[asterisk-dev] Asterisk 16.8.0-rc1 Now Available

[Asterisk Development Team]

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

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
      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 candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0-rc1

Thank you for your continued support of Asterisk!
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