summary
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To distribute Polycom default configuration files to tftproot directory
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Create new menu "PIN management"
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Deploy interop.sipxecs.org with sipXinterop config upon main (4.1) sipXecs
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When performing consultative transfer of a call to VM and AA, IVR session is not always restarted after transfer completion
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Higher load average and SUT becomes slow, results in call failures during soak testing.
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Change precedence of sipXNatRoute
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Integrate with Exchange 2010 MWI
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add support for "sips" urls
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Improve the efficiency of IMDB and XML dataset replication
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Add Wait for Moderator feature into FreeSwitch conferencing
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Improved process start/stop/control system from sipXsupervisor
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Move registry and proxy plugin configurations to separate files
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re-structure of the sipXbridge service configuration.
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Phonebook csv name change causes duplicate/incorrect entries on IP sets
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Activate multiple Polycom firmware version, select which version to run from phone "Firmware" config setting
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CSS errors when navigating Admin and User portal pages
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E911 Alarming and Notification Function
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Allow creation of E911 zones in sipXconfig
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remove both Always Available and Monitor Presence
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Caller gets connected to two voicemail on consultative transfer
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"Phones" screen in User Portal should display all phones on which the user is defined
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Dialplan: Allow blocking of incoming calls (call blocking) and source routing based on Caller ID
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UI and administration code to support source routing/blocking and source routing attendant feature
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Enable call routing based on incoming CallerID ("Source Routing Attendant")
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dial plans should not allow all 'Resulting Call' fields to be empty
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disambiguate transfer target addresses
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remove jars from subversion repository
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Add high availability / clustering support to Voice Mail
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Toggle column visibility |
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