2011-05-09 Leif Madsen * Asterisk GUI 2.1.0-rc1 released. * This release is the first release candidate for Asterisk GUI 2.1 which includes support for Asterisk 1.8. Please help test Asterisk 1.8 support in the GUI by reporting issues to https://issues.asterisk.org in the Asterisk-GUI project. Thanks! 2011-05-06 16:11 +0000 [r5200] Russell Bryant * / (added): Create trunk, copy of current 2.0 branch ================================================================================== Release 2.1.0 ================================================================================== List of Changes since 2.0.6 New Features: ------------- # Compatible with Asterisk 1.6.x and 1.8 # New option to record MeetMe conferences # 14459 - AddQueueMember support Improvements: ------------- # Better compatibility with webkit browsers # Real Asterisk version comparison with greater than/less than functionality # Cache Asterisk version calculations for improved browser responsiveness # default to alwaysauthreject = yes # New options in configuration of vpmadt032. Bug Fixes: ---------- # Fix display of sample bulk user input lines # Fix use of reserved Javascript keyword "char" in context parsing method. # ASTGUI-383 - Modify file-caching behavior to cache contexts separately. # ASTGUI-354 - Fix bug in extensions.conf app argument parsing function where string was added to return array only if it was empty. # Fix bug in function to empty a context where an already-empty context would cause an error and for callback functions to be ignored. # Handle errors pertaining to unknown status from manager events more gracefully for 1.8 compatibility. # Show agents with technology "DAHDI" correctly. # 18646 - Fix issue where BLF does not work for users configured in the GUI # 17316 - Fix issue where conference interface always empty # 18080 - Fix issue where subscribecontext was not validated correctly ================================================================================== Release 2.0.6 ================================================================================== List of Changes since 2.0.5 New Features: ------------- Improvements: ------------- # Print a more helpful error message when dahdi_guiread.conf cannot be read. # 18577 - add -h option to tar command so symlinks are followed when making backup Bug Fixes: ---------- # Fix bug where value for local destination by DID is not gotten correctly, so the value is always empty. # Fixes bug where group = null printed to users.conf for voip trunks. # ASTGUI-316 - Fix bug in dahdi configuration where advanced options weren't saved correctly. # ASTGUI-335 - Fix bug with recent channel groups feature where digital spans are treated as analog when dealing with calling rules. # ASTGUI-316 - Fix issue where previous options line was not deleted before adding new options line. # ASTGUI-317 - Fix issue where values for flash, rxflash, and signalling are not saved because value for dahdichan is being updated after. # Fix issue where macaddress equals extension number when the user is NOT a SIP user. # ASTGUI-166 - Add missing tooltip for MWI From under Sip Advanced Settings. ================================================================================== Release 2.0.5 ================================================================================== List of Changes since 2.0.4 New Features: ------------- # Allow caller ID to be set per outgoing calling rule. # Add ability to configure IAX2 call token and call number limits Improvements: ------------- # Expose channel groups in the GUI. # Faster loading of extensions on status page. # Updated Digium logo # Better error message when agent callback extension is invalid. Bug Fixes: ---------- # Split comma-separated IP ranges into separate localnet lines in sip.conf. # When validating input, allow spaces, (), - and + in user caller ID field. # ASTGUI-219 - Add tooltip for MWI from setting under Users. # ASTGUI-302 - Correct tooltip for pattern field in calling rules. # ASTGUI-238 - Correct a wrong tooltip in Calling Rules. # ASTGUI-236 - Fix issue with advanced SIP and IAX settings where blanking out a value in the GUI resulted in the parameter remaining in the config file in the form "paramater=". # ASTGUI-250 - Fixed bug when creating or editing calling rule where a blank filter creates an unusable dialplan. # Fix error with dialplan extension priorities. # Fix spelling error in a pathname in the listfiles script. # ASTGUI-251 - Fix bug where lines in users.conf were not deleted when field values were emptied. # ASTGUI-139 - Disallow any characters except \d+-\d+ in Extensions for Parked Calls on Call Features page. # 'g' not being added to analog trunk global variable value in extensions.conf # ASTGUI-156 - Fix problem with application map table where last-added application map was the only visible. # ASTGUI-100 - Allow # and * in field which configures number to be dialed through analog trunk from voice menu. # ASTGUI-116 - Added null checking to followMe_MiscFunctions.push_newdest(). # ASTGUI-244 - Fixed a bug with filter in system status where some extensions that should be shown are hidden. # Add callingrules filter tooltip. # ASTGUI-242 - Filter tooltip displaying failover trunk tooltip text. # ASTGUI-199 - Fixed issue where filter function was being added to the calling rule incorrectly. ================================================================================== Release 2.0.4 ================================================================================== List of Changes since 2.0.3 New Features: ------------- # AA50-2035 - Paging & Intercom # AA50-1951 : view DHCP Leases on the AA50 # Option to set the global outbound 'CallerId Name' # g729 registration on AA50 # SMTP authentication for voicemail-Emails on AA50 # AA50-2302 - Add LINEKEYS variable to allow for the user to set the number of keys assigned to a given line. # AA50-2273 Show parked calls on status/welcome page Improvements: ------------- # Upgrading from jquery 1.2.2 to 1.2.6 # improvements to CDR page # ASTGUI-50: Add additional Trunk Edit options - 'fromdomain, fromuser, and insecure' # AA50-2266 : Add a button for AA50 to call the update_tz script # AA50-1771 : An option to allow a "full" backup, from the GUI, that includes voicemails, recorded prompts # Refresh active conferences list every 5 seconds # Option to manually refresh current state of all users. # AA50-2307 - Setting 'Direct Voicemail Dial' doesn't add the 'a' extension for VoicemailMain # AA50-2005 : Add option to specify what format to record prompts in # Avoid having to answer confirmation for each user voicemailbox while deleting multiple users. Bug Fixes: ---------- # Fix: GUI adding duplicate queue members while editing queues # guiversion.js was getting overwritten during 'make install' # Use 'module reload' instead of 'reload' on 1.6 and above. # AA50-2301 - outgoing cid not set unless cid matches extension # Fix for Asterisk 1.6 trunk - infinite reload loop # ASTGUI-47 - Fails to change time interval on existing incoming call rules # Allow "pass-through callerid" for FollowMe calls via trunks # Fix followme issues on analog lines # AA50-2234: fixing wiered 'ls' behavior on AA50 (flashupdate.html) # AA50-2298 : Dial outside number in FollowMe has problem with hyphen ================================================================================== Release 2.0.3 ================================================================================== List of Changes since 2.0.2 New Features: -------------- * Show Realtime Status & updates of Conference Rooms in Welcome page Display the number of people in each conference room and the duration * Add 'AGI' in Voicemenus, Load list of available AGI scripts from agi-bin directory * Exposing some more apps in VoiceMenus - SayDigits, SayAlpha & SayNumber, Ringing * New Feature/Improvement: Advanced mode editing of voicemenus without opening FileEditor * Show list of available drivers in hardware page * Adding a buy now button to the 'IVR Voice Prompts' on the digium store * Adding button to delete an application map (instead of having to leave the feature name to blank ) * Adding Macro option in voicemenus * Introducing.... The CDR Page! Originally written by bbryant in 1.0, then taken to the 2.0 GUI by bkruse. * Adding an Options section where you can enable/disable CallScreening & other followme preferences Improvements: -------------- * Improvement: Add Checkboxes in debugger to show/hide types of Log messages * Avoid calling getElementById twice * Fixes for calling rules as reported by dkerr. (closes issue #13528) * Fix for AA50-2233 : Cannot add VoIP service provider without a username and password * Some minor performance improvements to Array prototype functions * Improving on the Error/Debug log functions ASTGUI.Log.(Debug/Error/Info/Warn/Console) instead of ASTGUI.(ErrorLog/debugLog/FirebugLog) * Remember 'Show Advanced Options' preference using a cookie Bug Fixes ---------- * config/js/astman.js: Fixing a condition that prevented displaying errors in IE (debug mode) and then reload * Fixing some firebug errors about 'sessionData undefined' that has been around for a while * config/js/index.js: Fix: Resize main iframe when switching to/from Advanced mode * Fix for : Cannot add multiple trunks with same username using version 2.0 of GUI http://bugs.digium.com/view.php?id=13497 Fix: use tr.username as trunk name when no trunk exists by tr.username, else use trunk_x (closes issue #13497) * config/js/astman.js: RemoveCookie function * Improvement: make sure hasvoicemail is checked when followme is enabled ( as followme is called from macro-stdexten ) * Fix: GUI overwriting any custom queue settings in queues.conf ( reported by David Kerr ) * config/js/index.js: Tell getChunksFromManager to return to us an object, not an array. (closes issue #13695) great find sumo. * config/iax.html: Change to the iax global settings to no longer overwrite settings with a blank (thus introducing a lot of unneeded behaivor). Cleaned up the codec parsing to alos truely allow the "disallow" field. Dbailey set some default values according to iax.conf * config/js/pbx.js: Bug report where setting a trunk to dynamic (where the other Asterisk box will register to it) is not working properly, as the GUI tries to register with the host "dynamic". This fixes the issue. (closes issue #13529) (patch by bkruse) (tested by dkerr) * config/sip.html: Changed to the sip global settings to no longer overwrite settings with a blank (thus introducing a lot of unneeded behaivor). Cleaned up the codec parsing to also truely allow the "disallow" field. Dbailey set some default values according to sip.conf as well. * Set the default timezone for a user to see. * Cleaning out the rest of the OnHold stuff. * We never get the "On Hold" status, removing it. * Make active channels only appear if you are in "Advanced Mode" * Cleaning up the BulkAdd description * Make BulkAdd only show up with in "Advanced Mode" * 'Digit Key Map' tooltip not being displayed properly. * Second attempt at fixing : AA50-2279 - gui panics when aa50 is rebooted while it is displaying the "call features" tab * requests.txt: Unixdawg request for loading paths via a config file * try loading guiversion.js and then continue with rest of the init process * IE going crazy when using default as an object property using 'default' instead. Took me more than a couple of hours to locate this. * Fix for : AA50-2279 - gui panics when aa50 is rebooted while it is displaying the "call features" tab * AA50-2224 - Missing tooltips in FollowMe Page * minor typos * Fix for http://bugs.digium.com/view.php?id=13540 Issue: Voice menu extensions are duplicated in extensions.conf after editing the voice menus in GUI. Note: This occurs after editing the voice menu, logout and login. Extensions.conf sample: [voicemenus] exten = 7000,1,Goto(voicemenu-custom-1|s|1) exten = 7001,1,Goto(voicemenu-custom-2|s|1) exten = 7001,1,Goto(voicemenu-custom-2|s|1) exten = 7000,1,Goto(voicemenu-custom-1|s|1) * Fix for AA50-2278 - MOH not working with certain versions of Asterisk 1.4 * ASTGUI-41 : No varying or more than 5 digit extensions allowed * Fix for http://bugs.digium.com/view.php?id=13541 Description: Create ring group and assign end action (e.g. go to voicemail for user) and save ring group settings. After save, go to edit ring group and the end action does not appear (e.g. go to user voicemail), it appears to be blank in GUI. I checked the extensions.conf file and the end action IS there, but does not show in GUI. This is confusing because it appears the end action was not set in the GUI when it was. Asterisk GUI 2.0 Build 3855 * Fix for Issue 2 of http://bugs.digium.com/view.php?id=13540 - Typo on Voice Menu settings. # options is mislabeled as * and vice versa. E.g. if you set # in the GUI it actually sets * option in extensions.conf file and not #. * AA50-2223 - Show 'Custom App' in Voicemenu options only in advanced mode * Check for duplicates in all extensions - not just users * While adding range of users, instead of throwing a 'duplicate alert' keep continuing until the number of extensions are added