Asterisk 17.2.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 17.2.0.

This release is available for immediate download at

https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 17.2.0 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:

———————————–

CURLOPT() needs a “followlocation” parameter / “maxredirs” doesn’t do anything
(Reported by candrews)
res_pjsip_endpoint_identifier_ip: Add ability to match on source port
(Reported by Sean Bright)

Bugs fixed in this release:

———————————–

stasis application is destroyed after its creation
(Reported by Francois Blackburn)
ARI causes STASIS Deadlock
(Reported by Ross Beer)
REGRESSION: Feature subscription_persistence_recreate (ASTERISK-27759) Causes Segfaults
(Reported by Ross Beer)
CDR billsec is always 0 for transferred calls
(Reported by Maciej Michno)
chan_dahdi: holding a channel via flash to dialtone times out after 0:16:40
(Reported by Andrew Siplas)
silk 24hHz doesn’t show up in ‘core show translation’ output
(Reported by Sean Bright)
Update documentation for statsd module – usage requirements unclear
(Reported by Dan Jenkins)
core: minmemfree watermark uses free RAM, not available RAM
(Reported by Kevin Flyn)
chan_sip: SIP MESSAGE beginning with a whitespace appears empty in the dialplan
(Reported by Frank Matano)
Segfault forwarding voicemail with ODBC storage enabled and realtime voicemail_data is used
(Reported by Stas Kobzar)
empty voicemail.conf required for ARA (realtime) voicemail to leave message
(Reported by Jim Van Meggelen)
Pause reason not reported in QueueMember AMI event
(Reported by Niksa Baldun)
CLI command ‘realtime update2’ syntax failure when using according to usage help
(Reported by Cedric BASSAGET)
res_pjsip_endpoint_identifier_ip: Document support for hostnames
(Reported by Joshua C. Colp)
res_pjsip_notify: Multiple Event headers can be present instead of just one
(Reported by AvayaXAsterisk)
app_record: Lack of `beep` audio file causes application to return error and hangup
(Reported by Corey Farrell)
Wiki docs missing for MessageWaiting
(Reported by David M. Lee)
res_pjsip_pubsub: Subscription persistence does not preserve XML version number
(Reported by Bryan Nelson)
chan_dahdi: Deadlock in Hangup Scenarios with concurrent command pri show span X
(Reported by Dirk Wendland)
stasis bridge topic leak
(Reported by Joeran Vinzens)
pjsip reload not reloading wizard endpoint/pickup_group endpoint/call_group
(Reported by Jean-Denis Girard)
SIP WSS message not processed until next frame arrives
(Reported by Robert Sutton)
contrib: valgrind.supp doesn’t suppress what it’s supposed to due to invalid syntax
(Reported by Richard Kenner)
func_odbc: truncating Unicode string on readsql
(Reported by Boris P. Korzun)
chan_sip: RTP frames not transmitted after emitting a COLP
(Reported by Jean Aunis – Prescom)
Asterisk ignores parsing of config files if a Byte order mark is present
(Reported by Robin Leffmann)
Playback of local files impacted by large media cache
(Reported by Kevin Reeves)
“trustrpid” is misspelled in sip_to_pjsip.py
(Reported by Pascal Cadotte Michaud)
app_chanisavail+cdr: ChanIsAvail sometimes fails to deactivate CDR.
(Reported by Frederic LE FOLL)
app_meetme, chan_ooh323 and cdr_mysql don’t build on 17.0.0
(Reported by George Joseph)
res_pjsip_sdp_rtp: Bundle includes non-existent media stream if codecs create additional streams and offer does not have them
(Reported by nappsoft)
res_fax: wrap Asterisk initiated negotiation with config option
(Reported by Kevin Harwell)
Missing arguments in PJSIP_CONTACT function documentation
(Reported by Pascal Cadotte Michaud)
Memory Leak in res_rtp_asterisk.c
(Reported by Ted G)
chan_sip logs errors on tx to non-existent TCP connections
(Reported by Jaco Kroon)
chan_pjsip incorrectly re-writes REGISTER 200 Response Contact
(Reported by Ross Beer)

Improvements made in this release:

———————————–

Should be able to disable the /httpstatus URI in the built-in HTTP server
(Reported by Sean Bright)
Simplify dialplan for Dial, Page, and ChanIsAvail
(Reported by cmaj)
GET FULL VARIABLE documentation clarification
(Reported by Jonathan Harris)
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-17.2.0

Thank you for your continued support of Asterisk!

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