[asterisk-dev] Asterisk 15.1.0-rc1 Now Available

Asterisk Development Team asteriskteam at digium.com
Fri Oct 13 14:24:20 CDT 2017


The Asterisk Development Team would like to announce the first
release candidate of Asterisk 15.1.0.
This release candidate is available for immediate download at 
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 15.1.0-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:

Improvements made in this release:
-----------------------------------
 * ASTERISK-27278 - [patch] chan_sip: Provide access to read the
      full SIP Request-URI from INVITE
      (Reported by David J.
      Pryke)
 * ASTERISK-27255 - alembic: Add support for Microsoft SQL
      server
      (Reported by Florian Floimair)
 * ASTERISK-27253 - [patch] libsrtp-2.1.x support
     
      (Reported by Alexander Traud)
 * ASTERISK-27220 - Enable CHANNEL function to get from and to
      tag from SIP Headers
      (Reported by Andre Nazario)
 * ASTERISK-27169 - Google OAuth 2.0 support for XMPP / Motif
  
      (Reported by Andrey)
 * ASTERISK-27173 - Support for GMIME 3.0
      (Reported by
      Tzafrir Cohen)
 * ASTERISK-27085 - [patch] chan_pjsip: Port SIPDtmfMode to
      chan_pjsip
      (Reported by Torrey Searle)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-27047 - res_pjsip: user=phone added to Anonymous
      caller-id when it shouldn't be.
      (Reported by dtryba)
 * ASTERISK-26988 - res_pjsip_session: user_eq_phone adds double
      user=phone parameters to URIs
      (Reported by dtryba)
 * ASTERISK-27270 - cdr_mysql: various crashes at second module
      reload if cdr_mysql.conf is configured
      (Reported by
      Tzafrir Cohen)
 * ASTERISK-25266 - Application Originate returns SUCCESS to
      ORIGINATE_STATUS upon failure to originate
      (Reported by
      Allen Ford)
 * ASTERISK-27192 - res_pjsip: Loss of SIP registrations causing
      unavailable endpoints
      (Reported by Richard Mudgett)
 * ASTERISK-27305 - res_ari: Memory leaks in ARI when using
      Content-Type: application/json
      (Reported by David Hajek)
 * ASTERISK-26922 - chan_sip: tcpbind uses wrong source address

      (Reported by Ksenia)
 * ASTERISK-27324 - [patch] Dual-Stack server cannot be used as
      IPv4 client via TCP/TLS
      (Reported by Alexander Traud)
 * ASTERISK-27317 - vector: multiple evaluation of elem in
      AST_VECTOR_ADD_SORTED.
      (Reported by Corey Farrell)
 * ASTERISK-27301 - [patch] app_queue: Music On Hold for
      real-time queues is not reset to default
      (Reported by
      Nathan Bruning)
 * ASTERISK-27318 - res_pjsip_mwi: uninitialized value from
      ast_strings_match
      (Reported by Corey Farrell)
 * ASTERISK-27284 - Status of RFC 3323 and PJSIP
      (Reported
      by dtryba)
 * ASTERISK-27296 - [patch] False positive busy checks when
      icalendar's recurrence-id mechanism is involved
      (Reported
      by Benoît Dereck-Tricot)
 * ASTERISK-27216 - app_queue: does its
      check-makeannouncement-logic twice each head-caller-loop
     
      (Reported by Stefan Engström)
 * ASTERISK-27298 - Problem with expires on pjsip /
      outbound-publish
      (Reported by Cyrille Demaret)
 * ASTERISK-27295 - Contact is improperly translated after
      d178f497
      (Reported by Sean Bright)
 * ASTERISK-27292 - Multiple RTP Stream Created Breaking RFC2833
      (SSRC Changes)
      (Reported by Ross Beer)
 * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
      configured codecs, but subset based on caller
      (Reported by
      lvl)
 * ASTERISK-27289 - A codeblock that maintains a bug,but maybe
      the codeblock will never run
      (Reported by Huangyx)
 * ASTERISK-27277 - bridge: Renegotiate if source stream
      changes.
      (Reported by Joshua Colp)
 * ASTERISK-27283 - Realtime config fail with PostgreSQL version
      before 9.1
      (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-27264 - res_pjsip_session: Crashes after sending
      PRACK and receiving 200 OK
      (Reported by Daniel Heckl)
 * ASTERISK-27260 - [pjsip] chan_pjsip_indicate: Don't know how
      to indicate condition 36
      (Reported by Daniel Heckl)
 * ASTERISK-27257 - bridge_native_rtp: half-way direct media
      when using early bridging
      (Reported by Jean Aunis -
      Prescom)
 * ASTERISK-16898 - SRTP unprotect: authentication failure when
      RTP sequence number switches from 65535 -> 0
      (Reported by
      Marcello Ceschia)
 * ASTERISK-27274 - RTCP needs better packet validation to
      resist port scans.
      (Reported by Richard Mudgett)
 * ASTERISK-27252 - RTP: One way audio with direct media and
      strictrtp=yes.
      (Reported by Richard Mudgett)
 * ASTERISK-27279 - Crash in pubsub_on_rx_request NULL pointer -
      Possible PJSIP Vulnerability
      (Reported by Ross Beer)
 * ASTERISK-25524 - module reload res_calendar.so does not
      reload everything in calendar.conf
      (Reported by Jesper)
 * ASTERISK-24588 - res_calendar does not process CalDAV from
      Owncloud [fix included]
      (Reported by Stefan Gofferje)
 * ASTERISK-25523 - res_calendar: Warning about invalid channel
      value (for notification) occurs even when event has no
      notification configured.
      (Reported by Jesper)
 * ASTERISK-21399 - RTP Multicast of L16 (type 10): Asterisk and
      wireshark disagree
      (Reported by Tzafrir Cohen)
 * ASTERISK-27248 - [patch]external_media_address and
      external_signaling_address don't always honor localnet
     
      (Reported by Walter Doekes)
 * ASTERISK-24066 - res_smdi: convert to astobj2
      (Reported
      by Corey Farrell)
 * ASTERISK-27217 - chan_sip: Asterisk crashing when
      subscription doesn't get set
      (Reported by Bryan Walters)
 * ASTERISK-17540 - SDP origin attribute modified when issuing
      re-INVITE because of directmedia=yes
      (Reported by saghul)
 * ASTERISK-27165 - CDR: CDR(start,u) function won't work in
      cdr_custom config
      (Reported by Jacek Konieczny)
 * ASTERISK-27254 - alembic: prune_on_boot fix erroneous
     
      (Reported by Florian Floimair)
 * ASTERISK-27232 - When in queue on g722 with interruptions,
      music on hold can get stuck and no longer play
      (Reported
      by Jens T.)
 * ASTERISK-27024 - nat/external_media settings ignored in
      14.4.1
      (Reported by Christopher van de Sande)
 * ASTERISK-26879 - PJSIP external_media_address ignored if no
      local_net options are provided
      (Reported by Matt Jordan)
 * ASTERISK-27236 - Segfault ast_channel_name (chan=0x0) at
      channel_internal_api.c:478 during T.38 Fax Receive
     
      (Reported by Ross Beer)
 * ASTERISK-27225 - Crash when freeing dtls_cfg->cafile
     
      (Reported by Richard Kenner)
 * ASTERISK-27177 - ooh323c: misleading indentation in
      addons/ooh323c/src/ooSocket.c
      (Reported by Tzafrir Cohen)
 * ASTERISK-27241 - libc segfault upon entry into app_directory

      (Reported by David Moore)
 * ASTERISK-27152 - Sending a "tel" uri in a From or To header
      in an unauthenticated message causes asterisk to crash
     
      (Reported by Ross Beer)
 * ASTERISK-27103 - core: ast_safe_system command injection
      possible.
      (Reported by Corey Farrell)
 * ASTERISK-27013 - res_rtp_asterisk: Media can be hijacked even
      with strict RTP enabled
      (Reported by Joshua Colp)
 * ASTERISK-27231 - res_rtp_asterisk: Allow remote SSRC to
      change due to renegotiation
      (Reported by Joshua Colp)
 * ASTERISK-26994 - Confbridge: CBAnn channels intermittently
      become stuck when caller hangs up before recording name
     
      (Reported by James Terhune)
 * ASTERISK-27222 - core: Don't queue up multiple video update
      frames.
      (Reported by Joshua Colp)
 * ASTERISK-20858 - app_minivm fails to clean up mkstemp files
 
      (Reported by Walter Doekes)
 * ASTERISK-16777 - several filename bugs in Record()
      application
      (Reported by klaus3000)
 * ASTERISK-27168 - alembic: PJSIP scripts are missing column
      dtls_fingerprint in ps_endpoints table
      (Reported by
      Florian Floimair)
 * ASTERISK-27209 - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE
      is used
      (Reported by Torrey Searle)
 * ASTERISK-19103 - When using realtime queues, function
      QUEUE_MEMBER_LIST() will return an error if no other
      app/function has loaded the queues first. This problem does not
      exist if queues.conf is used.
      (Reported by Jim Van
      Meggelen)
 * ASTERISK-21241 - When using voicemail as announce only
      (maxmsg=0), the star dtmf to enter the voicemail is not honored

      (Reported by Eelco Brolman)
 * ASTERISK-27212 - bridge_softmix: Quickly joining/leaving may
      cause video stream to remain in SFU
      (Reported by Richard
      Mudgett)
 * ASTERISK-27204 - [patch] app_queue: Wrong queue stat
      calculation
      (Reported by sungtae kim)
 * ASTERISK-27207 - XMPP OAuth not working due to inverted
      logic
      (Reported by Michael Kuron)
 * ASTERISK-27174 - res_calendar_icalendar: Recurring events not
      being loaded from Google calendar using ical
      (Reported by
      Mark Thompson)
 * ASTERISK-27202 - If wget is not installed and "or" is not
      available, external components (excluding pjsip) are not
      installed
      (Reported by Seán C. McCord)
 * ASTERISK-27200 - manager: hook event is not being raised
    
      (Reported by Kevin Harwell)
 * ASTERISK-27147 - Either asterisk or pjproject isn't re-using
      tcp connections (again)
      (Reported by George Joseph)
 * ASTERISK-27193 - IPv6 receive address in message doesn't
      include brackets
      (Reported by Scott Griepentrog)
 * ASTERISK-27158 - [patch] res_rtp_asterisk: RTCP statistics
      are not available when native bridge is used
      (Reported by
      Torrey Searle)
 * ASTERISK-26745 - Asymmetric codecs when
      asymmetric_rtp_codec=no
      (Reported by Jesse Ross)
 * ASTERISK-27189 - Make --with-pjproject-bundled the default
      for Asterisk 15
      (Reported by George Joseph)
 * ASTERISK-27110 - RTP session is not fully destroyed on
      channel hangup
      (Reported by Matt Jordan)
 * ASTERISK-27182 - bridge: Crash when mapping streams
     
      (Reported by Joshua Colp)
 * ASTERISK-27180 - channel: requester leaks joint_cap on
      success.
      (Reported by Corey Farrell)
 * ASTERISK-27179 - res_pjsip_session: Handling of 'msid' is
      incorrect
      (Reported by Kevin Harwell)
 * ASTERISK-27119 - res_pjsip: parse/add msid attribute when
      webrtc is enabled
      (Reported by Kevin Harwell)
 * ASTERISK-27171 - Asterisk 15.0.0-Beta1 does not compile
     
      (Reported by Ira Emus)
 * ASTERISK-26659 - res_pjsip: PJSIP presence - missing braces
      around the status element in XML
      (Reported by Abraham
      Liebsch)
 * ASTERISK-27156 - Asterisk won't compile on Fedora 26 with
      devmode enabled.
      (Reported by Corey Farrell)
 * ASTERISK-27001 - res_pjsip: TLS connection not stable
     
      (Reported by Ian Gilmour)

New Features made in this release:
-----------------------------------
 * ASTERISK-27215 - [patch]AMI : Add CancelAtxfer Action
     
      (Reported by Thomas Sevestre)

For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0-rc1

Thank you for your continued support of Asterisk!
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