Date: March 07, 2011

Change Log V214-1.0002

Critical: During this upgrade, All SIP trunks using host=dynamic, will not work right away. Please ensure the IP of your SIP trunk is added in the permit networks. If you need any assistance, please email support@positrontelecom.com or call - +1-514-664-4719

Feature: Accounting Feature
Feature: Create SIP-Provider Templates
Feature: Bulk Import Phone Book
Feature: Bulk Import Phone-Provisioning
Feature: Bulk Import Templates on the GUI
Feature: Enable "ring extension before forwarding" and "ring time before forwarding" for any forward location.
Feature: Add phone book support
Feature: Find IP Address of the Device using a FXS extension
Feature: Update User status page
Feature: Firewall Implemented
Feature: New Dialplan Rules Page Layout with Pattern Generator and Help
Feature: Permit/Deny Networks for SIP/IAX Trunks and user Templates
Feature: Add multiple host ip support
Feature: Blockip tool
Feature: Add "CallerID Prefix" and "Answer" incoming call rule options
Feature: Monitoring Multiple mailboxes
Feature: SSH, TFTP, SNMP, Astmanproxy Services added to Network Page
Feature: Add "SIP Trunk Registration Timer" to the PBX settings page
Feature: Add 'requirecalltoken' option for IAX trunks
Feature: Split "network" configuration file into "network", "firewall", "smtp" and "sip_rtp"
Feature: Split "PBX Settings" Page to multiple Tabs
Feature: Add support for mail log in diagnostics
Feature: Add support for mail log, GUI Log, Web server Log to Storage page
Feature: astmanproxy support
Feature: Write detailed "Create configuration" errors to the GUI log
Feature: Add "Default Mailbox" option for user extensions
Feature: Show warning if phone password is weak
Feature: New step "Disa-nopassword" in IVR menus
Feature: Diagnostics Page now has "Start/Stop" console and Execute asterisk command
Feature: Show DHCP leases in Logs
Bug: Fix 3G default IPs
Bug: Fix 3g - invalid interface
Bug: Fix DISA
Bug: Fix DHCP when using in DMZ mode
Bug: Show message to reconfigure phones when SIP/RTP ports change
Bug: GUI with a new menu layout
Bug: Fix bugs in blocked network
Bug: Fixed an issue for Voicemail application - maximum messages for each mailbox
Bug: Show Yealink line key configuration - phone provisioning
Bug: Fix double factory reset
Bug: Apply SMTP settings when restoring backup
Bug: Changed default password for 6005 to a 12 digit passsword
Bug: zoom key 3g fix
Bug: Display "You have no messages" instead of "This table is empty"
Bug: Alphabetically sort files in Sound Manager 
SIPCarrier: Dezmocom-Canada added to the GUI

Date: August 23, 2010

Change Log V214-1.0001

Critical: During this upgrade, All the Phone Config Files uploaded under phone provisioning will be lost, It is advised to take a backup of them and if you need any assistance, please email support@positrontelecom.com or call - +1-514-664-4719
Below are commands to backup your phone provisioning files
 
Windows (Pscp software: http://the.earth.li/~sgtatham/putty/latest/x86/pscp.exe)
pscp -scp -unsafe -r root@192.168.1.2:/var/lib/postel/http/pp/* pp/
pscp -scp -unsafe -r root@192.168.1.2:/tftpboot/* pp/
 
Linux, you can use the command
scp -r root@192.168.1.2:/var/lib/postel/http/pp/* pp/
scp -r root@192.168.1.2:/tftpboot/* pp/

Critical:  Enforce phone password for SIP extensions. Generate random password on upgrade if empty.

Notice: Add alert message if IP address was changed.
Bug: Selected VoIP trunk not remembered on the GUI for Dial Group
Bug: CF Formatting errors are now logged in the GUI Log
Bug: Listen for key presses when executing the Play Invalid Step in IVR.
Bug: Fix for missed phone hangup events rendering FXS port unavailable
Bug: Fix Incoming Calls CDR
Bug: Fix Brazil and UK callerID.
Bug: Three way calling and call waiting fixed for FXS
Bug: Loud-ring.gsm is replaced with a new ring file
Bug: default music on hold added to the configuration
Bug: Updated Startup scripts for webserver and cron
Bug: Cron application fixed
Bug: RotateCDR and RotateLog scripts are updated.
Bug: Fix Early Media calls on the BRI Interface
Feature: Configurable ProgressInband option for SIP
Feature: "Call Answered elsewhere" to avoid missed calls on the phones when the call is answered by another member in the ringgroup
Feature: SIP/RTP ports are now configurable on Network Page 
Feature: Mass upload of extensions using a CSV File 
-- Extension,first name, last name, language, voicemail (0/1), caller id, external callerid, name directory (0/1), email address, cellphone
Feature: Download Phone Configuration files other than Yealink
Feature: Yealink auto phone provisioning
Feature: Configurable Ethernet Port 4 for WAN/LAN/Monitor mode
Feature: "Get External IP" button in Network/SIP/RTP to find the external IP address
Feature: DHCP DNS and Gateway are configurable which is provided to the Board's DHCP Clients 
Feature: Hot Line Feature on FXS Ports
Feature: Add "Music on Hold" option under trunks (analog and voip).
Feature: Download button under "Sound Manager/IVR"
Feature: Configurable Ring Times in dial plans/rules/groups
Feature: Add option to ring extension before forwarding (admin/user)
Feature: User Level login for SIP/Analog/Virtual Voicemail Users
Feature: "System Name" PBX option to configure SIP Useragent for the System
Feature: NTPD support
Feature: Fix blind transfer from forwarded calls
Feature: Sort Call Recordings by time and show one day at a time
Feature: Forwarding to external number under User --> Extensions
Feature: Hard Factory reset implemented when using the reset button on the unit.
Feature: Soft Factory Reset and Hard Factory reset option added in Reset to defaults Section
Feature: Enhanced "Follow me" (Announce Callers Option)
Feature: Add "Qualify" option for VoIP trunks and user templates - needed for some SIP/GSM gateways
Feature: Brazil CallerID Support
Feature: "On Demand" Time Frames
Feature: Validation to ensure the right firmware is uploaded
Feature: SIP Video Support with H264 codec
Feature: BLF Implemented for Parking Rooms. Can be used to simulate the SLA Behavior
Feature: Language Option added at extension level
Feature: Dial Command added in IVR menus
Feature: DHCP Boot Server added under Network Page
Feature: Polycom Presence
Feature: New IVR Design implemented (Steps can be created under each key/group)
Feature: Cell Phone Prefix and cell phone field under each extension
Feature: BLF Pickup Prefix/Pickup Prefix
Feature: Voicemail Prefix used for sending calls directly to an extensions voicemail
Feature: On Demand Recording
Feature: Extension Paging Prefix - Allows users to page a specific extension
Feature: Voicemail to Email body is now configurable
Feature: Playback and Unlimited ring time added for Ring Groups 
Feature: Deleteall button added for the CDR page, to clear the CDRs
Feature: Ring Time per extension
VoIPCarrier: BroadConnect VoIP Provider added to the GUI

Date: March 5, 2010

Change Log V214-0.0007

Bug: Data on Compact Flash and USB is lost when doing factory reset or firmware upgrade and is fixed as part of this release.
Bug: DMA Fix resolve loss of packets 
Bug: Fix Format Compact Flash Webpage
Bug: Fixed Arizona Time Zone for NTP
Feature: Up, down buttons implemented in dialplan rules and incoming rules. Useful for sequential parsing. 
Feature: Introduced interoffice checkbox in dialplan rules, when checked calls dialed using interoffice pattern will only use the callerid of the extension. 
Feature: Add DID as an option in the To Field for Incoming calls. This is for calls to be send to the same extension as the DID, useful when we do a SIP trunk between two Offices. 
Feature: Added GSM codec as an option for SIP and IAX trunks 
SIP Carrier: SIP Gate UK VoIP provider added on to the GUI 
Feature: CallGroup and Pickupgroup Options added for the sip extensions. Used with Pickup option specified under PBX Settings. 
Feature: Disable Voicemail option for sip and analog extensions 
Feature: Optimized GSM library for better transcoding between various codecs.
Feature: Ring All Extension added under Users -> RingGroups
Feature: Phone Provisioning is added under PBX. All the phones can now get their config files from the PBX using Http
Feature: SIP localnet option added under System -> Network

Fix factory reset module
Update Validation Script