[asterisk-dev] Asterisk 17.6.0-rc1 Now Available
Asterisk Development Team
asteriskteam at digium.com
Thu Jul 9 11:30:25 CDT 2020
The Asterisk Development Team would like to announce the first
release candidate of Asterisk 17.6.0.
This release candidate is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 17.6.0-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:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-28878 - chan_pjsip: PJSIP_MEDIA_OFFER Broken
asterisk 16
(Reported by Joseph Ades)
* ASTERISK-28965 - res_pjsip: Apply outbound proxy to static
contacts on AOR
(Reported by Joshua C. Colp)
* ASTERISK-28930 - ./configure --without-ssl build failure
(Reported by Jaco Kroon)
* ASTERISK-28886 - chan_pjsip: PJSIP_SC_NULL does not exist in
pjproject 2.7.2
(Reported by Jared Smith)
* ASTERISK-28957 - chan_sip: chan_sip does not process 400
response to an INVITE.
(Reported by Frederic LE FOLL)
* ASTERISK-28888 - res_corosync: causes asterisk crash in huge
distributed environment.
(Reported by Universit�� di
Bologna - CESIA VoIP)
* ASTERISK-28955 - "setvar" doesn't work properly in
dahdi-channels.conf
(Reported by Marin Odrljin)
* ASTERISK-28954 - StreamEcho() only returns 1 active stream
(Reported by Bill Kervaski)
* ASTERISK-28953 - res_pjsip_session: Preserve stream label
(Reported by Joshua C. Colp)
* ASTERISK-28942 - res_sorcery_memory_cache: Individual object
expiration behaves unexpectedly with full backend caching
(Reported by Joshua C. Colp)
* ASTERISK-28952 - Queue wrapuptime sometimes not respected
(based on stale lastcall time)
(Reported by Walter Doekes)
* ASTERISK-28950 - Stale code in app_queue to check untouched
channel
(Reported by Walter Doekes)
* ASTERISK-28644 - Stale comment in app_queue about ring_entry
exception
(Reported by Walter Doekes)
* ASTERISK-28938 - core_unreal / core_local: Add support for
multistream and re-negotiation
(Reported by Joshua C.
Colp)
* ASTERISK-28948 - ARI channel create doesn't referencing the
channel_id parameter
(Reported by sungtae kim)
* ASTERISK-28939 - res_rtp_asterisk: Don't have send/receive
buffers on non-WebRTC
(Reported by Joshua C. Colp)
* ASTERISK-28944 - bridge_softmix: Transitioning a stream from
inactive -> sendrecv/sendonly doesn't re-negotiation
(Reported by Joshua C. Colp)
* ASTERISK-28923 - T.38 Segfaults in chan_pjsip_queryoption
(Reported by Yury Kirsanov)
* ASTERISK-28940 - /channels/create doesn't get any parameters
from the body
(Reported by sungtae kim)
* ASTERISK-28936 - res_pjsip: crash when dialing non-sip uri
(Reported by Walter Doekes)
* ASTERISK-28900 - res_fax: Double frame free when gateway in
use with off-nominal format usage
(Reported by Gregory
Massel)
* ASTERISK-28929 - pjproject_bundled: Honor
--without-pjproject.
(Reported by Alexander Traud)
* ASTERISK-28932 - res_pjsip_logger writing too big packets
(Reported by nappsoft)
* ASTERISK-28920 - bridge show all causes crash
(Reported
by sungtae kim)
* ASTERISK-28921 - Wrong return value check for fwrite when
writing to pcap file
(Reported by nappsoft)
Improvements made in this release:
-----------------------------------
* ASTERISK-28959 - res_pjsip: Added option for disable rport
parameter set
(Reported by sungtae kim)
* ASTERISK-28958 - Continue reading string when ping received
by websocket
(Reported by Nickolay V. Shmyrev)
* ASTERISK-28945 - AMI SendText - add Content-Type parameter
(Reported by Kevin Harwell)
* ASTERISK-28949 - res_http_websocket: Add masking to websocket
client
(Reported by Moises Silva)
* ASTERISK-28899 - Upgrade Asterisk to bundled pjproject 2.10
(Reported by Kevin Harwell)
For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0-rc1
Thank you for your continued support of Asterisk!
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