Asterisk Development Team
2020-Feb-04 16:34 UTC
[asterisk-users] Asterisk 16.8.0 Now Available
The Asterisk Development Team would like to announce the release of Asterisk
16.8.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 16.8.0 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:
New Features made in this release:
-----------------------------------
* ASTERISK-17491 - CURLOPT() needs a "followlocation" parameter
/ "maxredirs" doesn't do anything
(Reported by candrews)
* ASTERISK-28639 - res_pjsip_endpoint_identifier_ip: Add
ability to match on source port
(Reported by Sean Bright)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-28679 - stasis application is destroyed after its
creation
(Reported by Francois Blackburn)
* ASTERISK-28423 - ARI causes STASIS Deadlock
(Reported
by Ross Beer)
* ASTERISK-28714 - REGRESSION: Feature
subscription_persistence_recreate (ASTERISK-27759) Causes
Segfaults
(Reported by Ross Beer)
* ASTERISK-28677 - CDR billsec is always 0 for transferred
calls
(Reported by Maciej Michno)
* ASTERISK-28702 - chan_dahdi: holding a channel via flash to
dialtone times out after 0:16:40
(Reported by Andrew
Siplas)
* ASTERISK-28706 - silk 24hHz doesn't show up in 'core show
translation' output
(Reported by Sean Bright)
* ASTERISK-24484 - Update documentation for statsd module -
usage requirements unclear
(Reported by Dan Jenkins)
* ASTERISK-28695 - core: minmemfree watermark uses free RAM,
not available RAM
(Reported by Kevin Flyn)
* ASTERISK-28693 - chan_sip: SIP MESSAGE beginning with a
whitespace appears empty in the dialplan
(Reported by
Frank Matano)
* ASTERISK-23739 - [patch]Segfault forwarding voicemail with
ODBC storage enabled and realtime voicemail_data is used
(Reported by Stas Kobzar)
* ASTERISK-27622 - empty voicemail.conf required for ARA
(realtime) voicemail to leave message
(Reported by Jim Van
Meggelen)
* ASTERISK-28349 - Pause reason not reported in QueueMember AMI
event
(Reported by Niksa Baldun)
* ASTERISK-21794 - CLI command 'realtime update2' syntax
failure when using according to usage help
(Reported by
Cedric BASSAGET)
* ASTERISK-25429 - res_pjsip_endpoint_identifier_ip: Document
support for hostnames
(Reported by Joshua C. Colp)
* ASTERISK-27775 - res_pjsip_notify: Multiple Event headers can
be present instead of just one
(Reported by
AvayaXAsterisk)
* ASTERISK-28682 - app_record: Lack of `beep` audio file causes
application to return error and hangup
(Reported by Corey
Farrell)
* ASTERISK-28507 - Wiki docs missing for MessageWaiting
(Reported by David M. Lee)
* ASTERISK-27759 - res_pjsip_pubsub: Subscription persistence
does not preserve XML <dialog-info> version number
(Reported by Bryan Nelson)
* ASTERISK-28605 - chan_dahdi: Deadlock in Hangup Scenarios
with concurrent command pri show span X
(Reported by Dirk
Wendland)
* ASTERISK-28633 - stasis bridge topic leak
(Reported by
Joeran Vinzens)
* ASTERISK-28492 - pjsip reload not reloading wizard
endpoint/pickup_group endpoint/call_group
(Reported by
Jean-Denis Girard)
* ASTERISK-28562 - SIP WSS message not processed until next
frame arrives
(Reported by Robert Sutton)
* ASTERISK-27243 - contrib: valgrind.supp doesn't suppress what
it's supposed to due to invalid syntax
(Reported by
Richard Kenner)
* ASTERISK-28497 - func_odbc: truncating Unicode string on
readsql
(Reported by Boris P. Korzun)
* ASTERISK-28647 - chan_sip: RTP frames not transmitted after
emitting a COLP
(Reported by Jean Aunis - Prescom)
* ASTERISK-28667 - Asterisk ignores parsing of config files if
a Byte order mark is present
(Reported by Robin Leffmann)
* ASTERISK-28664 - "trustrpid" is misspelled in
sip_to_pjsip.py
(Reported by Pascal Cadotte Michaud)
* ASTERISK-28604 - app_meetme, chan_ooh323 and cdr_mysql don't
build on 17.0.0
(Reported by George Joseph)
* ASTERISK-28659 - res_pjsip_sdp_rtp: Bundle includes
non-existent media stream if codecs create additional streams
and offer does not have them
(Reported by nappsoft)
* ASTERISK-28660 - res_fax: wrap Asterisk initiated negotiation
with config option
(Reported by Kevin Harwell)
* ASTERISK-28636 - app_chanisavail+cdr: ChanIsAvail sometimes
fails to deactivate CDR.
(Reported by Frederic LE FOLL)
* ASTERISK-28626 - Missing arguments in PJSIP_CONTACT function
documentation
(Reported by Pascal Cadotte Michaud)
* ASTERISK-28609 - Memory Leak in res_rtp_asterisk.c
(Reported by Ted G)
* ASTERISK-28651 - chan_sip logs errors on tx to non-existent
TCP connections
(Reported by Jaco Kroon)
* ASTERISK-28502 - chan_pjsip incorrectly re-writes REGISTER
200 Response Contact
(Reported by Ross Beer)
* ASTERISK-28625 - Playback of local files impacted by large
media cache
(Reported by Kevin Reeves)
Improvements made in this release:
-----------------------------------
* ASTERISK-28710 - Should be able to disable the /httpstatus
URI in the built-in HTTP server
(Reported by Sean Bright)
* ASTERISK-28638 - Simplify dialplan for Dial, Page, and
ChanIsAvail
(Reported by cmaj)
* ASTERISK-28673 - GET FULL VARIABLE documentation
clarification
(Reported by Jonathan Harris)
* ASTERISK-28658 - app_confbridge: Add support for setting
maximum sample rate
(Reported by Joshua C. Colp)
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
Thank you for your continued support of Asterisk!
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