The VoIP Connection Asterisk Administrator Version History

The VoIP Connection Asterisk Administrator Version History:

Version 4.5 5/30/07

Minor fixes to the web interface.
Minor fixes and cleanup to the web interface.
Fixed some Sangoma driver issues.
Fixed the backup script so that it will not backup voicemail that is stored on secondary storage.
Fixed the parsing of the card destination for the Modify Zaptel Device command.

Version 4.4 5/20/07

Updated Asterisk and all related packages to the latest versions.
Updated Sangoma drivers to the latest version.
Removed the underline style from all menu items in the web interface.
Added XML::Simple Perl module.
Updated spandsp to version 0.0.3. Also updated app_rxfax and app_txfax to the latest versions.
Added interface to control the location for recorded calls just like CDR data and voicemail.

Version 4.3 5/7/07

Updated the kernel to 2.6.20.6. Also built additional modules and drivers.
Fixed the backup scripts so that each one can only run once to avoid any potential data corruption.
Fixed backup scripts to only backup voicemail that's stored on the ram disk.
Changed the default digit map for Polycom phones to "x.T".
Fixed the path handling for the Asterisk *.conf files when running the different versions.
Added a new error messge to the main web interface if the Asterisk version number is incorrect. This only happens when upgrading from an earlier version.
Fixed the display of the "UNSAVED DATA" message.
Added support for the Digium TDM800P card.
Added support for Sangoma cards with FXO/FXS configuration.
Added support for Redfone foneBRIDGE2 with FXO/FXS configuration.
Fixed route commands so that they can handle old dial plans.
The catchall extensions now go to congestion instead of sending the call back to the incoming context.
Upgraded webmin to version 1.340.
Added the shorewall firewall package.
Added the Asterisk Recording Interface package.

Version 4.2 4/9/07

Switched back to MP3 hold music as default since the WAV hold music seems to be problematic.
Added the php-imap and php-gd packages.
Added Asterisk-Stat web interface for CDR reporting.
Added link to Asterisk-Stat on the Reports menu of the web interface.
Upgraded Nagios package to version 2.8 and Nagios Plugins to version 1.4.7.

Version 4.1 3/28/07

Added mac address output to rebootpolycom script.
Added code to remove all non-numeric characters from the CALLYOU/CALLME number.
Added new commands CALLYOU0 and CALLME0 that ask for both the channel and number to connect.
Fixed minor issues in default dial plan.
Added new Rhino zaptel drivers.
Added support for Fractel voice service.

Version 4.0 3/16/07

Switched from Fedora Core 3 to CentOS 4.4 as the base operating system.
Upgraded the Linux kernel to version 2.6.20.
Added php5 and mysql5 server packages.
Mysql is now running by default and Asterisk cdr data is automatically sent to the asteriskcdrdb database.
Added Sugar CRM package.
Added a script to Sugar CRM to allow calling a phone number by clicking on it.
Added documentation on how to configure Sugar CRM.
Added enhanced PickupChan package by Thorsten Knabe. This package adds the Asterisk applications PickUp2, PickDown2 and Steal2.
Upgraded Asterisk and supporting code to the latest versions.
Password protected the Flash Operator Panel.
Added a link to the Administrator API Reference on the help page.
Added the pbxadminservice.cgi script. This script returns results in a minimal HTML format instead of full HTML format.
Changed the call forwarding extensions (*42, **42, *43, **43) to use the channel information to determine the extension instead of using the callerid.
Added command to delete recorded call files.
Added webmin and pam-devel packages.
Added Authen-PAM-0.16 and Net_SSLeay-1.30 perl modules.
Added menus for 411 and 911 call configuration.
Added extension 2000 to the dial plan. This extension asks for a number to call and your callback number and then places the call for you.
Added ChanSpy extension 2222 to the dial plan.
Added WEB-MeetMe Control interface.
Added command line script rebootgs to reboot Grandstream phones remotely.
Added command line script rebootpolycom to reboot Polycom phones remotely.
Added support for Pipeline SIP1 and SIP2 services.
Added support for Junction Networks IAX2 service.
Added support for Teliax IAX2 service.
Added over 30 existing commands to the API and added 3 new voicemail commands.
Marked the conferences context as an invalid context for adding regular phone extensions.
Fixed the automatic numbering of ports and spans for the Add Zaptel Device command.
Added option 0 to the Add/Modify IVR Menu commands.
Fixed the handling of extension 0 in all Add/Modify commands.
At the end of the boot cycle, Asterisk, Zaptel and Wanrouter are now restarted to make sure that all configuration changes are active.
Added the Upgrade Dial Plan command to automatically fix incompatibilities when upgrading.
Added a password to all of the recording extensions to prevent accidental re-recording of the ivr menus.
Changed the default number of days to keep voicemail messages to 7.
Enabled the FTP service by default to make it easier to configure Polycom phones. The only valid login for ftp is vsuser.
Added the following packages: perl-libwww-perl, arptables_jf, perl-Crypt-SSLeay, perl-HTML-Parser, perl-Net-DNS, telnet, perl-Digest-HMAC, perl-Convert-ASN1, perl-XML-SAX, perl-XML-Parser, perl-HTML-Tagset, perl-Digest-SHA1, perl-XML-NamespaceSupport, utlramonkey.
Drives with Linux file systems will now also be searched for voicemail or cdr data storage during boot.
Added new nightring routing to the default dial plan.
Added new time based routing examples to the default dial plan.
Added new command to unlock voicemail boxes.
Added new script to load test Asterisk.
Developed a new high availability and scalable cluster version.
Added OpenSER packages to the cluster version.
Changed the system backup file format to a compressed file format.
Fixed a parsing error when getting the outgoing line number for the Modify Service command.
Updated the Voicepulse service to the new settings (connect01.voicepulse.com).
Added documentation on how to configure AudioCodes FXO/FXS gateways.
Added Flite text to speech package.
Added mdadm and apcupsd packages.
Fixed a problem with the CALLYOU links. They would not call the local extension correctly.
Added MySQL menu and MySQL Query command to run generic query on the local databases.
Added the fax extension to the incoming context in addition to the default context in extensions.conf.
Added a Hangup command to the end of group extensions to make sure that calls don't drop through.
Updated the included Polycom firmware to version 2.1.0.
Updated the Polycom provisioning templates to support presence and buddy watch.
Added web link to the data directory for easier access to the data.tgz backup file.
Added support for 4x20 LCD display in addition to the existing 2x16 LCD support.
Fixed a minor problem in the Asterisk service script.
Added lm_sensors and SNMP packages.
When adding zaptel devices, added a check to make sure the group number is in the valid range 0-31.
Added commands to Add/Modify global IVR menu options.
Moved blind transfer from # to #1 to avoid conflict with external IVR menus.
Added "Unavailable Timeout" setting to the Set Dialing Options command.
When adding an extension, a voicemail and paging extension are now automatically added to the system as well. The voicemail extensions is the extension number + '*' and the paging extension is the extension number + '#' (exmaple: 601, 601*, 601#).
Changed the Vonage server name to the new high redundancy server var1.voncp.com.
Removed Zap from the Voice Service menu. When adding a Zaptel card to the system, the outbound calling is now configured from the same menu.
Enabled automatic log rotation for the PBX log files.
Added vconfig package to support VLAN configuration.
Added Rhino R1T1 and R4FXO drivers and configuration support.
Updated Flash Operator Panel to version 0.27.
Added WakeMe application. Added two wakeup call methods and extensions 771 and 772 to configure a wakeup call. Extension 771 uses a php script and extension 772 uses app_wakeme.
Added web interface to create a wakeup call.
Added the capability to set ring timeout for each individual extension.
Added outgoing call access control on a per extension basis.
Added multi-tenant support to the web interface and dial plan.
Optimized the dial plan through increased use of variables.
Added a red, blinking notification to the web interface to notify the user that changes need to be saved to disk.
Added Sigma Digital Voice as a supported voice service.
Added Zap extension support to Add Group Extension and Add Multiple Extensions commands.
Changed the dhcpd setting ddns-update-style from none to ad-hoc.
The script that backs up all settings now runs at the lowest priority to make sure that it has minimal impact on audio quality.
Modified the configuration of Zaptel devices and the Add Extension command so that the callerid information is now set for each Zap channel.
Added menu commands to modify all of the call routes.
The web interface now automatically makes redundant backups of the critical configuration files.
Optimized the content and usage of the RAM disk.
Added a configuraton option to automatically remove recorded call files after the specified number of days.
Added configuration support for the Xorcom Astribank gateways.
Added screen and utempter packages.
Added balance, ploticus, iftop and ttyload executables.
Modified the Add Multiple Extensions commmand to use a list of extensions instead of a start and end number.
Added new command to resequence the messages in all voicemail boxes. When an extension dials itself to get to its voicemail, both the channel and the callerid will now be checked for a match.
Added automatic refresh capability to all web interface commands by adding metarefresh=5 as a parameter.
Added ngrep, ntop and iptraf programs.
Added Nagios package.

Version 3.3 3/14/06

Added documentation on how to update SNOM phones with TFTP.
Added the Cut application to the list of incompatible commands.
Fixed the handling of the gateway when working with the system settings.
Fixed the handling of Sangoma cards with hardware echo cancel. Also fixed the config files to make sure that the d-channel is handled in hardware.
Fixed the default span timing values when adding a zaptel card.
The tftp server output is now more verbose in the log files.
Increased the size of the /tmp ramdisk.
Moved all administrator configuration files to the /etc/administrator directory.
Fixed the handling of file system errors after a hard reboot. The system should no longer go into an endless fsck and reboot loop.
Fixed incorrect results in two of the tests in the testsystem script.
Fixed problem with CALLYOU command.
Updated Asterisk, Libpri, Zaptel to the latest versions.
Added additional check to make sure the wanrouter service is (re)started only when enabled and configured.
Added restartservices command line script.
Fixed the html frames for the client web interface.
Fixed parsing error for call data reports when a callerid had a "+" in it.
Changed the feature timeout values to make them easier to use.
Added command to add messages to the system log.
Changed the locations of the stable and development versions of Asterisk.
Added autoconf, automake and m4 packages.
Changed the web interface to be compatible with IE7 Beta 2.

Version 3.2 2/18/06

Updated the Customize Web Interface command to also include the command descriptions. The command now also displays the data in a table format.
Fixed the Delete Context command to also delete the context from voicemail.conf. Also fixed the list of contexts to delete so that all contexts show up no matter what.
Fixed an issue when configuring a Digium TDM400P card.
Fixed the function that gets the current list of used zap channels. It now also counts the d-channels.
Upgraded the Linux kernel to version 2.6.15.3. We are now using a custom built kernel.
Updated Asterisk, Zaptel, Libpri, Wanpipe, Spandsp and RxFax/TxFax to the latest versions.
Fixed the call forwarding command. The list of extensions is now complete and sorted. Also the inputs are now being verified to make sure they are all numeric.
Enabled the courtesytone setting in features.conf to avoid the bug where you can't retrieve a parked call.
Set crypto setting in Asterisk Makefile to make sure that Voicepulse accounts work.
Fixed the use of a deprecated function in the default dial plan.
Improved the checking reliability in the checkasterisk script.
Fixed a bug when handling dtmf settings in Add/Modify Extension Table commands.
Fixed the provisioning template selection for Add/Modify Extension Table commands.

Version 3.1 2/1/06

Added CALLYOU links to complement the CALLME links. The CALLME links will call the local extension first and then the supplied phone number, whereas the CALLYOU links will call the number first and then the local extension.
Fixed the web link for SNOM provisioning.
Fixed a problem when running "Save All Changes To Disk" which caused the backup to fail.
Removed all incompatible applications from the default dial plan and added a check to the web interface to show any incompatible applications (along with the new applications that replace the old ones) that are in the current dial plan. This is helpful when upgrading from earlier versions.
Changed the default DTMF mode to RFC2833 for all devices.
Enabled the resizing of the web interface frames.
Added templates to all supported devices for provisioning.
The provisioning code has been rewritten and improved.
Added new provisioning menu and moved all provisioning commands from the phone extensions menu to the new menu.
Added new command to provision multiple devices at once in a table format.
Updated the DHCP server configuration. Added support for options 66 and 67 and also the boot file name. Also added check boxes to control which options will be used.
Updated the backup script to also back up the network settings. This will make it easier to upgrade to a new system flash in the future.
Also added /etc/wanpipe and /etc/*.conf to the backup script.
Added a command on the Asterisk menu to view the recorded call files in the Asterisk monitor directory.
Updated the default extension macro so that when an extension calls itself it will be dropped directly into its own voicemail box without having to enter a password.

Version 3.0 1/4/06

The zaptel commands now try to fill in the menu with reasonable default information.
Removed pwlib and openh323 in favor of ooh323c in asterisk-add-ons.
Restructured the html directories. The top level web directory is no longer password protected. All important files that need to be protected have been moved to subdirectories with password protection.
The user mode of the Administrator web interface has been removed. The custom mode should now be used instead.
Added a check for spaces in file names when uploading files.
Reduced the permissions of the vsuser account.
Added new web only vsoperator account.
Renamed the API commands CREATESVC to ADDSVC3 and MODIFYSVC to MODSVC3.
Fixed the Set Time Zone Command to show all time zones.
Added the Time::HiRes Perl module required by the modified vmail.cgi from Digium.
Fixed the default dial plan so it's possible to call numbers that don't send a connect, such as 800-843-3000.
Fixed the parsing of the queue log file for reporting. Reduced the amount of memory needed to process the file.
Fixed the handling of extensions that contain the "*" character.
Upgraded the Linux kernel to version 2.6.12-1.1381_FC3smp. This new kernel supports SMP and IO-APIC.
Added a check to warn the user if the /tmp directory is full, since this causes problems.
Added new command "Clean Temp Directory" to the System menu.
Fixed the parsing of the CDR data file for reporting. Reduced the amount of memory needed to process the file.
Prevented the web browser from timing out when processing for a long time.
Updated zaptel, libpri and asterisk to the latest versions (1.2 and SVN TRUNK).
The Add/Mod Extension commands did not handle the port number for Zap extensions. This has been fixed by adding the zap ports to the protocol list.
Added documentation on how to create a new web server certificate.
Added dial plan and support scripts to deliver voicemail messages by phone to one or more phone numbers. This needs to be manually configured. See ouboundmsg context in extensions.conf.
Added the lynx package.
Added subversion package and all necessary support packages.
When editing a configuration file the current file name is now displayed above the edit area.
Added a very basic file manager to the Configuration Files menu.
Added command to set dialing options such as call recording and transfer.
Modified the ramdisk contents. The way the ramdisk is initialized at boot time has also been improved.
Added configuration menu for "callme" url links. This allows the user to create a web page from which someone can initiate a phone call to the user's extension without having to publish the actual extension number.
Added the current macro context to the Voicemail app in the extension macros of the default dial plan.
Updated the Polycom phone provisioning. It is now template based and includes the 1.6.3 firmware.
Added support to boot from USB drives.

Version 2.2 11/4/05

Added a new configuration file (c:\etc/asteriskadmin.conf) to set the color scheme of the administrator web interface. Also added a new Preferences command to the interface to set the colors.
Changed the boot sequence to automatically find the system disk.
Added support for Sangoma PRI cards.
Fixed some minor pattern matching problems when modifying the call forwarding settings for an extension.
Added provisioning for Polycom and Cisco SIP phones.
Modified the backup script so it only backs up files that have changed instead of entire directories as before.
Modified the CPU/Model identification to be more reliable.
Changed the web interface so that the main frame updates when a new sub menu is selected.
Added a menu to set a dialing prefix for each of the 25 lines.
Added LCD support. During boot and while running, status messages are displayed on the LCD if present.
Fixed a potential routing/security issue in the default sip.conf file.
Added Show All Channels command to the Asterisk menu.
Added Show Active SIP Users And Peers command to the Asterisk menu.
Added zip package.
Added command to add multiple voicemail boxes at once.
Added command to delete log files to the web interface.
Added command to delete music on hold files.
Added call queue reports to the Report menu.
The text edit area is now wider (90 columns) and wrap is turned off to make it easier to edit files.
Removed the setting of the SIP IP addresses when initially configuring the static IP.
Fixed support for TDMoE FXO/FXS and added support for TDMoE PRI.
Added support for the Redfone foneBRIDGE quad T1 box.
Added getfromcvs script in /usr/src directory to make it easier to download new Asterisk source code.
Updated zaptel, libpri and asterisk to the latest versions.
Added Add/Modify/Delete Paging Extension commands.
Changed the IVR menus to repeat only 3 times before hanging up the call.
Updated the default dialplan to be compatible with the changes within Asterisk as of version 1.2 beta1.
Added the canreinvite setting to the Modify Extension menu.
Added new custom version of the administrator interface. The administrator can select which commands will be available in the new custom interface. The custom configuration is stored in c:\etc/admincustom.conf.
Added Vovida STUN server and client (/usr/local/bin/stunserver and /usr/local/bin/stunclient). These were added with no support or documentation.
Added a new default route to dialplan IVR menus to handle unknown inbound DIDs.

Version 2.1 9/12/05

Added new Grandstream templates for provisioning.
Updated provisioning for new Grandstream devices.
Updated provisioning for SNOM phones.
Updated documentation on how to use intercom with SNOM and other SIP phones.
Fixed problem with qualify not being set correctly when adding a new extension.
Updated the testsystem script to use 99.98% as the minimum value as suggested by Digium.
Fixed the check for existing extensions so that it takes contexts into account.
Changed all extension commands to handle an unlimited number of actions per extension.
Added a check when modifying extensions to make sure that the protocol is correct in the extension actions.
Added cvs and tcsh packages.
Added new Voicemail menu with ADD/MOD/DEL commands. Also moved the Delete Voicemail Messages command to this new menu.
Added new Upload Music On Hold command to the Asterisk menu. This could already be done with the generic upload file command.
Expanded the preconfigured sub menus from 5 to 9.
Expanded the preconfigured conference rooms from 5 to 10.
Moved attended transfer from * to *2 to avoid conflict with disconnect.
Added a check to make sure Asterisk is running before sending manager commands.
Fixed a parsing issue when getting a list of channels for the CDR reports.
Fixed a parsing issue when modifying a group extension.
Added new commands to Add/Modify/Delete IVR menus and their options.
Modified the incoming context. It does not have an IVR menu anymore. It is now only used for routing calls to the appropriate destinations.
Renamed all sub menus to IVR menus.
Added call route by time command.
Added outbound call route command.
Added a new monitoring script that makes sure Asterisk is working correctly.
If Asterisk fails to respond the script will restart Asterisk automatically.
Changed the Initial Context value for the Add/Modify Voice Service commands to Initial Destination. This allows for more flexibility in routing the calls for the voice service.
Changed the initial web page that requires the selection of a static IP address to also require the selection of a FQDN for the server.
Added support for Pipeline Voice Service.
Moved the call routing commands to the Dial Plan menu.
Upgraded the Linux kernel to version 2.6.12-1.1376.
Added packages ncurses-devel, patch and rpmbuild.
Fixed the SIP registration timeout problem in the Asterisk source code.

Version 2.00 8/4/05

Changed the OS from Fedora Core 1 to Fedora Core 3.

Version 1.20 8/4/05

Documented the default security code for the Flash Operator Panel in the online help.
Added pwlib and openh323 libraries to provide OH323 support in Asterisk.
Modified cleanup script so that the CDR file is not deleted automatically.
Modified the backup script so that /var/log/messages is saved during system restart.
Added new links for Firefly and AstTapi to the documentation.
Added new Standard CDR reports.
Added new commands to backup and restore system configuration. This is useful when upgrading to make sure that nothing gets lost.
The system configuration file is now automatically backed up if a backup does not already exist.
Increased the font size of the statement that asks the user to read all of the online documentation first.
Fixed minor bug in CDR parsing.
Added new command to configure sshd root login and port.
Modified the cgi code that brings up the java ssh client to get the port and user name from the current system configuration.
Changed the default context for zaptel devices to "incoming".
Added new commands to the interface to test the zaptel device and to check the zaptel configuration.
Added new commands to the interface to set the time zone of the system.
Fixed Execute System Command so that it returns the output of the command that was executed.
Added packages traceroute and symlinks.
Added new Create Advanced Route command to the interface.
Changed the default number of days to keep voicemail from 2 to 4.
Changed the arguments to kudzu so that it will not prompt anymore for hardware configuration during boot.
Updated the preconfigured hardware database for kudzu.
Added the rpm database to make it easier to add new packages.
Fixed the Add Service command so that it does not add duplicate extensions.
Removed local copies of SNOM firmware and provided a link to the SNOM download page instead. The SNOM firmware changes too frequently to keep current.
Added download links for SNOM and Grandstream firmware to the main help page.
Added new command to download firmware from SNOM and Grandstream directly to the server.
Added new command to delete provisioning files from the server.
Added unzip and speex packages.
Added a script to test the system (/usr/local/bin/testsystem). Also added the Test System command to the System menu.
Added dial out delay setting for each of the 25 lines to the dial plan. Also added a new command Set Dial Out Delay to set this value for all lines.
Added a Reload Asterisk command to the menus. This command reloads all reloadable modules and updates changes to the configuration without killing active calls.
Added Queue support to the interface. New commands are Add Queue, Modify Queue and Delete Queue.
Added E&M and E&M Wink support to the interface when adding Zaptel cards.
Added command to set the outgoing callerid in the dial plan.
Added documentation for the administrator API.
Changed the outgoing macro in extensions.conf. The Trunk and Number are now split into two arguments.
Added documentation about groups to the add zap service command.
Changed the Restart Zaptel and Stop Zaptel commands to also restart and stop Asterisk.
Moved the "Please read the documentation" statement up on the main page and made it blink since everyone chooses to ignore it.
Removed the references to WinSCP and IPSwitchBoard from the Upload/Download page since they are now listed on the help page.
Added server side file and directory browsers to the upload/download files page.
Modified the main web page. The first line is now a link to the online documentation.
Added documentation for vmail.cgi.
Modified the recording extensions 32950* to play back the recorded message before hanging up.
Added G729 and G723 codecs that can be loaded for testing purposes only.
Updated the CGI with new CSS popup menus that work on all web browsers.
Added over 400 new sound files recorded by Allison Smith.
Replaced the application Wait(1) with Background(silence/1) in the default dial plan.
Added new command to play the Asterisk GSM sound files to the Asterisk menu.
Removed the JPlayer download from the system since it is no longer necessary.
Fixed the search for Linux FS hard drives during boot to avoid conflicts with /etc/fstab.
The kudzu service is now off by default since the system is preconfigured.
The kudzu service can be run from the command line or it can be turned back on.
The Flash Operator Panel is now turned off by default. Since it needs to be configured before it is usable this makes more sense. Added new commands to enable/disable FOP.
Added new command to view system and Asterisk log files under the System menu.
All commands that make changes to the Asterisk configuration files will now automatically reload Asterisk at the end of the command.
Split the billing increments into inbound and outbound. Updated all of the reports to handle inbound and outbound calls separately.
Added free minutes setting for both inbound and outbound calling for all CDR reports.
Changed the web access to https. When a client connects with http it is automatically redirected to https.
Added a check to force the voicemail password to be all numeric.
Added some useful Asterisk CLI commands to the Asterisk menu.
Fixed the formatting of the output from the Execute CLI Command and Execute System Command commands.
Modified the Add Call Route command. The list of destinations now includes voicemail boxes and shows the type (application, extension, sub menu,
voicemail) for each entry.
Fixed a bug in the provisioning of an IAXy device.
Added the script resettodhcp that resets the network and host information to its original state.
When adding a Zap voice service the port number input is now a drop down list with all configured ports instead of an empty input area.
Fixed some parsing issues when looking for Zap services in the configuration files.
All menus that have popups now also have a Description section below the Submit button.
Removed the IP Address field from the Add Extension menu to make adding an extension easier.
Added the Port field to the Modify Extension menu.
Removed the IP Address field from the Add Multiple Extensions menu.
Fixed the voicemail prompt in the stdexten macro. It was playing the busy message instead of unavailable.
Fixed # for blind transfer and turned on * as attended transfer as default.
The Save All Changes To Disk command was not saving the voicemail to disk.
This is necessary to maintain the user's custom messages.
Added a new Festival sub menu to manage the festival service.
Changed the /tmp directory to be mounted on it's own ram disk instead of being a softlink.
Added a web interface to set the different ram disk sizes.
Added support for multiple preconfigured hardware platforms.
Removed the following packages: libcap-devel, libdbi, libdbi-dbd-mysql, libogg-devel, libvorbis-devel, mysql-server, ncurses-devel, newt-devel, rdate and slang-devel.
Removed the shorewall firewall package.
Added documentation on how to run a SIP load test on the server.
Added Add/Modify/Delete Group Extension commands. A group extension will ring multiple extensions in 3 tiers and will then terminate with a selected no answer destination.
Changed the Set SIP IP Addresses command so that it only allows input if the public and private IP addresses are different.
Added packages vsftpd, ethereal, tcpdump and dhcpdump to make debugging of network problems easier.
Added madplay mp3 player to replace mpg123. The mpg123 player can cause stability problems.
The root and vsuser passwords can now be set individually from the web interface The main server web page is now also password protected.
Fixed issue with not being able to pick up parked calls.
Updated the Add/Mod/Del Zaptel device commands to support the new 2 port Digium cards.
Upgraded Flash Operator Panel to version 0.22.

Version 1.19 5/14/05

Added a second version of Asterisk to the system. The user can now select between a stable release version and a development version.
Added a menu to select the Asterisk version to the CGI interface.
Configuration backup directories are now automatically backed up to CF card on reboot or shutdown.
Added bison package. This is needed to recompile Asterisk.
Fixed problem with first and last name not being set correctly in voicemail.conf.
Added default extension 7 for company directory.
Changed default ring time to 25 seconds for all dial plan macros.
Fixed a problem when loading CDR data that could cause an infinite loop.
Fixed minor issue with reconfiguration of hardware.
Added CGI menu to add call routes.
Added preconfigured extensions 601 for SIP and 602 for IAX2.
Added 5 predefined sub menus to the dial plan. To record greetings for these sub menus use extensions 329501-329505.
Fixed minor problem with Modify Extension command.
Added new field for Voicemail Password to Add/Modify Extension commands.
Removed Username field from Add Extension command.
Added mohmp3 directory to backup script so that user updates are maintained.
Changed call forwarding extensions from *72 and *73 to *42 and *43 to avoid conflicts with existing systems.
Modified the default echo cancel parameters for zaptel cards to get better echo canceling results.
Turned off blind transfer for outgoing calls to allow the # key to go through.Redesigned the CGI interface.
Changed to a 3 pane configuration in order to better group the commands.
Turned off busy detect for zaptel devices and increased the number of jitter buffers.
The server now comes preconfigured for DHCP. All hardware is now preconfigured. Added a warning to the CGI interface to let the user know that the server must be reconfigured with a static IP.
Added the IP address to the data being displayed above the login prompt. It also shows if the IP address is static or DHCP.
Removed the local copies of WinSCP and IPSwitchBoard from the server and provided links to their home pages instead.
Modified the Create FOP Buttons File command. It now shows a list of available channels. Only the selected channels will be added as buttons to the FOP.
Added new configuration item to the Data File Configuration menu. The new Automatic Save setting is on by default. If turned off, only CDR data is saved automatically during reboot.
At the end of system startup the server will announce its IP address through the built-in sound card.
Created new default extension 8888 that will read back the current IP address of the server.
Changed the default IAX2 jitter buffer settings.
Once the system is started it will now announce its IP address through the sound card headphone output.
The web interface will only allow the configuration of the static IP address as long as the system is running with a DHCP IP address. Once the system is running with a static IP address the web interface works as expected.
Changed some of the default iax and sip settings to improve reliability.
Modified the Set SIP IP Addresses command so that if all the fields are empty it removes the previous settings from sip.conf.

Version 1.18 4/25/05

Added festival, asterisk-perl and krb5-workstation package.
Added intercom patch for SNOM phones. Also added documentation on how to configure intercom for SNOM phones.
Added new music on hold extension 1000 to fix the moh problem with SNOM phones. Just add "1000@ip_of_asterisk_server" in the "Music on hold server" setting of your SNOM phone.
Added new extension 8001 to dial plan to test if festival is working.
Added alternate mount location for /mnt/data if data CF card not present.
Updated asterisk, zaptel, libpri and asterisk-add-ons. Asterisk now includes the application ChanSpy.
Changed the default to send voice services to the "incoming" context. Also moved the main greeting to the new incoming context. Fixed the dial plan so that incoming callers cannot make outgoing calls.
Changed the kudzu timeout from 30 to 8 seconds.
Updated the documentation regarding the ztdummy module, since it is now automatically loaded.
Fixed a problem with the reconfigure hardware option during boot.
Removed several unused kernel drivers to save space on the CF card.
Changed the stdexten macro to handle call forwarding. Every extension can now specify a call forwarding number and a call forwarding when busy number.
To set the call forwarding number dial *42fwdnumber. To cancel dial **42. To set the call forwarding when busy number use *43... and **43.
Added web interface for call forwarding for both administrator and client.
Added IPSwitchBoard, a free Windows based operator panel for Asterisk.
Added spandsp library and rxfax/txfax applications to asterisk.
Added libtiff-devel package.
Added mime-construct, MIME-Types and Proc-WaitStat perl modules.
Added new menus to establish connections through the web interface.

Version 1.17 3/22/05

Updated Asterisk, libpri, zaptel and added asterisk-add-ons.
Added tftp to the list of services that need their port forwarded.
Changed the Add/Modify Voice Service form so that the service name cannot be changed.
Fixed the provisioning of phones that use DHCP.
Fixed the problem with sendmail taking a long time to start up during boot.
Removed the home page link from the client cgi menu.

Version 1.16 3/14/05

Changed boot sequence to automatically force fsck if necessary instead of dropping to a shell.
Added provisioning for SNOM, Uniden and Grandstream phones via tftp and http.
Added provisioning for Digium IAXy device.
Added tool tips to all menu items. Firefox is now the recommended browser since IE does not show all of the tool tips.
Updated the Flash Operator Panel to version 0.20. Updated the "Create FOP buttons" command to also add buttons for parking extensions.
Moved CF card and USB mount points to RAM disk.
Added support for up to 3 CF data cards and up to 4 USB drives.
Added the packages ncurses-devel, gcc, cpp, glibc-devel, glibc-headers, glibc-kernheaders and openssl-devel.
Added ALSA libraries and drivers. The onboard sound card has been preconfigured with ALSA drivers.
The ALSA drivers now make it possible to use the sound card for paging.
Added default paging extension 8000. This extension pages using the speaker output of the sound card.
Added code in rc.local to clean out any non-ALSA drivers from modules.conf and reboot if necessary.
The Add Multiple Extensions command now works without IP addresses for dynamic extensions.
Added commands to restart, start and stop the Zaptel service.
Added commands to add, modify and delete Zaptel devices.

Version 1.15 2/22/05

Fixed problem with zaptel driver not working correctly.
Updated Asterisk, libpri and zaptel to the latest version.
Fixed the handling of bad data in the CDR data fields.

Version 1.14 2/15/05

Fixed inbound calls for Vonage users by changing the default of the insecure setting to insecure=very.
Fixed a minor typo in extensions.conf. The extension 600 was not commented out correctly.
Replaced the outgoingavail macro with outgoingavail25 as the default. The old default does not work with NuFone accounts, the new one does.
Changed the ssh port from 22 to 32950.
Turned off root login for ssh. Added new user vsuser that can login through ssh.
The Quick Start Guide is now included with a link on the main web page.
Updated SNOM specific documentation to include list of available firmware downloads.
Fixed configuration of ntpd. It now starts by default with a valid configuration.
Upgraded Asterisk to the latest version. The MeetMe application was not working when using two SIP phones. Now it works as expected.
Also upgraded to the latest version of zaptel and libpri.

Version 1.13 2/11/05

Updated the SNOM firmware to the latest versions
Fixed a problem with the hint statement for extensions.

Version 1.12 2/7/05

Created new client mode interface that requires login by extension and password.
This new client mode is designed to be used by customers of service providers.
Fixed pattern matching when parsing voicemail.conf.
Fixed pattern matching when parsing call detail records file.
Modified the Add Extension command to handle missing Caller ID and IP Address.
Added a check box to the Add/Modify Extension commands to skip the creation of a voicemail box.
Changed the defaults for TDM cards. Echo training and busydetect are now on by default.
Added support for USB hard drives. A USB hard drive is now detected at boot and can be used to store CDR data and voicemail.
Updated the Data File Location menu to support Memory, CF Card and USB Drive.
Added libusb and usbutils packages.
Added a new extension 999 to the dial plan. This extension will connect directly to The VoIP Connection office as long as the client machine has Internet access, no voice service required.
Added a link to download JPlayer to the main web page. JPlayer can be used to play gsm files.
Added "Hide Comments" check box to "View Configuration Files" command.
Added callgroup and pickupgroup settings to Add/Modify Extension commands.
Fixed the parsing and handling of caller id for extensions.
Added descriptions for each field for the Add/Modify Extension/Service commands.
Set the default codec for SIP to ulaw in sip.conf.
Set the default codecs for IAX to ulaw and gsm in iax.conf.
Removed several extensions from the filter, so they can be changed with the interface.
Fixed the problem with iax.conf not being updated correctly when adding/modifying/deleting an extension.
Modified the Add/Modify Extension command so that it is possible to add an extension only to the dial plan.
The list of extensions for the Add/Modify/Delete Extension commands is now sorted.
The Add/Modify Extension command now lists the commands of the extension in the report at the end.
The View TFTP files command is now only visible when necessary.
Changed the Add/Modify Service command to show only relevant fields when adding/modifying a Zaptel service.
Working versions of SNOM firmware are now included for easy download.

Version 1.11 2/1/05

Changed the voicemail format from the default setting to wav49 to save space.
Changed the minimum voicemail message length to 6 seconds.
Removed unneeded sections in default configuration files.
Added search pattern (regular expression) to View Call Detail records.
Added new user mode version of the CGI script. This user mode version allows the viewing of data but does not allow administrative access.
Added "Upload/Download Files" command. This allows the uploading and downloading of any file.
Added dhcpd package to allow discovery of tftp server.
Added interface commands to start, stop, restart and get the status of the dhcp server.
Added interface commands to view and edit the dhcp settings.
Added samba-client, samba-common, libjpeg, libtiff, libpng and cups-libs packages.
Added a new boot option to move the data.tar file. This makes it easier to fix problems.
Added "Secure Shell Login" command. This allows shell access to the system using the mindterm java based ssh client.
Added a link to the "Upload/Download" page to download WinSCP.
Added a "Home Page" link to the interface.
Upgraded Asterisk, libpri and zaptel to the latest version.
Upgraded the Linux kernel to version 2.4.22-1.2199 to improve system stability.
Fixed a pattern match error when an extension contains '*'.
Updated all CGI error messages so that they show the value that is wrong.
Updated prompts during boot process to give more information.

Version 1.10 1/28/05

Mount point /mnt/data is now always created even if there is no CF card.
Added new extension macros for calling two and three extensions in sequence.
Added command to CGI interface to set administrator email address. This address will be used to send error and status messages if necessary.
Added command to scale the Flash Operator Panel buttons by a given percentage.
Added command to add multiple, sequential extensions in a single step.
Updated the CGI interface so that an email address is no longer required for adding an extension.
Modified "Delete Extension" command to allow deletion of multiple extensions.
Expanded the Add/Modify Extension menus to handle 8 actions instead of 5.

Version 1.9: 1/26/05

Added new outgoing macro which works with all services.
Updated settings for Broadvoice and Vonage services.
Reworked the View Call Detail Records command. It now has filters for date, time and channels. It also includes fields to enter the billing increment and dollars per increment charge. This can be used as a very simple billing system.
Also added an option to control the CDR data fields that will be displayed.
Added new interface command to delete voicemail messages from one or more mailboxes.
Added ntp and libcap packages.
Added new outgoing macro that works for all cases Fixed the pattern matching when processing voicemail.conf.

Version 1.8: 1/17/05

Changed the caller ID input format when adding/modifying extensions.
Removed all currently unsupported inputs from add/modify extension forms.
Updated the required inputs note at the top of the Add Voice Service form.
Fixed problem with not matching the service correctly when using the Modify Voice Service command.
Fixed the line number setting for services so that it automatically picks the next available for adding and it correctly selects the current one for modifying a service.
Fixed the creation of soft links for directories that are moved to the CF card.
Fixed a problem with sendmail, daemon mode is now enabled.
Upgraded Flash Operator Panel to version 0.19.
Modified the "Add Extension" form so that it automatically fills in the next available default extension information.
Fixed the pattern matching when searching if an extension already exists.
Added Flash Operator Panel Start, Stop, Restart, and Status commands to the CGI menu.
Added command to create the Flash Operator Panel buttons file op_buttons.cfg.
Added command to set the Flash Operator Panel IP address/name.
Fixed the handling of Zaptel devices for Add/Modify/Delete Voice Service command.

Version 1.7: 1/11/05

WARNING: The file /etc/asterisk/extensions.conf has changed substantially. Your extensions in your version of extensions.conf need to be migrated to the new version supplied with this release. The CGI interface will not be fully functional unless this is done.
Added warning about extensions.conf incompatibility to the main interface page.
Added sox, libogg and libvorbis packages Updated dial plan to handle 25 outgoing lines through the CGI interface Modified direct access to outgoing lines.
Before it was 9,8,7 now it is 9 and 0901 through 0925 for a dialing prefix.
This makes it possible to access each of the 25 outgoing lines directly by dialing the correct prefix.
When dialing 7 or 11 digit numbers the first available outgoing line is automatically selected (out of the 25 available).
Fixed the CGI menu to set the local area code so that the existing values are shown.
Added dial plan entries for 911 and 411.
Fixed a problem when editing files with the CGI interface.
Added documentation on how to configure a Digium TDM card.
Added documentation on how to configure the function buttons on a SNOM phone.
Added "Restart Asterisk" command to CGI interface.
Changed default of Add Extension NAT setting to no.
Fixed call parking, not all dial strings allowed transfer.
Renamed "Trunk Number" to "Line Number" in the CGI interface.

Version 1.6: 12/31/04

Added support for native and uncompressed CF file system.
As of this version the OS file system is no longer compressed.
Changed welcome message and added version number.
Added list of ports that may need to be forwarded to main CGI menu.
Added key press to startup to reconfigure the hardware.
Modified startup process to show "Press key" prompts only if relevant user prompts now show a 5 second countdown.
Added additional documentation to the help page.

Version 1.5: 12/22/04

Updated Asterisk, zaptel, and libpri.
Changed boot sequence to make system startup more reliable.
Renamed "Set Location Of Data Files" menu item to "Data File Configuration" and added a setting to specify the number of days to keep voicemail.
Added support for detecting USB based data devices (USB readers, USB keys).
Added hdparm, kbd and usermode packages.

Version 1.4: 12/12/04

Added CGI interface for version history.
Added new outgoing dial plan: when dialing a phone number without prefix (9,8,7) the system will use the first available channel.
Added command line script (/usr/local/bin/vsupgrade) to upgrade the system to a newer version.
Added CGI interface to set new root password (system and http).
Added CGI interface to set local area code.
Added nat setting to extension and service input forms.
Added qualify setting to extension and service input forms.
Modified the data backup script (/usr/local/bin/backupdatatocf).
The /etc/rc* directories are no longer backed up to make upgrades easier.
Added rdate, portmap, mysql and shorewall packages.
Fixed and updated Asterisk installation, specifically: MeetMe application.
Added documentation on how to connect two asterisk boxes via IAX2.
Added documentation on how to turn on the ztdummy module for conference calling.
Added "/etc/modprobe.conf" to the list of editable files to make it easier to enable ztdummy.
Added CGI interface to execute a system command.
Modified boot sequence to make system startup more reliable.
Added CGI interface to relocate voicemail and cdr-csv directories to the CF data card.
Added insecure setting to service input form.
Added port setting to service input form.
Fixed settings for Vonage service.

Version 1.3: 11/24/04

Minor fixes to extensions.conf.
Added CGI interface to set SIP public and private IP addresses.
Added support for NuFone service.
Added third outgoing trunk line.

Version 1.2: 11/22/04

Changed boot sequence.
Removed partition checks for hard drives.
Added an option to skip loading of configuration files at boot.
Added CGI interface to add/modify/delete a context.

Version 1.1: 11/15/04

Added G729 codec and registration utility.
System now finds CF card automatically.
Added CDR records to automatic data backup.
Fixed setting of hostname.
Increased available ram disk space.

Version 1.0: 11/04/04

Initial Release.