[asterisk-announce] Asterisk 13.5.0 Now Available

Asterisk Development Team asteriskteam at digium.com
Fri Aug 7 16:56:18 CDT 2015

The Asterisk Development Team has announced the release of Asterisk 13.5.0.
This release is available for immediate download at

The release of Asterisk 13.5.0 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:

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, please see the ChangeLog:


Thank you for your continued support of Asterisk!

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