[asterisk-dev] Asterisk 17.9.0-rc1 Now Available
Asterisk Development Team
asteriskteam at digium.com
Thu Nov 12 07:38:11 CST 2020
The Asterisk Development Team would like to announce the first
release candidate of Asterisk 17.9.0.
This release candidate is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 17.9.0-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-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 lvl)
* 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 ���������)
* ASTERISK-29085 - func_curl: Segmentation fault when using
CURL after setting httpheader CURLOPT
(Reported by P��ter
Juh��sz)
* 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 candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0-rc1
Thank you for your continued support of Asterisk!
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