Asterisk 1.8.5 Released! ¿Adiós a los bloqueos de canales SIP?

Han tardado, pero la espera ha merecido la pena.

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)