Asterisk 17.0.0 Now Available!

The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.

This release is available for immediate download at

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

The release of Asterisk 17.0.0 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:

Security bugs fixed in this release:
-----------------------------------

  • [ASTERISK-28495] -
  • res_pjsip_t38: 200 OK with SDP answer with declined stream causes crash
    (Reported by Alexei Gradinari)
  • [ASTERISK-28447] -
  • res_pjsip_messaging: In-dialog MESSAGE with no body causes crash
    (Reported by Gil Richard)
  • [ASTERISK-28465] -
  • Broken SDP can cause a segfault in a T.38 reINVITE
    (Reported by Francesco Castellano)
  • [ASTERISK-28260] -
  • Asterisk segfault when rtp negotiation is wrong or fails
    (Reported by Sotiris Ganouris)
  • [ASTERISK-28127] -
  • Buffer overflow for DNS SRV/NAPTR records
    (Reported by Jan Hoffmann)
  • [ASTERISK-28013] -
  • res_http_websocket: Crash when reading HTTP Upgrade requests
    (Reported by Sean Bright)

    New Features made in this release:
    -----------------------------------

  • [ASTERISK-28403] -
  • Add native Prometheus support to Asterisk
    (Reported by Matt Jordan)
  • [ASTERISK-28375] -
  • res_pjsip: New configuration setting to allow disabling norefersub
    (Reported by Dan Cropp)
  • [ASTERISK-28320] -
  • Added ARI resource /ari/channels/{channelid}/rtp_statistics
    (Reported by sungtae kim)
  • [ASTERISK-28267] -
  • res_stasis: Add ability to switch applications
    (Reported by Benjamin Keith Ford)
  • [ASTERISK-28087] -
  • add flag to allow CALLERID(num) to be placed in Contact header in chan_pjsip
    (Reported by Torrey Searle)
  • [ASTERISK-27971] -
  • res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability
    (Reported by Nick French)

    Improvements made in this release:
    -----------------------------------

  • [ASTERISK-28443] -
  • app_voicemail: remove dependency on stasis cache
    (Reported by Kevin Harwell)
  • [ASTERISK-28442] -
  • stasis_state: Create a stasis module to cache last known state
    (Reported by Kevin Harwell)
  • [ASTERISK-28385] -
  • res_ari_channels: Added detail hangup code settings
    (Reported by sungtae kim)
  • [ASTERISK-28234] -
  • pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
    (Reported by Kirsty Tyerman)
  • [ASTERISK-28401] -
  • app_confbridge: Add *_all remb behavior variants
    (Reported by Joshua C. Colp)
  • [ASTERISK-28400] -
  • res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for transport-cc
    (Reported by Joshua C. Colp)
  • [ASTERISK-28363] -
  • Millisecond-resolution call stats including PDD in channel variables
    (Reported by Antoni Goldstein)
  • [ASTERISK-28378] -
  • Added detail subscriber/subscription info for stasis show app cli
    (Reported by sungtae kim)
  • [ASTERISK-20207] -
  • Asterisk should clear out any .lock files in the voice mail directory on startup.
    (Reported by Steven Wheeler)
  • [ASTERISK-28111] -
  • build: CHANGES/UPGRADE are irritating to work with.
    (Reported by Corey Farrell)
  • [ASTERISK-28264] -
  • Added topic_all container
    (Reported by sungtae kim)
  • [ASTERISK-28343] -
  • Added app_name, app_data to channel type
    (Reported by sungtae kim)
  • [ASTERISK-28326] -
  • ari: Added timestamp for some ari events.
    (Reported by sungtae kim)
  • [ASTERISK-28317] -
  • Add logical group at DAHDIChannel event and create "dahdi_group" at CHANNEL function
    (Reported by Cirillo Ferreira)
  • [ASTERISK-28279] -
  • Added creation timestamp for bridge
    (Reported by sungtae kim)
  • [ASTERISK-27483] -
  • Allow wrapuptime to be set for each queue member
    (Reported by Rodrigo Ramirez Norambuena)
  • [ASTERISK-28055] -
  • app_queue: Per-member wrapup time missing from AddQueueMember application
    (Reported by Niksa Baldun)
  • [ASTERISK-28292] -
  • Changed to show all channel stats including wrong media
    (Reported by sungtae kim)
  • [ASTERISK-28253] -
  • res_pjsip_session: Adding rtcp stats result into the session
    (Reported by sungtae kim)
  • [ASTERISK-28246] -
  • Support skipping on the g726 format
    (Reported by Eyal Hasson)
  • [ASTERISK-28196] -
  • bridge_softmix: Does not support WebRTC source with multi video tracks.
    (Reported by Xiemin Chen)
  • [ASTERISK-28198] -
  • res_ari: Add new hangup causes for ARI Channel DELETE command
    (Reported by Sebastian Damm)
  • [ASTERISK-28144] -
  • [patch] New function PJSIP_PARSE_URI to parse an URI and return a specified part of the URI
    (Reported by Alexei Gradinari)
  • [ASTERISK-28136] -
  • Allow the sip_to_pjsip script to be used in a pipe
    (Reported by Pascal Cadotte Michaud)
  • [ASTERISK-28046] -
  • Remove stale nonoptreq references
    (Reported by Walter Doekes)
  • [ASTERISK-27164] -
  • [patch] Add IPv6 Support for DUNDi
    (Reported by Adam Secombe)
  • [ASTERISK-28006] -
  • PJSIP: Missing "party=calling"/"party=called" in Remote-Party-ID
    (Reported by Eric Dantie)
  • [ASTERISK-27995] -
  • pjproject_bundled: Find shared libraries in root --with-ssl=PATH.
    (Reported by Alexander Traud)
  • [ASTERISK-27993] -
  • pjsip_wizard example gives wrong info about unsupported SRV records
    (Reported by Jonathan Harris)
  • [ASTERISK-27970] -
  • res_rtp_asterisk: T.140 packets containing backspace or end of line are merged with regular text and it causes some UA to break
    (Reported by Emmanuel BUU)

    Bugs fixed in this release:

    -----------------------------------

    Too many to list

    -----------------------------------
    For a full list of changes in this release, please see the ChangeLog:
    https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0

    Thank you for your continued support of Asterisk!

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