v4.3.5

• Include support for asterisk-19

Full Changelog: v4.3.4...v4.3.5

v4.3.4

Major Resolved Issues:

- Reverted (375ef06d): faulty hangup channel fix

v4.3.3

Major Resolved Issues:

• Fixed segfault when handling outgoing calls over pjsip during codec resolution.

v4.3.2-eta

Enhancement:

• Added '--with-asterisk-version=major.minor' configure option

Resolved issues:

• Fix: CLI:module reload causing corruption of GLOB(config_file_name) (FreePBX related)
• Fix: configure asterisk version detection
• Fix: Correct cXML when no callerid_name is available on an outbound parked call
• Fix: Call Answer issue
• Fix: null pointer dereference in sccp_astwrap_rtp_write
• Fix: compilation on FreeBSD

Minor Bug Fixes:

• fix: default CONNECTED keyset buttons
• typo: sccp show softkeyssets -> sccp show softkeysets

v4.3.2-zeta

Enhancements:

• Increased control over Button Lamps on Devstate Feature Buttons
• Added support for asterisk-18
• Added support for AUTO_ANSWER dialplan variable
• Added support for SCCP signaling over TLS (SCCPS)

Resolved issues:

• Parallel answer issue causing segfaults (full rewrite)
• CallInfo when dealing with parking and retrieval of parked calls
• RTP holepunching / dealing with NAT firewalls
• DND/Record/Forward handling on 89xx phones

Minor Bug Fixes:

• cancelling scheduled events
• fix resource leak
• incominglimit handling

v4.3.2-epsilon

Resolved issues:

• Refactor and Cleanup of RTP Handling Improved Async Semantics

Stabilized Features:

• Shared Line Barge Support (Using ChanSpy)
• Single Line Barge Support (Using ExtenSpy)

Minor Bug Fixes:

• Fix ast_channel_varhead causing memory corruption
• Fix for non-atomic operation
• Fix los format_cap reference
• Segfault in SPCPTokenRequest Handler
• Fix Barge Cleanup during Channel Destructor

v4.3.2-delta

Resolved issues:
#464 Conference barge issue after hangup branch_development confirmed enhancement in progress prio 4
#462 Disable cfwdall from CLI branch_development in progress prio 3
#457 allocPBXChannel causing segfault when codecs of called and called parties don't match branch_development bug in progress prio 1 regression waiting_for_reply

Version: v4.3.2-gamma

Resolved issues
#446 Transfer fails from incoming pjsip call (missing asterisk module)
#450 missing audio when codec translation needed (codec reanscoding)

Version: v4.3.2-beta

Fixed #445 One Way Audion when dialing multiple destinations (ie: DIAL(SCCP/123&SIP/321&PJSIP/111))

Version: v4.3.2-alpha

This is a temporary release to provide a tagged version of the pre issue #420 merge

Includes:

• Added 'force_dtmf' to realtime/database example files
• Moved call specific pickup setting to be defined per line instead of per device
• Added 'callhistory_answered_elsewhere = Missed Calls' to fix
  callhistory/phonebook for picked calls.