The Asterisk Development Team would like to announce the release of Asterisk 
17.9.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 17.9.0 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:

Security bugs fixed in this release:
-----------------------------------
 * ASTERISK-29057 - pjsip: Crash on call rejection during high
      load
      (Reported by Sandro Gauci)

Improvements made in this release:
-----------------------------------
 * ASTERISK-29055 - Create a Bridge with video_single mode
     
      (Reported by sungtae kim)
 * ASTERISK-29056 - Increase reg_server column size for
      ps_contacts table realtime
      (Reported by sungtae kim)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-29013 - res_pjsip: Asterisk doesn't stop sending
      invites (with auth) on 407 replies
      (Reported by Sebastian
      Damm)
 * ASTERISK-29124 - res_pjsip: flow transport broken for
      outbound requests
      (Reported by Nick French)
 * ASTERISK-29123 - logger.conf.sample missing comment mark on
      line 115
      (Reported by Andrew Siplas)
 * ASTERISK-28430 - res_rtp_asterisk.c: FRACK!, Failed assertion
      errno != EBADF
      (Reported by under)
 * ASTERISK-29108 - resource_endpoints.c : Memory leak if
      endpoint not found
      (Reported by Jean Aunis - Prescom)
 * ASTERISK-29097 - res_pjsip_config_wizard: Crash when freeing
      string when failing to add extension
      (Reported by Vieri)
 * ASTERISK-26424 - app_voicemail: Undocumented behavior from
      VMSayName
      (Reported by Eric Smith)
 * ASTERISK-29091 - Crash when ast_translator_build_path fails
 
      (Reported by Jasper van der Neut)
 * ASTERISK-29099 - res_musiconhold: Realtime MOH only loads a
      single entry
      (Reported by laszlovl)
 * ASTERISK-29051 - res_pjsip_sdp_rtp: Does not set correct
      values on RTP instance when "auto" DTMF is used
      (Reported
      by Sebastian Damm)
 * ASTERISK-28311 - dsp: ast_dsp_silence_noise_with_energy wrong
      judgment of frame format
      (Reported by 周家建)
 * ASTERISK-29085 - func_curl: Segmentation fault when using
      CURL after setting httpheader CURLOPT
      (Reported by Péter
      Juhász)
 * ASTERISK-24329 - Music On Hold announcement cuts intro of
      music the first time it is played
      (Reported by Thomas
      Frederiksen)
 * ASTERISK-29089 - RTP Ports not cleared after hangup
     
      (Reported by Ross Beer)
 * ASTERISK-29081 - res_stasis: Add compare function for bridges
      moh container
      (Reported by Hajek Michal)
 * ASTERISK-28416 - Unable to get rtp codec payload code for
      slin
      (Reported by Brian J. Murrell)
 * ASTERISK-29014 - res_pjsip_session: Re-INVITE collisions
      aren't handled correctly
      (Reported by George Joseph)

New Features made in this release:
-----------------------------------
 * ASTERISK-29027 - Implement support for History-Info
     
      (Reported by Torrey Searle)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0

Thank you for your continued support of Asterisk!
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