Log Message:
comms/asterisk16: Update to 16.17.0

This is a micro update that is mostly security fixes and bug fixes
with very small improvements.  In addition to this being a security
fix, asterisk16 is a leaf package.

Upstream changes:

Security bugs fixed in this release:
-----------------------------------
 * ASTERISK-29305 - ASTERISK-29203 / AST-2021-002 -- Another
      scenario is causing a crash
      (Reported by Gregory Massel)
 * ASTERISK-29260 - sRTP Replay Protection ignored; even tears
      down long calls
      (Reported by Alexander Traud)
 * ASTERISK-29227 - res_pjsip_diversion: sending multiple 181
      responses causes memory corruption and crash
      (Reported by
      Ivan Poddubny)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-29215 - res_pjsip_session: NULL active_media_state
      topology caused asterisk crash
      (Reported by sungtae kim)
 * ASTERISK-29035 - chan_local: Multistream support breaks T.38
      faxing
      (Reported by Matthias Hensler)
 * ASTERISK-29071 - app_confbridge: Memory rises when
      jitterbuffer enabled and muting over AMI occurs
      (Reported
      by Stefan Ruf)
 * ASTERISK-29329 - app_dial: DTMF to 'D' option gets duplicated
      if there are multiple progress events
      (Reported by N A)
 * ASTERISK-24434 - Fix differing usage of assignment operators
      in modules.conf
      (Reported by Rusty Newton)
 * ASTERISK-29306 - strings: Incorrect use of
      __attribute__((pure)) in ast_str_to_lower definition

      (Reported by Vitezslav Novy)
 * ASTERISK-29300 - res_rtp_asterisk: When native local bridging
      the remote SSRC becomes permanent
      (Reported by Sebastian
      Damm)
 * ASTERISK-29235 - res_pjsip_nat: Contact is rewritten on
      REGISTER responses with external_signaling_address

      (Reported by Brian Paboojian)
 * ASTERISK-29266 - ICE Role conflict with an unauthorized
      session
      (Reported by Salah Ahmed)
 * ASTERISK-29105 - chan_pjsip: 180 Ringing with SDP not changed
      into progress
      (Reported by Sebastian Damm)
 * ASTERISK-29297 - say: Y2021 problem – Asterisk cannot say
      year 2021 in Dutch
      (Reported by Jacek Konieczny)
 * ASTERISK-29315 - res_pjsip: re-registration gets stuck if
      setting initial auth credentials fails
      (Reported by Nick
      French)
 * ASTERISK-29312 - res_fax: asterisk fails to publish the
      Stasis and ReceiveFax status messages if the remote Station ID
      contains invalid UTF-8 characters
      (Reported by Alexei
      Gradinari)
 * ASTERISK-16799 - Callee declined when 'beep' audio file does
      not exist
      (Reported by IAMJames_)
 * ASTERISK-29313 - res_pjsip_refer:  Segfault in progress
      notify
      (Reported by George Joseph)
 * ASTERISK-29293 - res_config_pgsql: Limit realtime_pgsql() to
      return one (no more) record
      (Reported by Boris P. Korzun)
 * ASTERISK-29303 - pjsip: Re-invite occurs when it shouldn't

      (Reported by Benjamin Keith Ford)
 * ASTERISK-29311 - res_odbc_transaction sets forcecommit
      default value based on isolation level instead of forcecommit

      (Reported by Jaco Kroon)
 * ASTERISK-28452 - pjsip: <sess-version> of SDP is not
      incremented though SDP may be changed on reinvite without SDP
      offer
      (Reported by Michael Maier)
 * ASTERISK-29287 - app.h: C++ compatibility broken

      (Reported by Jean Aunis - Prescom)
 * ASTERISK-28369 - app_queue: Member device state "invalid"
      when second call is ringing and hint is used
      (Reported by
      Boolah )
 * ASTERISK-29203 - res_pjsip_t38: Crash when changing state

      (Reported by Gregory Massel)
 * ASTERISK-29205 - res_rtp_asterisk: Asterisk crashes when
      making hold/unhold from webrtc client
      (Reported by Edvin
      Vidmar)
 * ASTERISK-29196 - res_pjsip: Segmentation fault

      (Reported by Mauri de Souza Meneguzzo (3CPlus))
 * ASTERISK-29280 - chan_sip: Allow peers without audio
      (text+video).
      (Reported by Alexander Traud)
 * ASTERISK-29265 - chan_sip: Allow text+video media streams,
      again.
      (Reported by Alexander Traud)
 * ASTERISK-29261 - res_pjsip: user=phone validation fail for
      isup numbers containing *#
      (Reported by Mark Petersen)
 * ASTERISK-29259 - channel: Allow text+video media streams,
      again.
      (Reported by Alexander Traud)
 * ASTERISK-29258 - chan_sip: Audio stream rejected, Other
      stream present: Invalid SDP.
      (Reported by Alexander Traud)
 * ASTERISK-29220 - After T38 reinvite response of 488 a
      subsequent G711 reinvite is not processed correctly. Instead the
      previous T38 session media is used
      (Reported by Robert
      Cripps)
 * ASTERISK-29248 - res_pjsip_session: res sometimes
      uninitialized reported by compiler Clang.
      (Reported by
      Alexander Traud)

Improvements made in this release:
-----------------------------------
 * ASTERISK-29321 - sorcery: Add support for more intelligent
      reloading.
      (Reported by Joshua C. Colp)
 * ASTERISK-29325 - res_pjsip_registrar: Include source IP
      address and port in log messages
      (Reported by Joshua C.
      Colp)
 * ASTERISK-29326 - asterisk: Update copyright/company

      (Reported by Joshua C. Colp)
 * ASTERISK-29244 - Add MixMonitorStart / Stop / Mute AMI
      events
      (Reported by Sébastien Duthil)
 * ASTERISK-29275 - Support of MIME-type for wav16

      (Reported by Boris P. Korzun)
 * ASTERISK-29252 - TRANSFERSTATUSPROTOCOL variable to report
      Transfer (REFER) failure SIP code
      (Reported by Dan Cropp)
 * ASTERISK-29262 - Support of various URL-schemes by MoH

      (Reported by Boris P. Korzun)