Asterisk 13.5.0 Now Available

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:


  • [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


  • [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:

Thank you for your continued support of Asterisk!

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