[asterisk-dev] Asterisk 17.6.0 Now Available

[Asterisk Development Team]

The Asterisk Development Team would like to announce the release of Asterisk 17.6.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 17.6.0 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:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-28878 - chan_pjsip: PJSIP_MEDIA_OFFER Broken
      asterisk 16
      (Reported by Joseph Ades)
 * ASTERISK-28965 - res_pjsip: Apply outbound proxy to static
      contacts on AOR
      (Reported by Joshua C. Colp)
 * ASTERISK-28930 - ./configure --without-ssl build failure
    
      (Reported by Jaco Kroon)
 * ASTERISK-28886 - chan_pjsip: PJSIP_SC_NULL does not exist in
      pjproject 2.7.2
      (Reported by Jared Smith)
 * ASTERISK-28957 - chan_sip: chan_sip does not process 400
      response to an INVITE.
      (Reported by Frederic LE FOLL)
 * ASTERISK-28888 - res_corosync: causes asterisk crash in huge
      distributed environment.
      (Reported by Universit�� di
      Bologna - CESIA VoIP)
 * ASTERISK-28955 - "setvar" doesn't work properly in
      dahdi-channels.conf
      (Reported by Marin Odrljin)
 * ASTERISK-28954 - StreamEcho() only returns 1 active stream
  
      (Reported by Bill Kervaski)
 * ASTERISK-28953 - res_pjsip_session: Preserve stream label
   
      (Reported by Joshua C. Colp)
 * ASTERISK-28942 - res_sorcery_memory_cache: Individual object
      expiration behaves unexpectedly with full backend caching
     
      (Reported by Joshua C. Colp)
 * ASTERISK-28952 - Queue wrapuptime sometimes not respected
      (based on stale lastcall time)
      (Reported by Walter Doekes)
 * ASTERISK-28950 - Stale code in app_queue to check untouched
      channel
      (Reported by Walter Doekes)
 * ASTERISK-28644 - Stale comment in app_queue about ring_entry
      exception
      (Reported by Walter Doekes)
 * ASTERISK-28938 - core_unreal / core_local: Add support for
      multistream and re-negotiation
      (Reported by Joshua C.
      Colp)
 * ASTERISK-28948 - ARI channel create doesn't referencing the
      channel_id parameter
      (Reported by sungtae kim)
 * ASTERISK-28939 - res_rtp_asterisk: Don't have send/receive
      buffers on non-WebRTC
      (Reported by Joshua C. Colp)
 * ASTERISK-28944 - bridge_softmix: Transitioning a stream from
      inactive -> sendrecv/sendonly doesn't re-negotiation
     
      (Reported by Joshua C. Colp)
 * ASTERISK-28923 - T.38 Segfaults in chan_pjsip_queryoption
   
      (Reported by Yury Kirsanov)
 * ASTERISK-28940 - /channels/create doesn't get any parameters
      from the body
      (Reported by sungtae kim)
 * ASTERISK-28936 - res_pjsip: crash when dialing non-sip uri
  
      (Reported by Walter Doekes)
 * ASTERISK-28900 - res_fax: Double frame free when gateway in
      use with off-nominal format usage
      (Reported by Gregory
      Massel)
 * ASTERISK-28929 - pjproject_bundled: Honor
      --without-pjproject.
      (Reported by Alexander Traud)
 * ASTERISK-28932 - res_pjsip_logger writing too big packets
   
      (Reported by nappsoft)
 * ASTERISK-28920 - bridge show all causes crash
      (Reported
      by sungtae kim)
 * ASTERISK-28921 - Wrong return value check for fwrite when
      writing to pcap file
      (Reported by nappsoft)

Improvements made in this release:
-----------------------------------
 * ASTERISK-28959 - res_pjsip: Added option for disable rport
      parameter set
      (Reported by sungtae kim)
 * ASTERISK-28958 - Continue reading string when ping received
      by websocket
      (Reported by Nickolay V. Shmyrev)
 * ASTERISK-28945 - AMI SendText - add Content-Type parameter
  
      (Reported by Kevin Harwell)
 * ASTERISK-28949 - res_http_websocket: Add masking to websocket
      client
      (Reported by Moises Silva)
 * ASTERISK-28899 - Upgrade Asterisk to bundled pjproject 2.10
 
      (Reported by Kevin Harwell)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0

Thank you for your continued support of Asterisk!
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