[asterisk-announce] Asterisk 15.1.0 Now Available

Asterisk Development Team asteriskteam at digium.com
Mon Oct 30 13:50:45 CDT 2017

The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
This release is available for immediate download at

The release of Asterisk 15.1.0 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:

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.
 * ASTERISK-27255 - alembic: Add support for Microsoft SQL
      (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
      (Reported by Torrey Searle)

Bugs fixed in this release:
 * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
      curl is loaded
      (Reported by Ronald Raikes)
 * ASTERISK-27372 - ARI: Node ARI client broken in latest
      versions of 13 and 14
      (Reported by Benjamin Keith Ford)
 * 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
      (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
      (Reported by Corey Farrell)
 * ASTERISK-27284 - Status of RFC 3323 and PJSIP
      by dtryba)
 * ASTERISK-27296 - [patch] False positive busy checks when
      icalendar's recurrence-id mechanism is involved
      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 /
      (Reported by Cyrille Demaret)
 * ASTERISK-27295 - Contact is improperly translated after
      (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
 * 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
      (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 -
 * 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
      (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
      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
      by Jens T.)
 * ASTERISK-27024 - nat/external_media settings ignored in
      (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
      (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
      (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
      (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()
      (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
 * 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
 * ASTERISK-27204 - [patch] app_queue: Wrong queue stat
      (Reported by sungtae kim)
 * ASTERISK-27207 - XMPP OAuth not working due to inverted
      (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
      (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
      (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
      (Reported by Corey Farrell)
 * ASTERISK-27179 - res_pjsip_session: Handling of 'msid' is
      (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
 * ASTERISK-27156 - Asterisk won't compile on Fedora 26 with
      devmode enabled.
      (Reported by Corey Farrell)

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, please see the ChangeLog:

Thank you for your continued support of Asterisk!
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