3CX Build History

• Added: Snom centralized phone book generation.
• Added: Support for Vegastream 50 Europa 2 BRI PSTN gateway.
• Added: Snom Phones Firmware Version 7 provisioning templates including retrieving of centralized phone book.
• Added: Default dial plan for all Linksys phones is now set automatically via provisioning templates.
• Fixed: Caller ID of caller in queue not being displayed correctly.
• Fixed: Incorrect missed call display number of missed calls which are forwarded from IVR.
• Fixed: Call to voice mail special menu with exchange integration on not being redirected properly.

• Fixed: Problem in redirecting voice mail menu to Exchange when Exchange Server integration is enabled.
• Fixed: Fax server would consume too much processor time if faxes had been received from particular incompatible fax devices.

• Added: Ability to allow all network users to send out faxes via Microsoft Fax
• Added: Patton PSTN gateway templates for Firmware version 5.1.
• Added: Call pickup uses INVITE/Replaces.
• Added: Notifications when tunnel is disconnected.
• Added: IVR delivers From field with original display name.
• Fixed: incorrect presence of parking orbits.
• Fixed: Call by Name dialog is set to 5 seconds instead of 2 seconds.
• Fixed: Proper handling of empty parking codes.
• Fixed: Outbound proxy now overrides DNS SRV records.
• Fixed: Expiration check in registrar problem.
• Fixed: IVR transfers calls using original SIP ID to form the From header.
• Fixed: Extension status disabled/away overrided each other.
• Fixed: Make call module uses Display Name.

• Improved IVR - Its no longer necessary to specify extension number when you are picking up your voice mail from your extension. It is also possible to listen to own voice mail greeting from the personal voice mail menu.
• Active calls page allows admins to see all active calls in the system and optionally disconnect them.
• Improved backup and restore process which is much faster then previous versions
• Ability to associate DID numbers with VOIP providers
• Ability to trigger backup and restore from the command line, allowing for scheduled backups.
• Greatly improved SIP interoperability
• Windows 2008 support.
• Sip ping feature which can detect calls that have not been terminated properly by the endpoints. (To switch this feature on, add this section to the 3CXPhoneSystem.ini file sipPingPeriod = <interval in seconds>)
• Support for Patton gateways with firmware version 5.1, and support for more country tone sets. (available in next beta)
• Support for Vegastream gateways (available in next beta)

• Call conference service – allows you to create conferences with up to 32 participants (license permitting)
• Intercom – ability to call an extension and force immediate pickup (phone will automatically go to speaker phone). This can be used as intercom at doors, or by managers. Audio will be 2 way. SNOM, Aastra and Linksys phones are supported.
• Paging – ability to setup a ring group that allows one extension to page many extensions at one go and broad cast a message. SNOM, Aastra and Linksys phones are supported.
• Support for BLF provisioning – BLF lights indicating extension status on phones can now be provisioned automatically. SNOM, Aastra, Grandstream and Linksys phones are supported.
• Improved Call Queue performance.
• Call Queueing status - Ability to view all queues, which extensions are logged in as agents, as well as a list of callers waiting in the queue.
• Ability to provision phonebooks to Aastra, Grandstream, Linksys and SNOM phones. All extensions will be listed, as well as the ability to add custom entries
• Ability to record all calls from a particular extension
• Extended HTTP API
• Ability to switch recording on / off per extension
• Ability to disable an extension
• Ability to disable outbound calls for an extension
• Ability to set away/available status

• Added: PBX now plays early media. Early media is used to play  messages such as 'This mobile is not in a position to respond'. Early media can be disabled from the 3cxphonesystem.ini file in the General section, enableEarlyMedia=0.
• Fixed: Wrong state of call shown in line status if outbound rule is assigned to more than 1 line.
• Fixed: PBX didn't work correctly with subnets which mask length is other than 0,8,16,24,32.
• Fixed: Mandatory NOTIFY packet was not sent in case of subscription request.
• Fixed: In ring group, busy detection of members was always overridden by "Use PBX Status".
• Fixed: Calls were stuck when calling a ring group and using busy detection as "Use Phone Status".

• Added: Setup file is now MSI instead of Exe. This will facilitate download and installation of future patches.
• Added: Support for Aastra 5X series phones.
• Added: Support for Linksys phones.
• Added: Support for provisioning Aastra phones (support for linksys to be provided over the next few days via internet updates).
• Added: Improved Presence functionality using SIP dialog-info.
• Added: ECM FAX option is enabled by default, reducing the number of truncated faxes.
• Added: maxNoAnswerTimout in general section of ini file - 180 seconds. Overrides Continue ringing option for extensions.
• Fixed: route for out of Dialog MWI notifications through tunnel.
• Fixed: VOIP line re-registration procedure correctly track line status in case of voip provider inaccessible.
• Fixed: update of voip lines configuration and updating registration status after changing line configuration.
• Fixed: PBX now drops a call if server leg (UAS on PBX) does not receiveACK from remote party (call hung issue).
• Fixed: memory leak in Media Server.
• Fixed: Fax header declarations (fax being caught as virus by antivirus software).
• Fixed: Memory leak in IVR.
• Fixed: Forked ID presence issues. If 1 contact from a forked ID is busy, all extension is market as busy. Same for away status.
• Fixed: Now PBX uses media server SDP if destination of blind(attendant) transfers is bound to Media Server. Old behavior - always use "invite without SDP". This new behavior is controlled by "allowNoSDPIfBoundToMS" ini file option. Default value is 0 (new behavior). To revert to previous behavior set this option as [General] allowNoSDPIfBoundToMS=1.
• Fixed: For ring group. Now RTP mode corresponds to extension settings (PBX delivers audio, support re-INVITEs). Previously proxy/bypass mode was used for all extension even if they are bound to Media server.
• Fixed: Media server doesn't "spam" trace log with "Can't receive RTP packet" message.

• Added: Authentication in tunnel (General settings page,  "Others" section)
• Added: Status of Queue availability is added to presence info
• Added backup and restore for bridges and Tunnel
• Fixed: Lines could get stuck in Digital receptionist because no 'Bye' was sent by DR at time out.
• Fixed: Changed text from no action to end call in Digital Receptionist time out option
• Fixed: refer memory leak fix for transfer to the queue through digital receptionist
• Fixed: Corrected incorrect information being sent in email when a new extension is created
• Fixed: Restore for German / Russian non standard characters.
• Fixed: Caller ID andmultiple outgoing line calling problem with some Patton devices.
• Fixed: Corrected support links in pbx web interface
• Fixed: Removed repeat prompt as an option in the timeout options.

• New installer which allows updates to be installed without complete re-installation
• Added a tunnel, which allows all SIP and RTP traffic to be tunneled via a single, configurable TCP port (by default 5090) Currently this tunnel can be used for bridges between phone systems and by hard phones on remote networks (which have to use the tunnel as an outbound proxy). The tunnel will also be included in the next version of the VOIP client, due out soon.
• Fixed: Improved backup and restore procedure. (see below)
• Fixed: Bug where a call could potentially remain stuck in the system and be displayed as active in the interface, even though the line would have been disconnected
• Fixed: Improved the patton 4554 template
• Fixed: Caller ID is now correctly passed to Patton devices
• KNOWN ISSUE: Affects calls via tunnel only: Call Transfer from a phone behind a tunnel, back through the tunnel will not work

• If you are using a VOIP provider, you router must be configured with STATIC PORT MAPPING for 5060. Incorrectly configured routers that are doing port translation rather then port fowarding are a cause of failing inbound calls, one way audio and so on. To check whether your router is doing port address translation run the firewall checker.
• We have created a correct sample configuration for a popular Linksys router at http://www.3cx.com/support/linksys-configuration.html
• If the PBX machine has multiple interfaces, and the fax service is being used, you must specify the IP in the 3cxphonesystem.ini file for the fax service. See this FAQ: http://www.3cx.com/support/fax-multipleinterfaces.html

• Please see detailed listing of phones, gateways and firmware used in our tests at http://www.3cx.com/support/testedphones.html
• Grandstream GXW4104 gateway - must be switched to PBX delivers audio for forwarding of calls to outbound numbers to work. This is an issue relating to Grandstream. This is not compatible with the fax feature unfortunately.

• C:\Program Files\3CX PhoneSystem\Data\Http\backup.php
• C:\Program Files\3CX PhoneSystem\Data\Http\support.php
• C:\Program Files\3CX PhoneSystem\Data\Http\functions\BackupParser.php
• C:\Program Files\3CX PhoneSystem\Data\Http\functions\BackupManager.php

• Music on hold when transferring from Digital receptionist.
• Ability to bypass STUN server resolution by removing stun server entries from general settings page.
• By default, 3CX will use both auth ID and external line number to identify source of call from a voip provider.
• By default, 3CX will use both LineID and Gateway host to indentify source of call from a PSTN gateway.
• By default, port will be set to :5060 when comparing host/port fields in source identification rules.
• Complete generation of Grandstream phones provisioning configuration without the need to use the GrandStream tool.
• Added templates for the following gateways: Patton SN-4112 (2-port Analog), Patton SN-4552 (1-port BRI), Patton SN-4960/E1 (1-port PRI E1), Patton SN-4960/T1 (1-port PRI T1)

• Firewall checker releases ports after use.
• Now it is possible to check multiple source identification rules, previously only the first one was checked.
• Removed "Route calls for this Bridge during office hours to" table as there was no use for it.

• Fixed: Multiple outbound calls over a single VoIP Provider account now works
• Improved handling of recognition of local devices and external devices
• Improved log messages - more complete information is now presented to help with creating source identification rules and inbound SIP Header field maps
• Improved caching engine
• Removal of OpenVPN components in preparation for new proxy + tunnelling protocol to ease NAT traversal.

• Fixed which causes systems installed in a DMZ or on a Public IP to not work correctly
• Fixed several issues VOIP providers
• Improved licensing information display
• Added a dialog to ask for FQDN of server, to allow for use of FQDN name of server in phone configuration
• Improved feedback of firewall checker
• Fix a bug where by rejected calls would work against license limit.

• Ability to create outbound rules / dial plans based on number of digits. This allows a dial plan to be setup that does not require a prefix.
• Extensions no longer need to be setup as internal or external - the PBX will recognize this automatically, providing full mobility to user.
• SIP ID forking allows multiple SIP phones to have same extension number and ring at the same time, allowing a user to have both a desk phone and use a software phone whilst on the road or at home.
• Ability to specify up to 3 outbound routes per rule - allowing you to easily configure back up / fail over routes.
• Ability to specify bank holidays in out off office hours section. This way, calls can be handled differently during bank holidays.
• Improved automatic configuration of Patton gateways, including the ability to automatically set the country tone set.
• Improved support for Audiocodes gateways.
• Overall performance has been increased drastically to support more simultaneous calls, users and call queues.
• Firewall test utility - allows automatic testing of the firewall configuration, and reports which ports still need to be opened in order to allow a VOIP provider to be used.
• Ability to make calls using just the SIP ID of the person you wish to call.
• Update console shows all available updates for 3CX Phone System, including software version updates, VOIP Provider and Gateway template updates and translation and system prompts updates.
• Improved system prompts and music on hold recordings.
• Extensions with 2 digits.
• DID routes can be given a name that will appear as the caller ID name.
• Improved handling of multiple network interfaces.
• Media server allows media pass thru, resulting in improved voice quality (e.g. with Grandstream and other gateways).

• Extension users can configure their forwarding options from within the 3CX VOIP (redirect to another extension on busy, to mobile etc).
• Ability to connect 3CX Phone Systems using a Bridge.
• Ability to change Voice mail PIN from the 3CX VOIP client.
• G729 support - 4 calls for Small Business, 8 for Pro and 16 for Enterprise
• Call Parking.
• T38 fax functionality - receive faxes as PDF files and route them to an email address. Fax feature works in combination with support gateways such as Patton, Audiocodes and Grandstream.
• Provisioning for Snom320 and Snom360 SIP Phones
• Call by Name (available via Digital receptionist).
• Call Recording (currently requires a SNOM phone).
• Added BLF capability for SNOM and Grandstream phones.
• Added a user portal to allow users to change their extension options

• Improved Vista support - Microsoft Windows Vista is now fully supported
• System can now be configured to listen on specific interfaces / address (internal or external) in the system's INI file
• Improved download mechanism for updating of languages, prompts, and templates
• Interface now also available in the following languages: Italian, German, Spanish, Greek, Danish
• Manual now also available in the following languages: Italian, German, Spanish, French
• Fixed signaling to handle header translation carried out by Cisco NAT devices
• Added configuration templates for PATTON / INALP gateways (ISDN BRI)
• Several bug fixes

• Reworked the Management Console to provide more information.
• Added Direct SIP Calling
• Added MWI (SB, PRO, ENT versions)
• Added Call Queues (ENT version)
• Added Call Pickup
• Reworked the Gateways/Providers templates system
• Introduced support for auto-generation of device configuration files
• Certified for Windows 2003 Server
• Backup outbound rule - in advanced if line is busy or not responding, use
• another voip gateway or voip provider
• Backup STUN server entry
• Ability to specify authentication details for an SMTP server
• SIP ID support
• Generates Patton configuration file
• Includes templatest for popular providers
• Includes templatest for popular gateways
• Revamped interface
• Internationalization of the interface
• Ability to download system prompts of other languages
• Added for support for Audiocodes MP 114, Linksys 3102
• Addded Exchange 2007 support (Enterprise edition only)
• Support 40 ms and 10 ms voice packets

• Improved DTMF detection which caused beeps on the line in some installations

• Added pre configured templates for popular VOIP providers and Gateways
• Implemented possibility to limit number of concurrent calls for a VOIP account (bandwidth management)
• Improved device registration - now explicitly checked
• Added support of out-of-dialog (without Contact header) provisional messages
• Moved settings from registry to ini file
• Upgradeable to Small Business / Pro, which adds outbound calling & Call Transfer features to the Call Assistant
• Added possibility to upload templates
• Added activation/licensing/upgrade
• Licensing support in Call Assistant
• Added list of IPs for source recognition (Use IP in 'Contact)
• First implementation of MS Exchange 2007 integration (requires Pro license)
• System parameter changes take effect on-the-fly
• Improved email notification functionality
• Digital Receptionist menu changes take effect even in-call
• Improved Call Assistant functionality, data retrieval, and error-handling
• Added full support of UNICODE to Call Assistant
• Call Assistant now allows fast user switching (can launch one instance per user)
• Improved error handling and connection restoring for Call Assistant
• Log entries for DTMF recognition/methodology
• Handling of non-sequential ports for audio (RTP/RTCP)
• PBX will now attempt to handle calls received from mis-configured sources
• FIX: Registration removal after extension is deleted
• FIX: Unregister extensions on change in credentials
• FIX: Disconnected endpoints will have correct line status displayed
• FIX: Use of most recent registration contact is implemented
• FIX: Fixed one-way audio when transfer target doesn't support 'replaces'header
• FIX: Log messages text improved
• FIX: Voicemail temporary files are stored to the 'Data\Ivr\Temp\ivr' folder
• FIX: Improved GSM-codec handling
• FIX: Media Server log entries description improved

• First implementation of support for external phone and gateway devices
• Improved logging to show media stream parameters when call legs are created
• Improved gateway/provider template handling including import facility
• Improved support for extensions/providers/gateways by adding some advanced options
• Improved audio prompts handling by IVR system
• Resolved minor bug with adding DID lines

• Better handling of custom prompts in Backup/Restore
• Advanced Options / Settings for VoIP Providers and PSTN-to-VoIP Gateways to better handle a wider range of providers and gateways
• Music-On-Hold now customisable
• Possibility to choose length of extension numbers during setup
• First introduction of templates mechanism to simplify VoIP Providers and PSTN-to-VoIP Gateways setup and configuration
• Possibility to trigger registration of VoIP provider from Interface
• Introduced the possibility for IVR to play back Caller ID and Date/Time of messages saved
• Introduced support of international characters
• Introduced VoiceMailBox as additional destination for incoming calls and as fallback for group calls
• Improved handling of DTMF detection and re-delivery (introduced SIP INFO support)
• First implementation of Outbound CallerID
• Improvements to Registration/Authentication mechanisms
• Improvements to VoIP Providers Support
• Improvements to Call Transfer handling mechanisms
• Improvements to Logging Mechanism
• Introduced the means to adjust logging levels from interface
• Improved Audio Prompts
• Interface Cleanup
• Introduced "Forward to Outside Number" functionality and transfers to outside numbers
• Improved handling of mp3 files for audio prompts

• Implemented DID rules.
• Added Call Assistant.
• Introduced the Held and On-Hold statuses.
• Now using STUN-resolved external IP for VoIP registrations.
• Implemented Busy detection on server.
• Improved identification of incoming calls from VoIP providers.
• A lot of improvements in transfer (blind and attended).
• Fallback to previous call on unsuccessful transfer is implemented.
• Implemented Transfer feedback from Digital Receptionist.
• Added support of 'RemotePartyID' for DID detection.
• FIX: Non-outbound VoIP lines now register.
• FIX: Improved addressing of VoiceMail while forwarding call through several points.
• FIX: 'Forward All Calls' state after importing extensions is now correct.
• FIX: VoIP lines registration bug with is fixed.

• Rewriting of Sip/PBX server
• Added new Mediaserver
• Added new IVR system
• Discontinuation of use of sipX Mediaserver code.
• Added support for the GSM codec.
• Added support for Grandstream gateways
• Added support for Vegastream gateways
• Status Monitor has been improved.
• Outbound Rules have been simplified.
• Better handling of upgrade during re-installation.
• Introduced forwarding of all calls option
• Redirection options now available when extension is busy, unregistered or no answer
• Calls can be forwarded to external numbers
• Digital Receptionist can now execute a specific action on timeout.
• Caller can enter extension number in any Digital Receptionist menu
• Call report graph contains link to a full sized image of the produced graph.
• System prompt have new additions and their descriptions have been improved.

• DTMF doesn't work when call is using GSM Codec. This is a limitation of the codec.
• Circular Forwarding will cause the PBX Server to terminate
• Deleting an extension which has a line forwarding to it will cause the line to be deleted
• Certain combinations of actions involving putting/retrieving calls on hold with transfers behave unpredictably. Mainly related to different handling of SIP transactions by different devices.

• FIX: Browser compatability issues. Now working with Opera9, FireFox, Firefox2, IE7, IE6 and older
• FIX: Database connectivity issues when using IPv6
• FIX: IVR bug fixes
• FIX: Default stun server was incorrect causing problems with VOIP providers that require a stunserver.

• Introduced Auto backup during re-installation.
• FIX: Fixed bug which emerged in last public build where calls were being terminated abruptly
• Moved binary files and logs to paths that are more readily accessible

• Added 'Reset Log's button as a troubleshooting Aide.
• Improved installation's error handling.
• Status line shows dialed number as opposed to line number.
• Implemented Call terminsation on calls with lengthy silence.
• Added call transfer handling for DLink & Micronet Type Gateways.
• Can now connect PhoneSystem to internal VoIP provider (e.g. as an Asterisks extensions)
• SDP conversion error FIX.
• Added transfer and hold support for Eyebeam and X-lite

• Voice mail only supports DTMF in RTP, not in band DTMF. Most phones use in band DTMF and the voice mail, IVR system wont recognise this. We are working to deliver this support asap.
• In some exceptional cases, the PHP & PostgresSQL services wont cooperate well and you will not be able to login to the configuration. If this occurs, please contact support and we will send you a special debug file which will allow us to resolve the issue.
• 3CX and Clipcom gateways are not compatible stable at this point. We are looking into this problem and attempting to determine if this is a 3CX or a Clipcomm issue
• D-Link / Micronet Gateways will sometimes show the lines as unregistered, even though they work as normal.

• Added Digital Receptionist Known issue: A pause of 6-9 seconds occurs when DTMF is entered (by some key pressing) while prompt playing, and before the dialog processing continues. This occurs with files over 250Kb and on certain machines only.
• Added VM (Voice Mail).
• Added DTMF (RFC2833) Support for Media Server.
• Media Server codec support via plugins.
• Server Status displays latest log entries first (at top).
• Customizable Voice prompts.
• Added support for transfer of calls from PSTN gateways.
• Backup and Restore function bug fixes.
• Added Support for VoIP providers with proxy servers.
• Allows VoIP Provider servernames with "-" in FQDN.
• Allows emtpy STUN server field in VoIP Provider definition (machines with interfaces with public IPs).
• Caller ID bug fixes.
• External lines now have own status monitor events and icons.
• Interface messages have been collected in central messages file.
• Added extra SIP Authentication support.

• Incomimg lines can now properly forward calls to ring groups.
• Logging messages are now more 'user friendly', and more understandable.
• Logging of registration failures (attempts) is now enabled by default.
• Update of troubleshooter file format. File format is now .zip instead of .bz.
• Known issue: Previous .bz backups cannot be imported. User must first extract the backup file from old .bz archive and rename the extracted file to have a .xml extention. User must then compress the xml into a .zip archive prior to attempting a restore in 'General settings' page.
• Known issue: VoIP providers that require clients to submit the Server's IP in the CONTACT field of SIP REGISTER requests will not work as incomming lines since the Provider will not know where to route incoming calls.

• Can receive inbound calls on VoIP Provider lines.
• Can create a backup of the 3CX PhoneSystem database, can also restore the backed up database file.
• "Generate support info" function creates a troubleshooter file that can contains 3cx PhneSystem Setup information that can be sent for reviewal by support.
• RTP port ranges used by the 3CX PhoneSystem Media Server for internal calls and calls placed through VoIP providers can be configured to operate on custom ranges.
• STUN Client for use of PBX server behind NAT.
• Various code changes and bug fixes.

• First release of 3CX PhoneSystem

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