[asterisk-dev] Asterisk 17.1.0-rc1 Now Available

Asterisk Development Team asteriskteam at digium.com
Thu Dec 12 06:28:17 CST 2019


The Asterisk Development Team would like to announce the first
release candidate of Asterisk 17.1.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 17.1.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release candidate:

Security bugs fixed in this release:
-----------------------------------
 * ASTERISK-28589 - chan_sip: Depending on configuration an
      INVITE can alter Addr of a peer
      (Reported by Andrey  V.
      T.)
 * ASTERISK-28580 - Bypass SYSTEM write permission in manager
      action allows system commands execution
      (Reported by Eliel
      Sarda��ons)
 * ASTERISK-28495 - res_pjsip_t38: 200 OK with SDP answer with
      declined stream causes crash
      (Reported by Alexei
      Gradinari)

Improvements made in this release:
-----------------------------------
 * ASTERISK-28602 - res_pjsip_outbound_registration: Maximum
      retries reached
      (Reported by Daniel)
 * ASTERISK-28586 - Typo in README-SERIOUSLY.bestpractices.md
  
      (Reported by Sam Banks)
 * ASTERISK-22192 - [patch] Allow voicemail forwards with ODBC
      backend when format differs from attachfmt column
     
      (Reported by cmaj)
 * ASTERISK-28567 - Problem with ASTERISK-20207: Asterisk should
      clear out any .lock files in the voice mail directory on
      startup.
      (Reported by Michael)
 * ASTERISK-28542 - [patch] add the ability for asterisk to
      generate on-hold re-invites
      (Reported by Torrey Searle)
 * ASTERISK-28512 - Add pass-through support for H.265 (HEVC)
      codec
      (Reported by Florian Floimair)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-28641 - res_pjsip Segfaults when realtime
      configuration to an AOR points to a not existent AOR
     
      (Reported by Ross Beer)
 * ASTERISK-28644 - Stale comment in app_queue about ring_entry
      exception
      (Reported by Walter Doekes)
 * ASTERISK-28637 - chan_sip+native_bridge_rtp: directmedia
      compatibility check failure when negociated ptime is not default
      ptime.
      (Reported by Frederic LE FOLL)
 * ASTERISK-28445 - res_pjsip_session: ast_json_vpack: Invalid
      UTF-8 string on hangup when TEST_FRAMEWORK enabled
     
      (Reported by Bernhard Schmidt)
 * ASTERISK-28631 - res_parking: Doesn't park when parkee and
      parker are the same
      (Reported by Ross Beer)
 * ASTERISK-28621 - Enforce T.38 error correction mode at 200 ok
      received  
      (Reported by Salah Ahmed)
 * ASTERISK-28624 - res_pjsip_outbound_registration: add SRV
      failover
      (Reported by Kevin Harwell)
 * ASTERISK-28608 - app_amd: Use time calculation to calculate
      timeout
      (Reported by Michael Cargile)
 * ASTERISK-28615 - chan_dahdi: PRI span status may stay "Down,
      Active" after a short alarm
      (Reported by Frederic LE FOLL)
 * ASTERISK-28576 - res_rtp_asterisk: ICE Completion Crash when
      sent packet length doesn't match
      (Reported by Joshua
      Elson)
 * ASTERISK-26481 - FILE function grabs garbage along with read
      data when target line has no newline
      (Reported by Jonathan
      Harris)
 * ASTERISK-28618 - bridge_softmix: hold not cleared when
      joining a softmix bridge
      (Reported by Kevin Harwell)
 * ASTERISK-28616 - parking: Deadlock when multi call parking
  
      (Reported by Joshua C. Colp)
 * ASTERISK-28423 - ARI causes STASIS Deadlock
      (Reported
      by Ross Beer)
 * ASTERISK-28604 - app_meetme, chan_ooh323 and cdr_mysql don't
      build on 17.0.0
      (Reported by George Joseph)
 * ASTERISK-28572 - Memory leaks in res_calendar_exchange and
      res_calendar_icalendar
      (Reported by Yoooooo Ha)
 * ASTERISK-28585 - ari/resource_events: Crash in event session
      cleanup
      (Reported by Kevin Harwell)
 * ASTERISK-28590 - utils.c throws repeated warnings;
      "pthread_attr_setstacksize: Invalid argument"
      (Reported by
      Speed Dial Dave)
 * ASTERISK-28578 - race condition on pjsip channelstats
      command
      (Reported by Salah Ahmed)
 * ASTERISK-28571 - cdr_pgsql: accesses obsolete (and finally
      removed) column
      (Reported by Christoph Moench-Tegeder)
 * ASTERISK-28575 - MWI Send Notify Crash on 16.6
     
      (Reported by Joshua Elson)
 * ASTERISK-28574 - pjproject fails to build on 16.6.0, works on
      16.5
      (Reported by Niklas Larsson)
 * ASTERISK-28561 - Asterisk Deadlocks
      (Reported by
      Aheliotech)
 * ASTERISK-28086 - chan_pjsip: Crash when initiating PlayDTMF
      over AMI
      (Reported by Jeremiah Gadd)
 * ASTERISK-28552 - res_pjsip_mwi: Frack during unload on
      unsolicited_mwi container
      (Reported by Kevin Harwell)
 * ASTERISK-28566 - CDR backend unload problem during active
      call(s)
      (Reported by Marian Piater)
 * ASTERISK-28553 - stasis.c: Crash during unload
     
      (Reported by Kevin Harwell)
 * ASTERISK-28544 - Wrong contact representation in ipv6 mode
  
      (Reported by J��rgen H)
 * ASTERISK-28534 - Segmentation fault when there is no priority
      for an extension
      (Reported by Timothy Vanderaerden)
 * ASTERISK-28463 - res_pjsip_path: Crash when invalid contact
      is configured
      (Reported by Juan Martin)
 * ASTERISK-28521 - pjsip: Memory Leak
      (Reported by Mark)
 * ASTERISK-28523 - Asterisk 16.5.0 Memory leak
      (Reported
      by Cyril Rami��re)
 * ASTERISK-28538 - chan_pjsip: Deadlock on fax detection
     
      (Reported by Joshua C. Colp)
 * ASTERISK-28536 - Asterisk release candidates fail to build on
      FreeBSD
      (Reported by Guido Falsi)
 * ASTERISK-23756 - setvar directive when used in template and a
      child of said template, results in duplicate variable names
    
      (Reported by Michael Goryainov)
 * ASTERISK-28527 - ChanIsAvail() creates a CDR if
      unanswered=yes is set in cdr.conf
      (Reported by Frederic LE
      FOLL)
 * ASTERISK-28525 - chan_dahdi: set CHANNEL(hangupsource) when a
      PRI channel hangs up
      (Reported by Frederic LE FOLL)
 * ASTERISK-28511 - codec_resample: Bad sound quality when up
      sampling from SLIN16 to SLIN32
      (Reported by Ruddy G)
 * ASTERISK-28499 - translate: Crash when frame does not have a
      "src" field set
      (Reported by Gregory Massel)
 * ASTERISK-25592 - chan_unistim: Clang Warning: variable sized
      type not at end of a struct
      (Reported by Alexander Traud)
 * ASTERISK-28488 - pjsip mwi: n+1 sip notify's sent on
      re-register
      (Reported by Chris Savinovich)
 * ASTERISK-28509 - PJSIP cnonce generated on Linux contains 36
      characters, NEC only supports up to 32 characters
     
      (Reported by Dan Cropp)
 * ASTERISK-28505 - app_voicemail/IMAP: segfault in
      leave_voicemail because not checking mailstream
      (Reported
      by Alexei Gradinari)
 * ASTERISK-28487 - compile menuselect on gentoo
      (Reported
      by Kilburn)
 * ASTERISK-28472 - Asterisk occasionally passes a NULL as
      srtp->session to srtp_protect/unprotect causing SEGV
     
      (Reported by Jonas Swiatek)
 * ASTERISK-28498 - cel / cdr: Event times may be incorrect
    
      (Reported by Joshua C. Colp)
 * ASTERISK-28480 - json integer overflow in ssrc and timestamp

      (Reported by Salah Ahmed)
 * ASTERISK-28228 - res_pjsip: pjsip show contacts prints double
      entries
      (Reported by Ian Jones)
 * ASTERISK-28483 - packet lost on UDPTL wrap around
     
      (Reported by Torrey Searle)

New Features made in this release:
-----------------------------------
 * ASTERISK-28614 - app_senddtmf: Allow "receiving" DTMF with
      PlayDTMF instead of only "sending"
      (Reported by lvl)
 * ASTERISK-28613 - func_curl: CURLOPT cannot set Content-Type
      header
      (Reported by Martin Tomec)
 * ASTERISK-28533 - func_jitterbuffer: Add support for video
      synchronization
      (Reported by Joshua C. Colp)
 * ASTERISK-17808 - [patch] Unregister a realtime moh class
    
      (Reported by Byron Clark)
 * ASTERISK-28489 - Channel variable SIPFROMDOMAIN for
      chan_pjsip to setup From header URI domain
      (Reported by
      Stas Kobzar)

For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0-rc1

Thank you for your continued support of Asterisk!
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20191212/866a7c1f/attachment.html>


More information about the asterisk-dev mailing list