Asterisk 18.1.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.1.0 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:
———————————–

pjsip: Crash on call rejection during high load
(Reported by Sandro Gauci)

New Features made in this release:
———————————–

Implement support for History-Info
(Reported by Torrey Searle)

Bugs fixed in this release:
———————————–

res_pjsip.so fails to load when bundled pjproject is compiled without libssl
(Reported by Walter Doekes)
Any curl response checks out as valid even if 404 is returned.
(Reported by dovid)
res_pjsip: Asterisk doesn’t stop sending invites (with auth) on 407 replies
(Reported by Sebastian Damm)
sip_to_pjsip.py: doesn’t read globbed includes
(Reported by Michael Newton)
GCC Warnings with OPTIMIZE=-Og make
(Reported by Alexander Traud)
GCC Warnings with OPTIMIZE=-Os make
(Reported by Alexander Traud)
GCC Warnings: ‘%s’ directive argument is null.
(Reported by Alexander Traud)
res_pjsip: flow transport broken for outbound requests
(Reported by Nick French)
config: Sample features.conf incorrectly includes ” around sound files
(Reported by Benjamin M.)
logger.conf.sample missing comment mark on line 115
(Reported by Andrew Siplas)
res_pjsip_session: Asterisk 18 does not progress calls due to codec negotiation after upgrading from Asterisk 16
(Reported by Ross Beer)
res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
(Reported by under)
resource_endpoints.c : Memory leak if endpoint not found
(Reported by Jean Aunis – Prescom)
app_voicemail: Undocumented behavior from VMSayName
(Reported by Eric Smith)
res_pjsip_config_wizard: Crash when freeing string when failing to add extension
(Reported by Vieri)
Crash when ast_translator_build_path fails
(Reported by Jasper van der Neut)
res_pjsip_sdp_rtp: Does not set correct values on RTP instance when “auto” DTMF is used
(Reported by Sebastian Damm)
res_musiconhold: Realtime MOH only loads a single entry
(Reported by laszlovl)
dsp: ast_dsp_silence_noise_with_energy wrong judgment of frame format
(Reported by 周家建)
Music On Hold announcement cuts intro of music the first time it is played
(Reported by Thomas Frederiksen)
func_curl: Segmentation fault when using CURL after setting httpheader CURLOPT
(Reported by Péter Juhász)
RTP Ports not cleared after hangup
(Reported by Ross Beer)
res_stasis: Add compare function for bridges moh container
(Reported by Hajek Michal)
Unable to get rtp codec payload code for slin
(Reported by Brian J. Murrell)
res_pjsip_session: Re-INVITE collisions aren’t handled correctly
(Reported by George Joseph)

Improvements made in this release:
———————————–

Logger: Add debug logging categories
(Reported by Kevin Harwell)
Increase reg_server column size for ps_contacts table realtime
(Reported by sungtae kim)
Create a Bridge with video_single mode
(Reported by sungtae kim)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0

Thank you for your continued support of Asterisk!

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