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The Asterisk Development Team has announced the release of Asterisk 13.5.0.

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Author: Matt Riddell
Daily Asterisk News
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Hi all, sorry I missed this one the other day. So, a little late but here nonetheless.

The Asterisk Development Team has announced the release of Asterisk 13.5.0.

This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

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 Mudgett)
  • 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 Serov)
  • 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 Newton)
  • 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 Colp)
  • 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 Engstrom)
  • ASTERISK-25127 - DTLS crashes following \"Unable to cancel schedule ID\" in dtls_srtp_check_pending (Reported by Dade Brandon)
  • ASTERISK-25168 - Random Core Dumps on Asterisk 13.4 PJSIP, in ast_channel_name at channel_internal_api.c (Reported by Carl Fortin)
  • 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 Michelson)
  • 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 Michelson)
  • 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:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.5.0

Thank you for your continued support of Asterisk!


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