Server
– Fixed the bug where the TCP/TLS handling couldn’t send a reply if Contact header is not parsable.
– Fixed the bug where the SIP proxy couldn’t parse “multipart/mixed” content correctly in the
default settings.
– Added the [Certificate Information] in the Status page when TLS is enabled.
– Fixed minor bugs
3.1.4.4 (December 14, 2012) – Standard Release
For PBX
– Improved call log feature and added link to download call recording files from the call log view.
– Added web service interfaces for modifying the ARS route values.
– Fixed some bugs in the SCA feature
– Fixed the bug where the ARS session counter occasionally did not work correctly.
– Made some improvements on ARS/DID pages
– Fixed the early media was not relayed correctly with 3PCC calls.
– Improved user access.
– Fixed minor bugs
For SIP Server
– Fixed the bug where the memory leak occurs when there are heavy traffic.
– Fixed the bug where the Advanced Edition might not release TCP/TLS resources when a phone disconnects TCP/TLS connection in each SUBSCRIBE request.
– Fixed the bug where the SIP server may not automatically enable RTP-relay in the default settings when content-type contain a capital letter.
– Fixed the bug where the SIP server may not parse some formats of body which sent from TCP/TLS.
– Added GUI settings for authenticating SUBSCRIBE and MESSAGE requests.
– Fixed minor bugs
3.1.3.0 (November 9, 2012) – Beta Release
For PBX
– Added an option to terminate to conference call when the host leave the line.
– Made some improvement on ARS/DID pages
– Fixed the bug where callback did not work when the caller’s [Max Inbound Session] = 1.
For SIP Server
– Improved performance when executing Session Plug-in with TCP/TLS connection.
– Fixed the bug where Record-Route headers in response packets occasionally replaced incorrectly when TLS is used.
– Fixed the bug where $transport, $ifsrc and $ifdst values were cleared if multiple rules were matched.
– Fixed the bug where Mirroring Virtual IP address was not applied as an interface address.
– Fixed the bug where TCP/TLS connecter causes a freeze.
– Fixed minor bugs
3.1.1.2 (August 17, 2012) – Beta Release
For PBX
– Added an option to set a user’s name as caller’s display name
– Improved ARS settings
– Added timer settings for switching the Inbound plan
– Discontinued setArsVariables method in Web service
– Fixed minor bugs
For SIP Server
– Added Radius port sharing
– Added DialPlan History
– Added DialPlan Concurrent Call Control
– Added the “$goto” method for Deply Patterns in DialPlan
– Added the “$radius.auth” method for Matching Patterns in DialPlan
– Fixed the bug where RTP-relay could not detect timeout
– Fixed the bug where the Failover plug-in forwarded PRACK incorrectly under a certain condition
– Fixed the bug where a spiraled session more than 3 hops cannot be disconnected by server.
– Fixed minor bugs
3.0.7.0 (July 20, 2012) – Update Release
For PBX
– Fixed the bug where tenant settings are failed to be applied without restart
– Fixed the bug where schedule was not applied when the end year field is left empty.
– Improved security for logging in the product administrative tool.
– Improved shutdown process
– Fixed minor bugs
For SIP Server
– Fixed the bug where the Subscribe Session does not accept NOTIFY at times.
– Fixed minor bugs
3.0.6.3 (June 31, 2012) – Update Release
For PBX
– Fixed the bug where Call Forwarding with 3xx response from callee did not work.
– Improve voice quality when there are multiple G.729 sessions used.
– Improve the upgrade method for version 2 to version 3.
– Fixed the bug where the license randomly resets when the program restarted.
– Fixed minor bugs
For SIP Server
– Change the default value of IPv6 to “off”
– Change the default value of TLS-handling to “off”
– Change the default value of Check Request-URI’s validity to “off”
– Fixed a bug that the hide loopback didn’t work.
– Fixed minor bugs
3.0.5.5 (May 23, 2012) – Standard Release
For PBX
– Fixed a bug where call recording did not work under certain operations
– Fixed a bug where auto monitoring did not work under certain operations
– Fixed a bug where input rule in IVR designer did not work
– Fixed a bug where 3PCC did not work with IVR
– Fixed a bug where date condition did not work correctly
– Fixed minor bugs
For SIP Server
– Added Failover plugin
– Increased the default max number of shared SIP session per thread to 50
– Fixed the bug where DNS SRV for load-balancing did not work under certain conditions
– Fixed minor bugs
3.0.3.8 (April 4, 2012) – Beta Release
For PBX
– Fixed minor bugs
For SIP Server
– Fixed minor bugs
3.0.3.2 (March 20, 2012) – Beta Release
ForPBX
– Added Export/Import Flows at IVR designer.
– Fixed memory leak of RTP handling.
– Fixed minor bugs
For SIP Server
– Add Load Balancing feature using DNS SRV (RFC2782)
– Support DNS SRV for TLS/TCP
– Fixed minor bugs
3.0.2.1 (February 3, 2012) – Beta Release
For PBX
– Support multiple phone ID
– Added IVR designer feature
– Added Remote Call Back feature
– Improved administrative tool
– Added user access settings
– Added 64-bit support
– Fixed minor bugs
For SIP Server
– Added IPv6 support
– Added TLS support
– Added Radius Pre-Authentication feature
– Added preliminary dial plan settings
– Improved administrative tool
– Improved product performance
– Removed Solaris OS support
– Fixed the bug where timeout didn’t work with TCP handling
– Fixed minor bugs
2.4.9.0 (May 5, 2011) – Update Release
For PBX
– Fixed the bug where KEEP_COPY parameter in RecordingFileHttpUploader of Audio File Plug-in did not work.
– Fixed the bug where failed to release memory during SUBSCRIBE session
– Fixed the bug where Brekeke PAL did not retrieve correct information when features like call pick up is used
– Fixed the bug where authorized settings for tenant administrator in Multi-Tenant Edition
– Fixed the bug where SCA’s information did not correctly removed
– Add option to include Phone ID in log database
2.4.8.6 (March 25, 2011) – Update Release
For PBX
– Optimized performance for BLF (Busy Lamp Field).
– Fixed the bug where Brekeke PAL had problem with releasing memory when it is used with a particular SIP UA.
For SIP Server
– Fixed the bug where Import Users was not working correctly.
– Support Redirection by Failover plugin.
– Fixed the bug where Failover plugin with B2B mode didn’t clear resending packets.
– Added session plug-in for modifying SDP content.
– Fixed the bug where the proxy could not handle CSeq which Cisco BTS 10200 Softswitch generates.
– Fixed the bug where the registrar may not point reachable port number if a user registers over TCP.
– Fixed the bug where the webget will hang if a destination web server doesn’t respond.
– Fixed the bug where a destination of resending “200 OK” responses may differ from the first packet.
2.4.7.3 (October 11, 2010) – Update Release
– Fixed the bug whersockets for internal commands remain occasionally.
– Minor changes related to the debug log.
– Improved security for admintool login.
2.4.7.0 (September 17, 2010) – Update Release
For PBX
– Fixed the bug with Multi-Tenant Edition where Call Forwarding (Round-Robin/Top down) feature did not properly work.
– Fixed the bug where session disconnection time was set as zero in log database.
– Fixed minor bugs.
For SIP Server
– Fixed the bug where $request definition in Deploy Patterns did not work.
– Fixed the bug where Upper/Thru registration did not include some headers and parameters.
– Included Failover-Plug-in
– Fixed minor bugs
2.4.6.7 (August 20, 2010) – Update Release
For PBX
– Added more improvement on security of administrative tool for preventing cross-site request forgery.
– Improve product performance when there are large number of voicemail exists.
– Fixed minor bugs.
For SIP Server
– Fixed minor bugs.
2.4.5.5 (June 18, 2010) – Update Release
For PBX
– Fixed the bug where a call is occasionally disconnected when attenadnt transfer feature is used.
– Fixed the bug where the defualt Audio File plug-in failed to upload or move files.
– Improve security of administrative tool for preventing cross-site request forgery.
– Fixed minor bugs.
For SIP Server
– Fixed minor bugs.
2.4.4.8 (May 14, 2010) – Update Release
For PBX
– Fixed the bug where line key remained lit while using SCA feature.
– Improved Brekeke PAL interface.
– Fixed the bug where tenant ID cannot be modified with log database export.
– Fixed the bug where files deleted at Default Audio File Plug-in “RecordingFileHttpUploader”, “RecordingFileMove”.
– Fixed minor bugs.
For SIP Server
– Improved Alias page.
– Fixed minor bugs.
2.4.3.9 (Feb 26, 2010) – Update Release
For PBX
– Fixed the bug where 2-step dialing did not work properly
– Send a response code to callee when 3PCC feature used via Brekeke PAL interface
– Change the smallest value of switch pattners to 1 (it was 2 before)
– Fixed the bug where tenant’s note was not backed up with Brekeke PBX Multi-Tenant Edition
– Fixed the bug where a session terminated earlier than the time set at session timer
– Improved Brekeke PAL interface
– Improved Brekeke CTI Server interface
– Fixed minor bugs
For SIP Server
– Fixed the bug where Unregister button failed to respond in multi-domain mode
– Fixed the bug where “Disconnected by System” recorded at call log when the session is not disconnected by system.
– Fixed minor bugs
2.4.3.0 (Dec 18, 2009) – Standard Release
For PBX
– Fixed the bug that known registration problem at ARS feature (issue existed v2.4.2.2 only)
– Fixed the bug where call hold and seize feature when SCA is used
– Fixed the bug related to ARS variables
– Fixed minor bugs
For SIP Server
– Fixed minor bugs.
2.4.2.2 (November 20, 2009) – Beta Release
For PBX
– Added IVR Script (Option)
– Improved PAL interface
– Fixed minor bugs
For SIP Server
– Added a SIP monitoring method for Heartbeat.
– Added auto start for Heartbeat.
– Fixed minor bugs.
2.3.8.4 (November 20, 2009) – Update Release
For PBX
– Fixed the bug where counting the number of ARS route sessions
For SIP Server
– Fixed the bug where Heartbeat settings failed to send alert email messages.
– Fixed minor bugs
2.3.8.2 (September 24, 2009) – Update Release
For PBX
– Fixed a bug where a session counter of ARS did not work correctly.
– Fixed a bug where ARS failover did not work correctly when [Disable on failure]=”This group”.
– Fixed minor bugs
For SIP Server
– New variable for DialPlan
& register.contact.remote – If true, SIP server uses the remote IP address to reach UA instead of using Contact URI.
– Fixed minor bugs
2.3.7.4 (July 22, 2009) – Update Release
For PBX
– Fixed bug related to Hold using SCA.
– Fixed minor bugs.
For Sip Server
– Fixed the bug where Heartbeat settings did not accept Remote Action settings correctly.
– Fixed the bug where Database settings lost the authentication password.
2.3.6.0 (May 7,2009) – Standard Release
For PBX
– Fixed memory bugs for PAL add-on.
– Fixed Hold and Talk remote control feature for PAL.
– Fixed minor bugs.
For SIP Server
– Fixed the bug where agents could not register from Admin GUI.
– Display **** for Password at Heartbeat page.
– Fixed the bug where restore did not work when there is no Heartbeat settings.
– Fixed minor bugs.
2.3.4.8 (April 17, 2009) – Beta Release
For PBX
– SSL for Email notification (added for heartbeat also)
– Some bug fixes related to Music on hold when you are in a conference or call recording.
– Changed default dial plan.
– Fixed a bug that you could not have 250 concurrent calls on linux.
– Optimized performance for sending NOTIFY.
– Minor GUI change.
– Reduced PAL notification packets, improved notification reliability.
– Fixed minor bugs.
For SIP Server
– Minor GUI changes
2.3.3.0 (March 27, 2009) – Beta Release
For PBX
– Added [Encrypted Connection (SSL)] field at [Options] for email notification.
– Optimized the number of threads.
– Removed action icons from user and message pages in favor of checkboxes and links.
– Fixed minor bugs.
For SiP Server
– Fixed a bug where the Brekeke SIP Server had the possibility of leaving SIP sessions.
2.3.1.8 (February 20, 2009) – Alpha Release
For PBX
– BLF(Busy Lamp Field), SCA (Shared Call Appearence), Presence (Polycom, X-Lite) support.
– RFC2833 at [Options] page.
– Fixed minor bugs.
For SIP Server
– Advanced Edition can handle multiple registration jobs on one thread.
– Advanced Edition can configure Mirroring and Redundancy settings on the GUI.
– Reject invalid REGISTER packets with “400 Bad Request”.
2.2.7.8 (March 11, 2009) – Version Update
For PBX
– Changed default dialplan.
For SIP Server
– Default values of authentication for REGISTER and INVITE are “on”
2.2.7.7 (March 9, 2009) – Version Update
for PBX
– No changes.
for SIP Server
– Fixed a bug where the Brekeke SIP Server Admintool may lose some settings under certain operations.
2.2.7.6 (March 6, 2009) – Version Update
for PBX
– Fixed the bug where there is no audio when call has been picked up with RTP relay set to “off”.
– Added default regular expression (^.+$) at user field under ARS Pattern OUT
– Fixed the bug related to 302 response
– Added restriction where calls will be rejected when the caller/callee is neither a PBX user nor going through ARS route settings
– Fixed the bug related to RTP relay
for SIP Server
– Fixed minor bugs
2.2.6.2 (November 20, 2008) – Version Update
For PBX
– Fixed the bug related to regular expression in ARS variables.
– Fixed minor bugs.
For SIP Server
– Fixed some bugs in the Mirroring-Mode.
– Fixed the bug where the Thread-Sharing did not handle multiple resends correctly.
– Fixed a minor bug in the NAT-detection.
2.2.5.8 (November 5, 2008) – Official Release
for PBX
-Fixed rare Access Violation that happens when PBX handles many concurrent call-recording sessions.
-Fixed the bug where calls could not be disconnected when a caller hangs up the call using auto attendant.
-Fixed the bug where [Codec Priority] at the Media Server settings was not applied.
-Fixed minor bugs.
for SIP Server
– Fixed some bugs in the Mirroring-Mode.
– Fixed the bug where the SIP server didn’t accept the setting of “RTP relay (UA on this machine)”.
– Improved the NAT detection.
2.2.4.5 (Sept 26, 2008) – Beta Release
for PBX
– Import .CSV file of ARS variables
– Fixed Minor Bugs
for SIP Server
– Advanced Edition can handle multiple SIP sessions on a thread.
– Advanced Edition can pre-create threads for SIP sessions.
– DialPlan’s Deploy Pattern accepts ‘#’ in a rule.
– New variables for DialPlan’s Deploy Pattern
$replaceuri.from – replace From’s SIP-URI
$replaceuri.to – replace To’s SIP-URI
(These defaults are “false”.)
– Fixed the bug where the B2B-UA mode doesn’t handle a spiral correctly.
– Fixed the bug where the registrar doesn’t accept a request if it doesn’t have Contact header.
2.2.1.6 (May 9, 2008) – Beta Release
for PBX
– Fixed Caller ID bug related to Auto-attendant and Call Recording
for SIP Server
– DialPlan’s Matching Pattern accepts the following definitions.
From = sip:user;para1=xxx@addr;para2=yyy
From =
2.2.1.5 (May 5, 2008) – Beta Release
for PBX
– Added Paging functions
– Added Registration Subscribe/Notify for PAL
– Fixed Minor Bugs
for SIP Server
– Support TCP for Upper/Thru registration
– Support TCP for UPnP
– Fixed the bug where SIP exchanger doesn’t handle spiral over TCP.
– DialPlan’s Matching Pattern accepts the following definitions.
To=sip:user;para1=xxx@addr;para2=yyy
To=
2.2.1.1 (March 28, 2008) – Alpha Release
for PBX
– Added Tutor Mode
– Fixed minor bugs
for SIP Server
– Fixed the bug where NOTIFY/OPTIONS/MESSAGE messages consume system memory and cause a system to go down.
– More stable TCP connectivity
– Support switching of transport in mid-session.
– Follow the RFC more tightly.
– DialPlan: Response Header Definition
– Send 407 before sending 404. In previous version, the SIP Server sends 404 when the not-found happens even if the authentication is enabled. From this version, the SIP Server authenticates the request before it sends 404.
2.2.0.7 (March 7, 2008) – Alpha Release
for PBX
– Added Confirm Call feature.
– Added Custom Voice Prompts page.
– Fixed minor bugs.
for SIP Server
– Added TCP support.
– Added B2B-UA Mode.
– Fixed the bug where the text message decoding/encoding don’t work with some languages.
2.1.6.6 (February 19, 2008)
for SIP Server
– Correctly handle REGISTER request packets containing Contact:*.
– Fixed the bug where configuring 3rd Party Database causes an Alias Database exception.
2.1.6.2 (December 5, 2007)
for PBX
– Fixed minor bugs
for SIP Server
– Corrected 3xx reponse’s Contact header.
2.1.6.1 (December 3, 2007)
– Fixed minor bugs
2.1.5.6 (October 26, 2007)
for PBX
– Fixed minor bugs
for SIP Server
– Fixed the bug where the SIP exchanger consumes system memory.
– Fixed the bug where the Upper/Thru Registration feature consumes system memory.
2.1.5.2 (October 2, 2007)
for PBX
– Fixed a bug that SIP Server was not included in Brekeke PBX Evaluation
2.1.5.1 (Sept 21, 2007)
for PBX
– Fixed bug where unsaved voicemail could be downloaded without entering the password
– Changed the music-on-hold sound file
– Improve email sending process (POP before SMTP)
– Fixed minor bugs
for SIP Server
– Advanced Edition was added to the Product Line
(The Edition Comparison is at http://www.brekeke.com/products/products_sip_2.php)
– Added ability to modify SDP’s addresses using the DialPlan
– Fixed minor bugs
[Advanced Edition]
– Added Alias Database management through the Admintool
– Added Web/SOAP DialPlan Interfaces
– Added the Multiple Targets Failover
– Ability to change User-Agent/Server headers
2.1.2.2 (Aug 23, 2007)
for PBX:
– Fixed the bug that the call entered in to call queue was disconnected occasionally.
– Fixed the bug that a PBX user (callee) was called occasionally even [Ringer Time] was set to 0.
– Fixed the bug that some dll did not work correctlly during software update.
– Fixed minor bugs
2.1.1.3 (July 31, 2007)
for PBX:
– Added Backup and Restore Management feature
– Allows creation of ARS Plugin
– Added ability to create notes
– Added a filter to Users List page.
– Allows modification to body and subject of notification emails.
– Fixed minor bugs
for SIP Server:
– Handles the telephone-subscriber format
– Includes Alias plug-in
– The Deploy Pattern can set $request as the new request line
– Fixed minor bugs
2.1.0.4 (May 22, 2007)
for PBX:
– Fixed the bug that doesn’t handle 30X response
– Fixed minor bugs
for SIP Server:
– Fixed the bug wher NAT-Keep-Alive feature monopolizes system resources
– Fixed the bug where the [Disconnect] button does not close second-spiralled sessions
– Fixed minor bugs
2.1.0.1 (May 4, 2007)
for PBX:
– Ability to edit property file by the GUI.
– Supports the new PBX Active Library (PAL) control
– Improved the DTMF detection
– Fixed minor bugs
for SIP Server:
– Supports spiral (legal loop)
– Allows editing of property files through the Admin tool
– Fixed minor bugs
2.0.7.2 (Mar 20, 2007)
for PBX:
– Improved voice quality when converting from G.711-Alaw to ILBC or G.729
– Fixed a minor bug
for SIP Server:
– Fixed a bug that missing Email field prevents user data import.
– Fixed a minor bug
2.0.7.0 (Feb 14, 2007)
for PBX:
– Call Forwarding methods when [Round Robin] is set has been updated and improved.
– Removed version display at SIP UAs.
– When voicemail is checked at Call forwarding settings, the entry field is left blank.
– Default length for [Ringing Timeout (ms)] under Option menu is changed to 4 minutes from 2 minutes.
– MARK flag in RTP packets is set to “off”.
– Fixed bug with cancelling calls when call was terminated simultaneously when session was connected.
– Fixed bug for when a UA rings for a second while the [Ringer time (sec)] setting was “0”.
– Fixed bug for returning “481” against “PRACK” when it has unexpected RSEQ.
– Fixed bug where the timestamp for RTP packets becomes out of order when a blind transfer is initiated from user agent/SIP phone.
for SIP Server:
– Fixed a bug that CANCEL requests might contain updated “sent-by” value.
– Fixed a minor bug
2.0.5.8 (Jan 10, 2007)
for PBX:
– Changed the behavior for retrieving the call after disconnecting when the call is on Hold/Transfer/Call parking
– Changed the default ARS settings/Dialplan
– Fixed some minor memory leaks
– Improved performance
– Minor bug fixes
for SIP Server:
– Registered Clients page now shows User Agent names.
– Checks the router’s existence frequently.
– Fixed a bug that the SIP Server missed a URI-parameter.
– Fixed a minor bug
2.0.4.4 Beta (Dec 13, 2006)
for PBX:
– Fixed a problem with the mediaserver from 2.0.4.1 beta
– Fixed the bug related to a memory leak.
– Changed the behavior for returning to a conversation by dialing “**” after a call is dropped while the call is on hold.
2.0.4.1 Beta (Dec 5, 2006)
for PBX:
– Fixed a problem related to music-on-hold.
– Fixed the bug which comes from version 2.0.3.1 with incorrect SDP.
– Fixed the bug related to RTP timestamp.
– Removed the dialplan rule “To MediaServer”. The PBX creates the session directly to the MediaServer.
– Start/Shutdown page.
– Combined SIP Server [Start/Shutdown] and PBX [Start/Shutdown] into one page for easier management.
for SIP Server:
– Cache the UPnP port mappings.
– Status page shows the peak number of sessions.
2.0.3.2 Beta (Nov 10, 2006)
for PBX:
-Added Codec priority setting in [ARS] and [User] menu.
-Supports receiving “Replaces” header.
-Supports Third Party Call Control interface.
-Minor bug fixes.
for SIP Server:
– Supports UPnP for detecting a router and its global IP address, and making the port mapping.
– Ability to specify the pattern of additional external IP addresses.
– Ability to search the string from the SDP by using the DialPlan.
2.0.2.4 Beta (Oct 06, 2006)
for PBX:
-Supports RTCP only when RTP relay=on
-Supports 100rel
-Supports Session Timer
-Added Codec priority setting in [Options] menu
-Minor bug fixes
for SIP Server:
-Supports “multipart/mixed” content type.
2.0.1.6 Beta (Sept 11, 2006)
for PBX:
– Included a new feature called “Automatic Monitoring” it allows the designated extension to monitor the calls made from/to the particular extension .
– Minor bug fixes
for SIP Server:
– Added the Dial Plan plug-in interface
– Added the redirection feature (by sending 3xx response)
– Added the Multiple Domains mode
– Added the IP address filtering
– Supports DNS SRV for detecting a session destination
– Supports Multiple Transport types (not only “RTP/AVP”)
– Minor bug fixes
2.0.1.2 Alpha (Aug 23, 2006)
for PBX:
– Included new customizable ARS templates for supported Internet service providers.
for SIP Server:
– Fixed a Minor bug
2.0.0.9 Alpha (Aug 14, 2006)
for PBX:
– Changed product name from OnDO PBX to Brekeke PBX
– New web user interafce
– The SIP sever for PBX was embedded in the PBX
– Added sub-menu “Busy Forwarding” in menu “User Setting”
– Voice mail notification’s email message is customizable
for SIP Server:
– Added sub-menu “Miscellaneous” in menu “Config”
– From this version on, SIP listening port can be any other UDP port besideS port 5060 and TCP port for embedded HSQL database connection can be any other TCP port besideS port 9001
– Added List Filter to “Registered”
– Added List Filter to “Session”
– Added List Filter to “Call Log Viewer”
– Added Dial Plan check box to hide disabled rules
– Ability to IMPORT dial plan rules
– Ability to EXPORT dial plan rules
– Ability to EXPORT Users in “Authentication”
1.5.3.0 (July 31, 2006)
– Fixed the bug that stops pbx running when logging off Windows
– Corrected the problem of attended transfer for some SIP phones.
– Fixed the bug that drops the call, after transferred, when it reaches the maximum value set in the field “Conversation recording length (sec)”.
1.5.2.0 (April 10, 2006)
– Fixed the RTP relay problem, caller can not hear callee, when it is set to on(G.711 u only) after the ARS fail over
– Fixed the intermittent problem of RTP relay
– PBX now can handle Alaw 30ms packet correctly.
1.5.1.5 (Jan 4, 2006)
– Reduced the buffer size for handling RTP (it was too big at previous versions 1.4.5.0 – 1.5.1.3).
1.5.1.3 (Dec 16, 2005)
-Fixed a bug that Call Park didn’t work when Call Park number range (Park number (min) & Park nmber (max)) set in Option menu is not 2 digits.
1.5.1.1 Beta (Nov 30, 2005)
– Added a feature that a user can call multiple users at a time for starting a conference by dialing +*+*+…
– Fixed a bug that listening-only mode didn’t work even you dialed the prefix 7* to join in the conversation.
– Fixed minor bugs
1.5.0.8 Beta (Nov 17, 2005)
– Add “Call Status” page in the OnDO PBX Admintool to show status of active calls.
– Fixed a bug that ARS fail over does not work when making calls to conference members.
1.5.0.4 Alpha (Nov 9, 2005)
– Added ARS failover feature (Only for Standard Edition)
– Supports to set multiple Call forwarding patterns in User Setting
– Fixed the problem that Rewind (dial 7) and Wind-forward (dial 9) didn’t work when
reviewing a voicemail message.
1.4.5.0 Beta (Aug 19, 2005)
– Added an option for SMTP authentication = on or off.
– Fixed the problem that periodical REGISTERs from PBX to
other SIP Server (such as ITSPs) stopped when no response
is arrived at OnDO PBX.
– From this version, OnDO PBX will retry sending REGISTER
request to other SIP Server (such as ITSPs) after a fixed
time even Authentication error has occurred.
By default, the fixed time = 1800000 milliseconds (30 minutes)
1.4.4.5 (Oct 18, 2005)
– Fixed the problem that voice wasn’t transmitted from OnDO PBX when RTP relay = on
at OnDO PBX.(It happened on very rare occasions)
– Fixed the problem that user can not hear audio after transfering a call using #9
when Call Recording was enabled. (It happened on very rare occasions)
– Fixed the problem that RTP relay stopped working after PBX receiving re-INVITE from UA.
(It happened on very rare occasions)
1.4.4.3 (Aug 19, 2005)
– Added an option for SMTP authentication = on or off.
– Fixed the problem that periodical REGISTERs from PBX to
other SIP Server (such as ITSPs) stopped when no response
arrived at OnDO PBX.
– From this version, OnDO PBX will retry sending REGISTER
request to other SIP Server (such as ITSPs) after a fixed
time even Authentication error has occurred.
By default, the fixed time = 1800000 milliseconds (30 minutes)
1.4.4.2 Beta (July 20, 2005)
– Fixed the problem that periodical REGISTERs from PBX to
other SIP Server (such as ITSPs) stopped when the Realm
for INVITE is different from the Realm for REGISTER.
– Added a Date header in the Email for a Voicemail Email
notification (Fixed the compatibility issue with SMTP Server).
– WAV sound format of Voicemail file that you can download
from OnDO PBX admintool or that is attached in the Email
notification is changed from PCM to u-law.
1.4.4.0 (Jun 27, 2005)
– Supports qop=”auth,auth-int” for some ITSPs.
– Fixed a bug that OnDO PBX didn’t send ACK when 200 OK from OnDO PBX
and CANCEL from a UA were sent at the same time.
– Fixed a problem that happened only on a 64-bit Solaris.
– Fixed a problem that some sessions were remained after calls for some cases.
– Fixed minor bugs
1.4.3.3 Beta (Jun 7, 2005)
– Fixed a voice cutting issue which happened when using Cisco phones
– Fixed minor bugs
1.4.2.6 Beta (May 2, 2005)
– Supports MWI for the phones which don’t send SUBSCRIBE
– Fixed RFC2833 DTMF problems
– Improved Inband DTMF recognition
– Fixed the problem of no voice after Call Park was executed
when RTP relay =OFF(G.711u only), ON(G.711u only)
1.4.2.1 beta (Apr 14, 2005)
– Supports Message Waiting Indicator (MWI) function for Grandstream phones.
1.4.1.8 beta (Mar 21, 2005)
– Supports to join Conference from Auto Attendant
1.4.1.7 beta (Mar 18, 2005)
– Supports ILBC
– Supports G.711 ALAW
1.4.0.10 Beta (Jan 19, 2005)
– Fixed minor bugs
1.4.0.9 Beta (Jan 18, 2005)
– Added Conference function
– Added Recording function
– Supported other codecs only for peer-to-peer conversation
– REFER (Transfer buttons on SIP phones now work) (experimental)
– Presence (experimental)
– Fixed to include Authorization header in the INVITE request
for the 401 challenge response from third party SIP proxy.
1.3.1.9 (Dec 15, 2004)
– A bug that a display name in FROM header is deleted when a call goes
through Auto Attendant, was fixed.
– A problem that no voice is transmitted after RTP session timeout
when [RTP relay] = on, was fixed.
1.3.1.8 (Nov 22, 2004)
– General Release with fixed minor bugs.
1.3.1.3 Beta (Nov 09, 2004)
– Added a flag whether to retry round robin call forwarding or not.
– More stable than ever.
1.3.1.0 Beta (Sep 24, 2004)
– Added a function to dial to notify “Not at a desk” to PBX.
– Added a function to retrieve a mistakenly dropped call that was on hold.
– Fixed bugs.
1.3.0.7 Beta (Sep 9, 2004)
– Fixed minor bugs
1.3.0.6 Beta (Aug 31, 2004)
– Improved performance of OnDO PBX by adding an option of RTP-non-relay mode.
– Call Queuing
1.2.1.0 (Jul 8, 2004)
– General Release stable version with fixed bugs
1.2.0.1 Beta (May 14, 2004)
– ARS (Automatic Route Selection) feature is added
– Call Pick Up feature is added
– Call Park feature is added
1.1.0.5 Beta (Mar 22, 2004)
– Multiple bug fixes and improvements
1.1.0.4 (Feb 05, 2004)
– Solved Windows 2003 issues
– Minor changes to Installer
1.1.0.3 (Jan 26, 2004)
– Redhat Linux support added
– Voicemail feature, English version added
1.0.1.1 (Dec 15, 2003)
– Email notification feature added
– Multiple bug fixes and improvements
1.0.0.1 (Nov. 3, 2003)
– Call Hold feature
– Call Transfer feature
– No Answer Call forwarding feature
– Unconditional Call Forwarding feature
– Auto Attendant feature
– Direct Inward Dialing feature
– Ring Group feature
– Call Hunting feature
– Voicemail(Japanese) feature