System message

Version History

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.

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.

2.3.4.8 (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