Asterisk 13.5.0-rc1 Now Available

The Asterisk Development Team has announced the release of Asterisk 13.5.0-rc1. This release 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:

Bug

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

Improvement

  • [ASTERISK-25040] – pbx: Improve performance of reloads by making hint destruction more performant
  • [ASTERISK-25067] – Sorcery Caching: Implement a new caching module
  • [ASTERISK-25072] – res_pjsip_outbound_registration: line functionality. Additional check for using the request URI
  • [ASTERISK-25114] – res_pjsip: Add AMI events for chan_pjsip contact lifecycle changes
  • [ASTERISK-25256] – Post AMI VarSet to empty string events when Asterisk deletes a dialplan variable.

New Feature

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.5.0-rc1

Thank you for your continued support of Asterisk!

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