Release Notes: Support for SIP "peg legged" call connect (and mid-call RTP suspend and codec re-negotiation) whereby one call leg is re-invited to shift codec mid-call before initiating RTP proxy join. Cleaner support for various G.72x adpcm formats. Some other general cleanup in the Bayonne SIP driver code.
Release Notes: Bayonne peer calling between SIP UAs through reinvites has been implemented. Other changes have been made to clarify the operation of Bayonne dialing plans.
Release Notes: Enhancements for building dialing plans in config files including merging of plans, for supporting passing of call completion states to external apps such as for voice broadcast, for management of call instance for ring notification, and better functionality for configured call limits. Greatly improved server shutdown.
Release Notes: Use of exosip2 stack release 3.0.0 to support snom and Cisco IP phones. Reduction in the use of global locking in the SIP stack driver to improve scaling. SIP re-INVITE support has been added to allow IP phones to place Bayonne on hold or transfer sessions. Invite session expiration support has been added. Always clean shutdown of active calls for exiting, and call disconnect event if the associated SIP registry expires or is deactivated.
Release Notes: Support was added for receiving SIP presence publication messages, including an option to support broken SIP clients which cannot authenticate SIP publish events. SIP state and call traffic information is now persistant over runtime server configure reloads.
Release Notes: Fixes issues related to loading multiple arbitrary drivers and selecting audio/channel encoding when performing gateway operations between card drivers and protocol stacks, such as for running Bayonne PSTN gateways. Fixes connect issues with timing control and cancelling call requests. Some small fixes for Java support, PPC, and x64 architectures.
Release Notes: Bayonne::Server.pm was introduced to access Bayonne Web services from Perl applications. Improvements were made in reducing RTP proxy latency and re-syncing RTP timestamps for lost and silent packets on joined calls. New conditional tests were added in Bayonne scripting to find if users are registered, available, and reachable. The "block" directive in BayonneXML is correctly used to create automatically numbered blocks if unnamed.
Release Notes: This release includes support for priority driven registration and failover for outbound proxy selection, and for selecting internally registered SIP endpoints to impliment meet-me/follow-me call groups.
Release Notes: SIP issues with tracking for session call termination were fixed. Deregistering on shutdown was fixed. The correct realm is used when authorizing to proxies. Call information (ID) is correctly passed for session joins. Call ID is properly presented on invite. Anonymous calling support defined by virtual host has been fixed. SIP externals can be placed into call groups. The SIP driver can now use multiple threads for event dispatch.
Release Notes: This release enables anonymous sessions to call registered users to create public contact sites for publicly reachable SIP URIs in building telephony presence on the Internet, rather than simply emulating 20th century telephone concepts over it. The Web services have been updated for basic authentication and support for new server call traffic statistics. Other improvements were made in calling of Bayonne managed SIP URIs, including the option for automatic forwarding for unregistered external users.