Asterisk News

Asterisk Releases

Asterisk 13.5.0-rc1 Now Available

Jul 27, 2015

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] - [patch]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] - [patch] 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] - [patch]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] - [patch]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] - [patch]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] - [patch]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] - [patch] 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] - [patch]Segfault when using DEBUG_FD_LEAKS
  • [ASTERISK-25219] - [patch]Source and destination overlap in memcpy in rtp_engine.c
  • [ASTERISK-25220] - [patch]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] - [patch]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] - [patch]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!


Asterisk 11.19.0-rc1 Now Available

Jul 27, 2015

The Asterisk Development Team has announced the release of Asterisk 11.19.0-rc1.

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

The release of Asterisk 11.19.0-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] - [patch]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-24550] - res_rtp_asterisk: Crash in ast_rtp_on_ice_complete during DTLS handshake
  • [ASTERISK-24651] - [patch] Fix race condition in DTLS
  • [ASTERISK-24717] - ASAN: global-buffer-overflow codec_{ilbc | gsm | adpcm | ipc10}
  • [ASTERISK-24832] - [patch]DTLS-crashes within openssl
  • [ASTERISK-24853] - Documentation claims chan_sip outbound registrations support WS or WSS as valid transports (not true)
  • [ASTERISK-25094] - PBX core: Investigate thread safety issues
  • [ASTERISK-25100] - asterisk coredump if host has an IPv6 address that end with ::80
  • [ASTERISK-25103] - Roundup - investigate Asterisk DTLS crashes
  • [ASTERISK-25127] - DTLS crashes following "Unable to cancel schedule ID" in dtls_srtp_check_pending
  • [ASTERISK-25139] - Malicious transfer sequence locks up Asterisk
  • [ASTERISK-25154] - [patch]fromtag may need to be updated after successful call dialog match
  • [ASTERISK-25202] - Hints extension state broken between 13.3.2 and 13.4
  • [ASTERISK-25212] - [patch]Segfault when using DEBUG_FD_LEAKS
  • [ASTERISK-25213] - [patch]Possibility of deadlock in chan_sip INVITE early Replace code
  • [ASTERISK-25219] - [patch]Source and destination overlap in memcpy in rtp_engine.c
  • [ASTERISK-25220] - [patch]Closing of fd -1 in chan_mgcp.c
  • [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-25257] - [patch]channels/sig_pri.h -> sig_pri_span -> force_restart_unavailable_chans in wrong scope

Improvement

  • [ASTERISK-25040] - pbx: Improve performance of reloads by making hint destruction more performant

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0-rc1

Thank you for your continued support of Asterisk!


Asterisk 13.4.0 Now Available

Jun 4, 2015

The Asterisk Development Team has announced the release of Asterisk 13.4.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.4.0 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:

Bug

  • [ASTERISK-14233] - [patch] Buddies are always auto-registered when processing the roster
  • [ASTERISK-17608] - func_aes.so cannot be loaded if res_crypto / openssl not compiled
  • [ASTERISK-18032] - [patch] - IPv6 and IPv4 NAT not working
  • [ASTERISK-19608] - Asterisk-1.8.x starts rejecting calls with cause code 44 after some time.
  • [ASTERISK-21211] - chan_iax2 - unprotected access of iaxs[peer->callno] potentially results in segfault
  • [ASTERISK-21777] - Asterisk tries to transcode video instead of audio
  • [ASTERISK-21893] - Segfault after call hangup, in ast_channel_hangupcause_set, at channel_internal_api.c
  • [ASTERISK-22352] - [patch] IAX2 custom qualify timer is not taken into account
  • [ASTERISK-22708] - res_odbc.conf negative_connection_cache option not respected, failover between DSNs doesn't work
  • [ASTERISK-22790] - check_modem_rate() may return incorrect rate for V.27
  • [ASTERISK-23231] - Since 405693 If we have res_fax.conf file set to minrate=2400, then res_fax refuse to load
  • [ASTERISK-23319] - Segmentation fault in queue_exec at app_queue.c
  • [ASTERISK-24142] - CCSS: crash during shutdown due to device lookup in destroyed container
  • [ASTERISK-24155] - [patch]Non-portable and non-reliable recursion detection in ast_malloc
  • [ASTERISK-24380] - core: Native formats are set to h264 with certain audio/video codec configuration, resulting in path translation WARNINGs
  • [ASTERISK-24442] - Outgoing call files don't work properly when set in the future
  • [ASTERISK-24683] - Crash in PBX ast_hashtab_lookup_internal during core restart now
  • [ASTERISK-24731] - res_pjsip_session cannot be unloaded
  • [ASTERISK-24749] - ConfBridge: Wrong language on playing conf-hasjoin and conf-hasleft when played to bridge
  • [ASTERISK-24774] - Segfault in ast_context_destroy with extensions.ael and extensions.conf
  • [ASTERISK-24780] - [patch] - Buddies are always auto-registered when processing the roster
  • [ASTERISK-24781] - PJSIP: Unnecessary 180 Ringing messages sent with undesireabe consequences.
  • [ASTERISK-24782] - StasisEnd event not present for channel that was swapped out for another after completing attended transfer
  • [ASTERISK-24805] - [patch] - ASAN: Race condition (heap-use-after-free) on asterisk closing
  • [ASTERISK-24835] - Early Media Not working with Chan SIP and Asterisk 13
  • [ASTERISK-24841] - ConfBridge: Strange sampling rates chosen when channels have multiple native formats
  • [ASTERISK-24845] - pjsip send notify not working with Cisco phone
  • [ASTERISK-24847] - [security] [patch] tcptls: certificate CN NULL byte prefix bug
  • [ASTERISK-24857] - [patch] "timing test", pjsip incoming/outgoing calls, voicemail prompts and recordings all fail when using the kqueue timer source on FreeBSD 10.x
  • [ASTERISK-24863] - res_pjsip: No endpoint events raised via AMI when contacts cannot be reached/qualified
  • [ASTERISK-24864] - app_confbridge: file playback blocks dtmf
  • [ASTERISK-24869] - Asterisk segfaults on DAHDI attended transfer due to application (appl) being NULL on unbridged channel
  • [ASTERISK-24881] - ast_register_atexit should only be used when absolutely needed
  • [ASTERISK-24887] - [patch]tags in a=crypto lines do not accept 2 or more digits
  • [ASTERISK-24894] - [patch] iax2_poke_noanswer expiration timer too short
  • [ASTERISK-24895] - After hangup on the side of the ISDN network no HangupRequest event comes for the dahdi channel.
  • [ASTERISK-24896] - [patch] Using force black background leads to colours not being reset
  • [ASTERISK-24899] - Parking fall-through behavior different in 13
  • [ASTERISK-24910] - "timer=no" and "timer=required" settings in pjsip.conf fail
  • [ASTERISK-24914] - Division by zero in file.c when playback of voicemail with video as h264
  • [ASTERISK-24920] - Asterisk handles duplicate SIP requests as if they were each a new request
  • [ASTERISK-24928] - [patch]t38_udptl_maxdatagram in pjsip.conf not honored
  • [ASTERISK-24932] - Asterisk 13.x does not build with GCC 5.0
  • [ASTERISK-24933] - T38 fails negotiation
  • [ASTERISK-24935] - res_pjsip_phoneprov_provider: Fix leaked OBJ_MULTIPLE iterator.
  • [ASTERISK-24937] - [patch]res_pjsip_messaging: Messages may be sent out of order
  • [ASTERISK-24938] - ARI Snoop Channel results in excessive escalating CPU usage
  • [ASTERISK-24944] - main/audiohook.c change prevents G722 call recording
  • [ASTERISK-24954] - Git migration: Asterisk version numbers are incompatible with the Test Suite
  • [ASTERISK-24955] - res_fax: v.27ter support baud rate of 2400, which is disallowed in res_fax's check_modem_rate
  • [ASTERISK-24958] - Forwarding loop detection inhibits certain desirable scenarios
  • [ASTERISK-24959] - [patch]CLI command cdr show pgsql status
  • [ASTERISK-24970] - Crash in res_pjsip_pubsub handling of failed notify
  • [ASTERISK-24975] - Enabling 'DEBUG_THREADLOCALS' Causes the Build to Fail
  • [ASTERISK-24976] - cdr_odbc not include new columns added on 1.8
  • [ASTERISK-24977] - Contacts that don't use qualify are being marked as unavailable
  • [ASTERISK-24982] - res_pjsip_mwi: Unsolicited MWI NOTIFY only sent on mailbox changes
  • [ASTERISK-24991] - Check for ao2_alloc failure in __ast_channel_internal_alloc
  • [ASTERISK-24996] - chan_pjsip: Creating Channel Causes Asterisk to Crash When Duplicate AOR Sections Exist in pjsip.conf
  • [ASTERISK-24997] - Astobj2: Some callers of __adjust_lock do not pre-check the object
  • [ASTERISK-24998] - res_corosync: res_corosync tries to load even if res_corosync.conf is missing
  • [ASTERISK-24999] - PJSIP crashes with malformed contact line
  • [ASTERISK-25003] - Asterisk crashes on attended transfer (using feature)
  • [ASTERISK-25004] - Crash in authenticated reinvite after originated T.38 FAX
  • [ASTERISK-25018] - pjsip show endpoints crashes asterisk when qualified aors present
  • [ASTERISK-25020] - Mismatched response to outgoing REGISTER request
  • [ASTERISK-25022] - Memory leak setting up DTLS/SRTP calls
  • [ASTERISK-25025] - Periodic crashes (in ast_channel_snapshot_create at stasis_channels.c) with Certified Asterisk 13.
  • [ASTERISK-25027] - Build System: Many ARI modules are missing dependencies.
  • [ASTERISK-25028] - Build System: Unneeded defines in asterisk/buildopts.h
  • [ASTERISK-25033] - Asterisk 13 (branch head) won't compile without PJSip
  • [ASTERISK-25034] - chan_dahdi: Some telco switches occasionally ignore ISDN RESTART requests.
  • [ASTERISK-25037] - res_pjsip_outbound_registration: Potential crash in off-nominal failure case when sending message
  • [ASTERISK-25038] - Queue log "EXITWITHTIMEOUT" does not always contain waiting time
  • [ASTERISK-25041] - [patch]Broken column type checking in res_config_mysql addon
  • [ASTERISK-25042] - asterisk.conf options override command-line options.
  • [ASTERISK-25048] - Astobj2: Initialization order wrong when both refdebug and AO2_DEBUG are both enabled.
  • [ASTERISK-25053] - Unit test category /main/presence missing trailing slash.
  • [ASTERISK-25054] - Formats interface's cannot be unregistered, needs to hold modules until shutdown.
  • [ASTERISK-25057] - res_pjsip_pubsub: Crash in send_notify due to invalid root pointer in sub_tree
  • [ASTERISK-25061] - pbx_config: Register manager actions with module version of macro.
  • [ASTERISK-25074] - Regression: Recent clang-related change broke cross compiling of Asterisk
  • [ASTERISK-25082] - Asterisk deletes message after doing a playback of an INBOX message using ast_vm_play when the Old folder is full for that mailbox.
  • [ASTERISK-25083] - Message.c: Message channel becomes saturated with frames leading to spammy log messages
  • [ASTERISK-25085] - [patch]Potential crash after unload of func_periodic_hook or test_message
  • [ASTERISK-25086] - [patch]PJSIP crashes if endpoint missing in Dial()
  • [ASTERISK-25089] - res_pjsip_config_wizard: Variable specified in templates aren't being processed correctly
  • [ASTERISK-25090] - CLI core show channel truncates cdr variables
  • [ASTERISK-25112] - Logger: Configuration settings are not reset to default during reload.

Improvement

New Feature

  • [ASTERISK-24922] - ARI: Add the ability to intercept hold and raise an event

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.4.0

Thank you for your continued support of Asterisk!


Asterisk 11.18.0 Now Available

Jun 4, 2015

The Asterisk Development Team has announced the release of Asterisk 11.18.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 11.18.0 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:

Bug

  • [ASTERISK-14233] - [patch] Buddies are always auto-registered when processing the roster
  • [ASTERISK-18032] - [patch] - IPv6 and IPv4 NAT not working
  • [ASTERISK-19538] - Asterisk segfaults on sippeers realtime redundancy
  • [ASTERISK-19608] - Asterisk-1.8.x starts rejecting calls with cause code 44 after some time.
  • [ASTERISK-21211] - chan_iax2 - unprotected access of iaxs[peer->callno] potentially results in segfault
  • [ASTERISK-21777] - Asterisk tries to transcode video instead of audio
  • [ASTERISK-21854] - Long Asterisk-version strings display improperly in the 'Connected to ...' line upon remote console connection
  • [ASTERISK-21893] - Segfault after call hangup, in ast_channel_hangupcause_set, at channel_internal_api.c
  • [ASTERISK-22352] - [patch] IAX2 custom qualify timer is not taken into account
  • [ASTERISK-22708] - res_odbc.conf negative_connection_cache option not respected, failover between DSNs doesn't work
  • [ASTERISK-22790] - check_modem_rate() may return incorrect rate for V.27
  • [ASTERISK-23231] - Since 405693 If we have res_fax.conf file set to minrate=2400, then res_fax refuse to load
  • [ASTERISK-23319] - Segmentation fault in queue_exec at app_queue.c
  • [ASTERISK-24142] - CCSS: crash during shutdown due to device lookup in destroyed container
  • [ASTERISK-24155] - [patch]Non-portable and non-reliable recursion detection in ast_malloc
  • [ASTERISK-24380] - core: Native formats are set to h264 with certain audio/video codec configuration, resulting in path translation WARNINGs
  • [ASTERISK-24442] - Outgoing call files don't work properly when set in the future
  • [ASTERISK-24683] - Crash in PBX ast_hashtab_lookup_internal during core restart now
  • [ASTERISK-24749] - ConfBridge: Wrong language on playing conf-hasjoin and conf-hasleft when played to bridge
  • [ASTERISK-24774] - Segfault in ast_context_destroy with extensions.ael and extensions.conf
  • [ASTERISK-24780] - [patch] - Buddies are always auto-registered when processing the roster
  • [ASTERISK-24805] - [patch] - ASAN: Race condition (heap-use-after-free) on asterisk closing
  • [ASTERISK-24847] - [security] [patch] tcptls: certificate CN NULL byte prefix bug
  • [ASTERISK-24864] - app_confbridge: file playback blocks dtmf
  • [ASTERISK-24881] - ast_register_atexit should only be used when absolutely needed
  • [ASTERISK-24887] - [patch]tags in a=crypto lines do not accept 2 or more digits
  • [ASTERISK-24894] - [patch] iax2_poke_noanswer expiration timer too short
  • [ASTERISK-24895] - After hangup on the side of the ISDN network no HangupRequest event comes for the dahdi channel.
  • [ASTERISK-24916] - Increasing memory usage when multiple reinvite during call
  • [ASTERISK-24932] - Asterisk 13.x does not build with GCC 5.0
  • [ASTERISK-24942] - Voicemail API: message is deleted when destination mailbox is at maxmsg
  • [ASTERISK-24944] - main/audiohook.c change prevents G722 call recording
  • [ASTERISK-24954] - Git migration: Asterisk version numbers are incompatible with the Test Suite
  • [ASTERISK-24955] - res_fax: v.27ter support baud rate of 2400, which is disallowed in res_fax's check_modem_rate
  • [ASTERISK-24959] - [patch]CLI command cdr show pgsql status
  • [ASTERISK-24975] - Enabling 'DEBUG_THREADLOCALS' Causes the Build to Fail
  • [ASTERISK-24976] - cdr_odbc not include new columns added on 1.8
  • [ASTERISK-24991] - Check for ao2_alloc failure in __ast_channel_internal_alloc
  • [ASTERISK-25022] - Memory leak setting up DTLS/SRTP calls
  • [ASTERISK-25028] - Build System: Unneeded defines in asterisk/buildopts.h
  • [ASTERISK-25034] - chan_dahdi: Some telco switches occasionally ignore ISDN RESTART requests.
  • [ASTERISK-25038] - Queue log "EXITWITHTIMEOUT" does not always contain waiting time
  • [ASTERISK-25041] - [patch]Broken column type checking in res_config_mysql addon
  • [ASTERISK-25042] - asterisk.conf options override command-line options.
  • [ASTERISK-25074] - Regression: Recent clang-related change broke cross compiling of Asterisk
  • [ASTERISK-25083] - Message.c: Message channel becomes saturated with frames leading to spammy log messages
  • [ASTERISK-25112] - Logger: Configuration settings are not reset to default during reload.

Improvement

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0

Thank you for your continued support of Asterisk!


Asterisk 13.4.0-rc1 Now Available

May 21, 2015

The Asterisk Development Team has announced the first release candidate of Asterisk 13.4.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.4.0-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:

Bug

  • [ASTERISK-14233] - [patch] Buddies are always auto-registered when processing the roster
  • [ASTERISK-17608] - func_aes.so cannot be loaded if res_crypto / openssl not compiled
  • [ASTERISK-18032] - [patch] - IPv6 and IPv4 NAT not working
  • [ASTERISK-19608] - Asterisk-1.8.x starts rejecting calls with cause code 44 after some time.
  • [ASTERISK-21211] - chan_iax2 - unprotected access of iaxs[peer->callno] potentially results in segfault
  • [ASTERISK-21777] - Asterisk tries to transcode video instead of audio
  • [ASTERISK-21893] - Segfault after call hangup, in ast_channel_hangupcause_set, at channel_internal_api.c
  • [ASTERISK-22352] - [patch] IAX2 custom qualify timer is not taken into account
  • [ASTERISK-22708] - res_odbc.conf negative_connection_cache option not respected, failover between DSNs doesn't work
  • [ASTERISK-22790] - check_modem_rate() may return incorrect rate for V.27
  • [ASTERISK-23231] - Since 405693 If we have res_fax.conf file set to minrate=2400, then res_fax refuse to load
  • [ASTERISK-23319] - Segmentation fault in queue_exec at app_queue.c
  • [ASTERISK-24142] - CCSS: crash during shutdown due to device lookup in destroyed container
  • [ASTERISK-24155] - [patch]Non-portable and non-reliable recursion detection in ast_malloc
  • [ASTERISK-24380] - core: Native formats are set to h264 with certain audio/video codec configuration, resulting in path translation WARNINGs
  • [ASTERISK-24442] - Outgoing call files don't work properly when set in the future
  • [ASTERISK-24683] - Crash in PBX ast_hashtab_lookup_internal during core restart now
  • [ASTERISK-24731] - res_pjsip_session cannot be unloaded
  • [ASTERISK-24749] - ConfBridge: Wrong language on playing conf-hasjoin and conf-hasleft when played to bridge
  • [ASTERISK-24774] - Segfault in ast_context_destroy with extensions.ael and extensions.conf
  • [ASTERISK-24780] - [patch] - Buddies are always auto-registered when processing the roster
  • [ASTERISK-24781] - PJSIP: Unnecessary 180 Ringing messages sent with undesireabe consequences.
  • [ASTERISK-24782] - StasisEnd event not present for channel that was swapped out for another after completing attended transfer
  • [ASTERISK-24805] - [patch] - ASAN: Race condition (heap-use-after-free) on asterisk closing
  • [ASTERISK-24835] - Early Media Not working with Chan SIP and Asterisk 13
  • [ASTERISK-24841] - ConfBridge: Strange sampling rates chosen when channels have multiple native formats
  • [ASTERISK-24845] - pjsip send notify not working with Cisco phone
  • [ASTERISK-24847] - [security] [patch] tcptls: certificate CN NULL byte prefix bug
  • [ASTERISK-24857] - [patch] "timing test", pjsip incoming/outgoing calls, voicemail prompts and recordings all fail when using the kqueue timer source on FreeBSD 10.x
  • [ASTERISK-24863] - res_pjsip: No endpoint events raised via AMI when contacts cannot be reached/qualified
  • [ASTERISK-24864] - app_confbridge: file playback blocks dtmf
  • [ASTERISK-24869] - Asterisk segfaults on DAHDI attended transfer due to application (appl) being NULL on unbridged channel
  • [ASTERISK-24881] - ast_register_atexit should only be used when absolutely needed
  • [ASTERISK-24887] - [patch]tags in a=crypto lines do not accept 2 or more digits
  • [ASTERISK-24894] - [patch] iax2_poke_noanswer expiration timer too short
  • [ASTERISK-24895] - After hangup on the side of the ISDN network no HangupRequest event comes for the dahdi channel.
  • [ASTERISK-24896] - [patch] Using force black background leads to colours not being reset
  • [ASTERISK-24899] - Parking fall-through behavior different in 13
  • [ASTERISK-24910] - "timer=no" and "timer=required" settings in pjsip.conf fail
  • [ASTERISK-24914] - Division by zero in file.c when playback of voicemail with video as h264
  • [ASTERISK-24920] - Asterisk handles duplicate SIP requests as if they were each a new request
  • [ASTERISK-24928] - [patch]t38_udptl_maxdatagram in pjsip.conf not honored
  • [ASTERISK-24932] - Asterisk 13.x does not build with GCC 5.0
  • [ASTERISK-24933] - T38 fails negotiation
  • [ASTERISK-24935] - res_pjsip_phoneprov_provider: Fix leaked OBJ_MULTIPLE iterator.
  • [ASTERISK-24937] - [patch]res_pjsip_messaging: Messages may be sent out of order
  • [ASTERISK-24938] - ARI Snoop Channel results in excessive escalating CPU usage
  • [ASTERISK-24944] - main/audiohook.c change prevents G722 call recording
  • [ASTERISK-24954] - Git migration: Asterisk version numbers are incompatible with the Test Suite
  • [ASTERISK-24955] - res_fax: v.27ter support baud rate of 2400, which is disallowed in res_fax's check_modem_rate
  • [ASTERISK-24958] - Forwarding loop detection inhibits certain desirable scenarios
  • [ASTERISK-24959] - [patch]CLI command cdr show pgsql status
  • [ASTERISK-24970] - Crash in res_pjsip_pubsub handling of failed notify
  • [ASTERISK-24975] - Enabling 'DEBUG_THREADLOCALS' Causes the Build to Fail
  • [ASTERISK-24976] - cdr_odbc not include new columns added on 1.8
  • [ASTERISK-24977] - Contacts that don't use qualify are being marked as unavailable
  • [ASTERISK-24982] - res_pjsip_mwi: Unsolicited MWI NOTIFY only sent on mailbox changes
  • [ASTERISK-24991] - Check for ao2_alloc failure in __ast_channel_internal_alloc
  • [ASTERISK-24996] - chan_pjsip: Creating Channel Causes Asterisk to Crash When Duplicate AOR Sections Exist in pjsip.conf
  • [ASTERISK-24997] - Astobj2: Some callers of __adjust_lock do not pre-check the object
  • [ASTERISK-24998] - res_corosync: res_corosync tries to load even if res_corosync.conf is missing
  • [ASTERISK-24999] - PJSIP crashes with malformed contact line
  • [ASTERISK-25003] - Asterisk crashes on attended transfer (using feature)
  • [ASTERISK-25004] - Crash in authenticated reinvite after originated T.38 FAX
  • [ASTERISK-25018] - pjsip show endpoints crashes asterisk when qualified aors present
  • [ASTERISK-25020] - Mismatched response to outgoing REGISTER request
  • [ASTERISK-25022] - Memory leak setting up DTLS/SRTP calls
  • [ASTERISK-25025] - Periodic crashes (in ast_channel_snapshot_create at stasis_channels.c) with Certified Asterisk 13.
  • [ASTERISK-25027] - Build System: Many ARI modules are missing dependencies.
  • [ASTERISK-25028] - Build System: Unneeded defines in asterisk/buildopts.h
  • [ASTERISK-25033] - Asterisk 13 (branch head) won't compile without PJSip
  • [ASTERISK-25034] - chan_dahdi: Some telco switches occasionally ignore ISDN RESTART requests.
  • [ASTERISK-25037] - res_pjsip_outbound_registration: Potential crash in off-nominal failure case when sending message
  • [ASTERISK-25038] - Queue log "EXITWITHTIMEOUT" does not always contain waiting time
  • [ASTERISK-25041] - [patch]Broken column type checking in res_config_mysql addon
  • [ASTERISK-25042] - asterisk.conf options override command-line options.
  • [ASTERISK-25048] - Astobj2: Initialization order wrong when both refdebug and AO2_DEBUG are both enabled.
  • [ASTERISK-25053] - Unit test category /main/presence missing trailing slash.
  • [ASTERISK-25054] - Formats interface's cannot be unregistered, needs to hold modules until shutdown.
  • [ASTERISK-25057] - res_pjsip_pubsub: Crash in send_notify due to invalid root pointer in sub_tree
  • [ASTERISK-25061] - pbx_config: Register manager actions with module version of macro.
  • [ASTERISK-25074] - Regression: Recent clang-related change broke cross compiling of Asterisk
  • [ASTERISK-25082] - Asterisk deletes message after doing a playback of an INBOX message using ast_vm_play when the Old folder is full for that mailbox.
  • [ASTERISK-25083] - Message.c: Message channel becomes saturated with frames leading to spammy log messages
  • [ASTERISK-25085] - [patch]Potential crash after unload of func_periodic_hook or test_message
  • [ASTERISK-25086] - [patch]PJSIP crashes if endpoint missing in Dial()
  • [ASTERISK-25089] - res_pjsip_config_wizard: Variable specified in templates aren't being processed correctly
  • [ASTERISK-25090] - CLI core show channel truncates cdr variables
  • [ASTERISK-25112] - Logger: Configuration settings are not reset to default during reload.

Improvement

New Feature

  • [ASTERISK-24922] - ARI: Add the ability to intercept hold and raise an event

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.4.0-rc1

Thank you for your continued support of Asterisk!


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