[asterisk-dev] Asterisk 16.8.0-rc1 Now Available

Asterisk Development Team asteriskteam at digium.com
Thu Jan 23 12:01:46 CST 2020

The Asterisk Development Team would like to announce the first
release candidate of Asterisk 16.8.0.
This release candidate is available for immediate download at 

The release of Asterisk 16.8.0-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:

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-28677 - CDR billsec is always 0 for transferred
      (Reported by Maciej Michno)
 * ASTERISK-28702 - chan_dahdi: holding a channel via flash to
      dialtone times out after 0:16:40
      (Reported by Andrew
 * 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
 * ASTERISK-28349 - Pause reason not reported in QueueMember AMI
      (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
 * ASTERISK-28682 - app_record: Lack of `beep` audio file causes
      application to return error and hangup
      (Reported by Corey
 * 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
 * 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
      (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
      (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
      (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
      (Reported by cmaj)
 * ASTERISK-28673 - GET FULL VARIABLE documentation
      (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 candidate, please see the ChangeLog:

Thank you for your continued support of Asterisk!
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