Por fín está disponible la versión de Asterisk 1.8.5 donde han corregido algunos bugs importantes como el bloqueo de canales cuando se realizan transferencias atendidas mediante SIP, y en determinadas configuraciones de conexiones mediante TCP/TLS.
- Fix Deadlock with attended transfer of SIP call
(Closes issue #18837. Reported, patched by alecdavis. Tested by Irontec, ZX81, cmaj) - Fixes thread blocking issue in the sip TCP/TLS implementation.
(Closes issue #18497. Reported by vois. Patched by dvossel. Tested by vois, rossbeer, kowalma, Freddi_Fonet) - Be more tolerant of what URI we accept for call completion PUBLISH requests.
(Closes issue #18946. Reported by GeorgeKonopacki. Patched by mmichelson) - Fix a nasty chanspy bug which was causing a channel leak every time a spied on channel made a call.
(Closes issue #18742. Reported by jkister. Tested by jcovert, jrose) - This patch fixes a bug with MeetMe behavior where the ‘P’ option for always prompting for a pin is ignored for the first caller.
(Closes issue #18070. Reported by mav3rick. Patched by bbryant) - Fix issue where Asterisk does not hangup a channel after endpoint hangs up. If the call that the dialplan started an AGI script for is hungup while the AGI script is in the middle of a command then the AGI script is not notified of the hangup.
(Closes issue #17954, #18492. Reported by mn3250, devmod. Patched by rmudgett) - Resolve issue where leaving a voicemail, the MWI message is never sent. The same thing happens when checking a voicemail and marking it as read.
(Closes issue ASTERISK-18002. Reported by Leif Madsen. Resolved by Richard Mudgett) - Resolve issue where wait for leader with Music On Hold allows crosstalk between participants. Parenthesis in the wrong position. Regression from issue #14365 when expanding conference flags to use 64 bits.
(Closes issue #18418. Reported by MrHanMan. Patched by rmudgett)