[asterisk-dev] Asterisk 13.5.0-rc1 Now Available

Asterisk Development Team asteriskteam at digium.com
Mon Jul 27 13:03:50 CDT 2015

The Asterisk Development Team has announced the first release candidate of
Asterisk 13.5.0. This release candidate is available for immediate
download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.5.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release candidate:

Improvements made in this release:
 * ASTERISK-25256 - [patch]Post AMI VarSet to empty string events
      when Asterisk deletes a dialplan variable. (Reported by Richard
 * ASTERISK-25067 - Sorcery Caching: Implement a new caching module
      (Reported by Matt Jordan)
 * ASTERISK-25040 - pbx: Improve performance of reloads by making
      hint destruction more performant (Reported by Matt Jordan)
 * ASTERISK-25114 - res_pjsip:  Add AMI events for chan_pjsip
      contact lifecycle changes (Reported by George Joseph)
 * ASTERISK-25072 - res_pjsip_outbound_registration: line
      functionality. Additional check for using the request URI
      (Reported by Dmitriy Serov)

Bugs fixed in this release:
 * ASTERISK-25250 - chan_sip - Despite the channel being answered,
      caller on a call established via Local channel continues to hear
      ringback (Reported by Etienne Lessard)
 * ASTERISK-25253 - confbridge volume options and other volume
      controls such as func_volume don't work (Reported by Dmitriy
 * ASTERISK-25247 - choppy audio when spying on a g722 channel,
      chan_sip or chan_pjsip (Reported by hristo)
 * ASTERISK-24867 - Docs for 'e' option in ResetCDR say to use
      CDR_PROP instead, CDR_PROP docs are unclear (Reported by Rusty
 * ASTERISK-24853 - Documentation claims chan_sip outbound
      registrations support WS or WSS as valid transports (not true)
      (Reported by PSDK)
 * ASTERISK-25242 - PJSIP: No audio when Asterisk inside NAT and
      endpoints outside NAT - implement functionality similar to
      chan_sip 'rtpkeepalive'? (Reported by Mark Michelson)
 * ASTERISK-25258 - chan_pjsip: Incorrect format switch on received
      RTP packet (Reported by Joshua Colp)
 * ASTERISK-25257 - [patch]channels/sig_pri.h -> sig_pri_span ->
      force_restart_unavailable_chans in wrong scope (Reported by
      Patric Marschall)
 * ASTERISK-24934 - [patch]Asterisk manager output does not escape
      control characters (Reported by warren smith)
 * ASTERISK-25255 - Missing AMI VarSet events when setting to an
      empty string. (Reported by Richard Mudgett)
 * ASTERISK-25254 - Crash if dialplan sets ATTENDEDTRANSFER to an
      empty string before Park. (Reported by Richard Mudgett)
 * ASTERISK-25183 - PJSIP: Crash on NULL channel in
      chan_pjsip_incoming_response despite previous checks for NULL
      channel (Reported by Matt Jordan)
 * ASTERISK-25201 - Crash in PJSIP distributor on already free'd
      threadpool (Reported by Matt Jordan)
 * ASTERISK-24782 - StasisEnd event not present for channel that
      was swapped out for another after completing attended transfer
      (Reported by John Bigelow)
 * ASTERISK-25240 - bridge_native_rtp: Direct media wrongfully
      started when completing attended transfer (Reported by Joshua
 * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
      (Reported by Rusty Newton)
 * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
      BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
      (Reported by Dmitry Burilov)
 * ASTERISK-24550 - res_rtp_asterisk: Crash in
      ast_rtp_on_ice_complete during DTLS handshake (Reported by
      Osaulenko Alexander)
 * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
      Badalian Vyacheslav)
 * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
      by Stefan Engström)
 * ASTERISK-25127 - DTLS crashes following "Unable to cancel
      schedule ID" in dtls_srtp_check_pending (Reported by Dade
 * ASTERISK-25168 - Random Core Dumps on Asterisk 13.4 PJSIP, in
      ast_channel_name at channel_internal_api.c (Reported by Carl
 * ASTERISK-25115 - Crash related to func
      sip_resolve_invoke_user_callback of res_pjsip/pjsip_resolver.c
      (Reported by John Bigelow)
 * ASTERISK-25226 - chan_sip: Channel leak in branch 13 on early
      replaces call pickup (Reported by Walter Doekes)
 * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
      (Reported by Walter Doekes)
 * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
      in rtp_engine.c (Reported by Walter Doekes)
 * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
      (Reported by Walter Doekes)
 * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
      Bad file descriptor" (Reported by Barry Chern)
 * ASTERISK-25165 - Testsuite - Sorcery memory cache leaks
      (Reported by Corey Farrell)
 * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
      13.4 (Reported by cervajs)
 * ASTERISK-25196 - res_pjsip_nat: rewrite_contact should not be
      applied to Contact header when Record-Route headers are present
      (Reported by Mark Michelson)
 * ASTERISK-24907 - res_pjsip_outbound_registration: crash during
      unload if registration attempts are still occuring (Reported by
      Kevin Harwell)
 * ASTERISK-25204 - res_pjsip_refer: Duplicated Referred-By or
      Replaces headers on outbound INVITEs. (Reported by Mark
 * ASTERISK-25171 - Early completion of feature code attended
      transfer results in intermittent one-way audio, "ghost ringing"
      and robotic sound. (Reported by Rusty Newton)
 * ASTERISK-25189 - AMI: Add Linkedid header to standard channel
      snapshot information. (Reported by Richard Mudgett)
 * ASTERISK-25172 - Crash in channels/sip/sip blind
      transfer/caller_refer_only test in
      ast_format_cap_append_from_cap during ast_request (Reported by
      Matt Jordan)
 * ASTERISK-25180 - res_pjsip_mwi: Unsolicited MWI requires reload
      (Reported by Joshua Colp)
 * ASTERISK-25182 - [patch] on CLI sip reload, new codecs get
      appended only (Reported by Alexander Traud)
 * ASTERISK-25163 - Deadlock in chan_sip between reload of sip peer
      container and MWI Stasis callback (Reported by Dmitriy Serov)
 * ASTERISK-25091 - Asterisk REST API - bridge.addChannel crash
      asterisk when calling channel hangup while adding to bridge
      (Reported by Ilya Trikoz)
 * ASTERISK-24900 - Manager event ParkedCallSwap is not documented
      (Reported by Rusty Newton)
 * ASTERISK-25162 - func_pjsip_aor: Leak of contact in iterator
      (Reported by Corey Farrell)
 * ASTERISK-25158 - res_pjsip: Add option to use AAL2 packing when
      negotiating g.726 (Reported by Kevin Harwell)
 * ASTERISK-24344 - CDR_PROP(disable) disables CDR only for first
      dialed party (Reported by Janusz Karolak)
 * ASTERISK-24443 - CDR fields (dst, dcontext) empty in transfer
      call started from Macro (Reported by Arveno Santoro)
 * ASTERISK-25154 - [patch]fromtag may need to be updated after
      successful call dialog match (Reported by Damian Ivereigh)
 * ASTERISK-25156 - chan_pjsip’s CHAN_START cel event lacks the
      correct context and exten (Reported by cloos)
 * ASTERISK-25157 - bridging: Performing a blonde transfer does not
      result in connected line updates (Reported by Joshua Colp)
 * ASTERISK-25087 - Asterisk segfault when using Directory
      application with alias option and specific mailbox configuration
      (Reported by Chet Stevens)
 * ASTERISK-24983 - IAX deadlock between hangup and scheduled
      actions (ex. largrq) (Reported by Y Ateya)
 * ASTERISK-25096 - [patch]Segfault when registering over
      websockets with PJSIP (in ast_sockaddr_isnull at
      /include/asterisk/netsock2.h) (Reported by Josh Kitchens)
 * ASTERISK-24963 - ASAN: heap-use-after-free with PJSIP and WSS
      (Reported by Badalian Vyacheslav)
 * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
      but asterisk doesn't detect it. (Reported by ibercom)
 * ASTERISK-25094 - PBX core: Investigate thread safety issues
      (Reported by Corey Farrell)
 * ASTERISK-25148 - res_pjsip NULL channel audit (Reported by Mark
 * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
      | adpcm | ipc10} (Reported by Badalian Vyacheslav)
 * ASTERISK-25137 - endpoint stasis messages are delivered twice
      (Reported by Vitezslav Novy)
 * ASTERISK-25116 - res_pjsip:  Two PeerStatus AMI messages are
      sent for every status change (Reported by George Joseph)
 * ASTERISK-25131 - chan_pjsip: In-dialog authentication not
      handled. (Reported by Richard Mudgett)
 * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
      that end with ::80 (Reported by Mark Petersen)
 * ASTERISK-25122 - Large SIP packet received via pjsip over
      websocket crashes Asterisk  (Reported by Ivan Poddubny)
 * ASTERISK-25121 - Stasis: Fix unsafe use of stasis_unsubscribe in
      modules. (Reported by Corey Farrell)
 * ASTERISK-24988 - func_talkdetect: Test is bouncing sporadically
      (Reported by Joshua Colp)
 * ASTERISK-25105 - res_pjsip:  Possible incompatibility between
      qualify_timeout and pjproject-2.4 (Reported by George Joseph)
 * ASTERISK-25117 - res_mwi_external_ami: Fix manager action
      registrations. (Reported by Corey Farrell)

New Features made in this release:
 * ASTERISK-25259 - chan_pjsip: Add rtptimeout support (Reported by
      Joshua Colp)
 * ASTERISK-25238 - ARI: Support push configuration (Reported by
      Matt Jordan)
 * ASTERISK-25173 - ARI: Add the ability to load/reload/unload an
      Asterisk module (Reported by Matt Jordan)

For a full list of changes in this release candidate, please see the ChangeLog:


Thank you for your continued support of Asterisk!

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