[asterisk-dev] Asterisk 13.6.0 Now Available

The Asterisk Development Team has announced the release of Asterisk 13.6.0.

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

The release of Asterisk 13.6.0 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:

New Features made in this release:

———————————–

* ASTERISK-25377 – res_pjsip: Change default “From user” from UUID

     to something more palatable (Reported by Mark Michelson)

* ASTERISK-25252 – ARI: Add the ability to manipulate log channels

     (Reported by Matt Jordan)

Bugs fixed in this release:

———————————–

* ASTERISK-25449 – main/sched: Regression introduced by

     5c713fdf18f causes erroneous duplicate RTCP messages; other

     potential scheduling issues in chan_sip/chan_skinny (Reported by

     Matt Jordan)

* ASTERISK-25438 – res_rtp_asterisk: ICE role message even when

     ICE is not enabled (Reported by Joshua Colp)

* ASTERISK-25383 – Core dumps on startup and shutdown with

     MALLOC_DEBUG enabled (Reported by yaron nahum)

* ASTERISK-25423 – Caller gets no Connected line update during

     call pickup. (Reported by Richard Mudgett)

* ASTERISK-25305 – Dynamic logger channels can be added multiple

     times (Reported by Mark Michelson)

* ASTERISK-25418 – On-hold channels redirected out of a bridge

     appear to still be on hold (Reported by Mark Michelson)

* ASTERISK-25384 – Regular Asterisk crashes when using Page

     application. “user_data is NULL” (Reported by Chet Stevens)

* ASTERISK-25407 – Asterisk fails to log to multiple syslog

     destinations (Reported by Elazar Broad)

* ASTERISK-25410 – app_record: RECORDED_FILE variable not being

     populated (Reported by Kevin Harwell)

* ASTERISK-25394 – pbx: Incorrect device and presence state when

     changing hint details (Reported by Joshua Colp)

* ASTERISK-25396 – chan_sip: Extremely long callerid name causes

     invalid SIP (Reported by Walter Doekes)

* ASTERISK-25399 – app_queue: AgentComplete event has wrong reason

     (Reported by Kevin Harwell)

* ASTERISK-25185 – Segfault in app_queue on transfer scenarios

     (Reported by Etienne Lessard)

* ASTERISK-25353 – Transcoding while different in Frame

     size = Frames lost (Reported by Alexander Traud)

* ASTERISK-25325 – ARI PUT reload chan_sip HTTP response 404

     (Reported by Rodrigo Ramirez Norambuena)

* ASTERISK-25390 – default_from_user can crash with certain

     configuration backends (Reported by Mark Michelson)

* ASTERISK-25387 – res_pjsip_nat: Malformed REGISTER request

     causes NAT’d Contact header to not be rewritten (Reported by

     Matt Jordan)

* ASTERISK-25227 – No audio at in-band announcements in ooh323

     channel (Reported by Alexandr Dranchuk)

* ASTERISK-25369 – res_parking: ParkAndAnnounce – Inheritable

     variables aren’t applied to the announcer channel (Reported by

     Jonathan Rose)

* ASTERISK-25295 – res_pjsip crash – pjsip_uri_get_uri at

     /usr/include/pjsip/sip_uri.h (Reported by Dmitriy Serov)

* ASTERISK-25381 – res_pjsip: AoRs deleted via ARI (or other

     mechanism) do not destroy their related contacts (Reported by

     Matt Jordan)

* ASTERISK-25367 – pbx: Long pattern match hints may cause “core

     show hints” to crash (Reported by Joshua Colp)

* ASTERISK-25365 – Persistent subscriptions have extra

     Content-Length/corrupted messages (Reported by Mark Michelson)

* ASTERISK-25362 – Deadlock due to presence state callback

     (Reported by Mark Michelson)

* ASTERISK-25356 – res_pjsip_sdp_rtp: Multiple keepalive scheduled

     items may exist (Reported by Joshua Colp)

* ASTERISK-25355 – sched: ast_sched_del may return prematurely due

     to spurious wakeup (Reported by Joshua Colp)

* ASTERISK-25318 –

     tests/rest_api/applications/subscribe-endpoint/nominal/resource:

     Sporadically failing (Reported by Joshua Colp)

* ASTERISK-25346 – chan_sip: Overwriting answered elsewhere hangup

     cause on call pickup (Reported by Joshua Colp)

* ASTERISK-25342 – res_pjsip: Repeated usage of pj_gethostip may

     block (Reported by Joshua Colp)

* ASTERISK-25341 – bridge: Hangups may get lost when executing

     actions (Reported by Joshua Colp)

* ASTERISK-25339 – res_pjsip: Empty “auth” sections from

     non-config backgrounds are interpreted as valid (Reported by

     Matt Jordan)

* ASTERISK-25215 – Differences in queue.log between Set

     QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne

     Gaetz)

* ASTERISK-25322 – Crash occurs when using MixMonitor with t() or

     r() options. (Reported by Richard Mudgett)

* ASTERISK-25320 – chan_sip.c: sip_report_security_event searches

     for wrong or non existent peer on invite (Reported by Kevin

     Harwell)

* ASTERISK-25315 – DAHDI channels send shortened duration DTMF

     tones. (Reported by Richard Mudgett)

* ASTERISK-25312 – res_http_websocket: Terminate connection on

     fatal cases (Reported by Joshua Colp)

* ASTERISK-25306 – Persistent subscriptions can save multiple SIP

     messages at once, leading to potential crashes. (Reported by

     Mark Michelson)

* ASTERISK-25309 – iLBC 20 advertised (Reported by

     Alexander Traud)

* ASTERISK-25304 – res_pjsip: XML sanitization may write past

     buffer (Reported by Joshua Colp)

* ASTERISK-25265 – DTLS Failure when calling WebRTC-peer on

     Firefox 39 – add ECDH support and fallback to prime256v1

     (Reported by Stefan Engström)

* ASTERISK-25296 – RTP performance issue with several channel

     drivers. (Reported by Richard Mudgett)

* ASTERISK-25297 – Crashes running

     channels/pjsip/resolver/srv/failover/in_dialog testsuite tests

     (Reported by Richard Mudgett)

* ASTERISK-25292 – Testuite:

     tests/apps/bridge/bridge_wait/bridge_wait_e_options fails

     (Reported by Kevin Harwell)

* ASTERISK-25271 – Parking & blind transfer: Transferer channel

     not hung up if no MOH (Reported by Kevin Harwell)

Improvements made in this release:

———————————–

* ASTERISK-24870 – ARI: Subscriptions to bridges generally not

     super useful (Reported by Matt Jordan)

* ASTERISK-25310 – on FreeBSD also pthread_attr_init()

     defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.6.0

[asterisk-dev] Asterisk 11.20.0 Now Available

The Asterisk Development Team has announced the release of Asterisk 11.20.0.

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

The release of Asterisk 11.20.0 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:

Bugs fixed in this release:

———————————–

* ASTERISK-25449 – main/sched: Regression introduced by

     5c713fdf18f causes erroneous duplicate RTCP messages; other

     potential scheduling issues in chan_sip/chan_skinny (Reported by

     Matt Jordan)

* ASTERISK-25438 – res_rtp_asterisk: ICE role message even when

     ICE is not enabled (Reported by Joshua Colp)

* ASTERISK-25427 – Callerid change does not always emit

     NewCallerid AMI event (Reported by Ivan Poddubny)

* ASTERISK-25407 – Asterisk fails to log to multiple syslog

     destinations (Reported by Elazar Broad)

* ASTERISK-25410 – app_record: RECORDED_FILE variable not being

     populated (Reported by Kevin Harwell)

* ASTERISK-25394 – pbx: Incorrect device and presence state when

     changing hint details (Reported by Joshua Colp)

* ASTERISK-25396 – chan_sip: Extremely long callerid name causes

     invalid SIP (Reported by Walter Doekes)

* ASTERISK-25353 – Transcoding while different in Frame

     size = Frames lost (Reported by Alexander Traud)

* ASTERISK-25227 – No audio at in-band announcements in ooh323

     channel (Reported by Alexandr Dranchuk)

* ASTERISK-25346 – chan_sip: Overwriting answered elsewhere hangup

     cause on call pickup (Reported by Joshua Colp)

* ASTERISK-25215 – Differences in queue.log between Set

     QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne

     Gaetz)

* ASTERISK-25320 – chan_sip.c: sip_report_security_event searches

     for wrong or non existent peer on invite (Reported by Kevin

     Harwell)

* ASTERISK-25315 – DAHDI channels send shortened duration DTMF

     tones. (Reported by Richard Mudgett)

* ASTERISK-25312 – res_http_websocket: Terminate connection on

     fatal cases (Reported by Joshua Colp)

* ASTERISK-25265 – DTLS Failure when calling WebRTC-peer on

     Firefox 39 – add ECDH support and fallback to prime256v1

     (Reported by Stefan Engström)

Improvements made in this release:

———————————–

* ASTERISK-25310 – on FreeBSD also pthread_attr_init()

     defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0

[asterisk-dev] 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 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 – 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 – channels/sig_pri.h -> sig_pri_span ->

     force_restart_unavailable_chans in wrong scope (Reported by

     Patric Marschall)

* ASTERISK-24934 – 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 – Fix race condition in DTLS (Reported by

     Badalian Vyacheslav)

* ASTERISK-24832 – 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

     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 – Closing of fd -1 in chan_mgcp.c

     (Reported by Walter Doekes)

* ASTERISK-25219 – Source and destination overlap in memcpy

     in rtp_engine.c (Reported by Walter Doekes)

* ASTERISK-25212 – Segfault when using DEBUG_FD_LEAKS

     (Reported by Walter Doekes)

* ASTERISK-19277 – 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 – 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 – 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 – 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!

[asterisk-dev] Asterisk 11.19.0 Now Available

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

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

The release of Asterisk 11.19.0 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:

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-25247 – choppy audio when spying on a g722 channel,

     chan_sip or chan_pjsip (Reported by hristo)

* ASTERISK-24853 – Documentation claims chan_sip outbound

     registrations support WS or WSS as valid transports (not true)

     (Reported by PSDK)

* ASTERISK-25257 – channels/sig_pri.h -> sig_pri_span ->

     force_restart_unavailable_chans in wrong scope (Reported by

     Patric Marschall)

* 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 – Fix race condition in DTLS (Reported by

     Badalian Vyacheslav)

* ASTERISK-24832 – 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

     Brandon)

* ASTERISK-25213 – Possibility of deadlock in chan_sip

     INVITE early Replace code (Reported by Walter Doekes)

* ASTERISK-25220 – Closing of fd -1 in chan_mgcp.c

     (Reported by Walter Doekes)

* ASTERISK-25219 – Source and destination overlap in memcpy

     in rtp_engine.c (Reported by Walter Doekes)

* ASTERISK-25212 – Segfault when using DEBUG_FD_LEAKS

     (Reported by Walter Doekes)

* ASTERISK-19277 – endlessly repeating error: “poll failed:

     Bad file descriptor” (Reported by Barry Chern)

* ASTERISK-25202 – Hints extension state broken between 13.3.2 and

     13.4 (Reported by cervajs)

* ASTERISK-25154 – fromtag may need to be updated after

     successful call dialog match (Reported by Damian Ivereigh)

* ASTERISK-25139 – Malicious transfer sequence locks up Asterisk

     (Reported by Gregory Massel)

* ASTERISK-25094 – PBX core: Investigate thread safety issues

     (Reported by Corey Farrell)

* ASTERISK-22559 – gcc 4.6 and higher supports weakref attribute

     but asterisk doesn’t detect it. (Reported by ibercom)

* ASTERISK-24717 – ASAN: global-buffer-overflow codec_{ilbc | gsm

     | adpcm | ipc10} (Reported by Badalian Vyacheslav)

* ASTERISK-25100 – asterisk coredump if host has an IPv6 address

     that end with ::80 (Reported by Mark Petersen)

Improvements made in this release:

———————————–

* ASTERISK-25040 – pbx: Improve performance of reloads by making

     hint destruction more performant (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-11.19.0

Thank you for your continued support of Asterisk!

[asterisk-dev] DAHDI-Linux and DAHDI-Tools 2.10.2 Now Available

The Asterisk Development Team has announced the releases of:

DAHDI-Linux-v2.10.2

DAHDI-Tools-v2.10.2

dahdi-linux-complete-2.10.2+2.10.2

This release is available for immediate download at:

http://downloads.asterisk.org/pub/telephony/dahdi-linux

http://downloads.asterisk.org/pub/telephony/dahdi-tools

http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete

* Fixes compile issues with linux kernel source up to 4.1

* Fixes “I/O error reported by firmware” issue on TE131,TE133,TE235,TE435,A4B,and A8B

Shortlog of dahdi-linux changes since v2.10.1:

John Sloan (1):

     wcxb: Fix “I/O error reported by firmware” followed by underruns

Michael Walton (1):

     dynamic: Prevent oops due to inverted compile flag

Russ Meyerriecks (1):

     wctc4xxp: Fix continuous “errored receive packets” with 2+ transcoders

Shaun Ruffell (10):

     dahdi: Fix failure to read / write on kernel 3.16+

     dahdi: smp_mb_{before,after}_clear_bit -> smp_mb_{before,after}_atomic.

     build_tools/dkms-helper: Use bash to process dkms-helper script.

     dahdi_dynamic: Release reference count on network device when destroying dynamic spans.

     dahdi: struct file.f_dentry macro was removed in kernel 3.19

     dahdi: Fix “void value not ignored…” error when compiling against kernel 4.0.

     dahdi: strnicmp() -> strncasecmp()

     Remove DAHDI_IRQF_[SHARED|DISABLED] flags.

     dahdi: Remove IRQF_DISABLED.

     wctc4xxp: Clear packet error count when reloading firmware.

Tzafrir Cohen (6):

     xpp: FPGA_1161.201.hex: module types detection

     xpp: firmware: 203 as alias to (newer) 201

     xpp: firmware: a stray ^Z in FPGA_1161.201.hex

     xpp: module_refcount is back to int on 3.19

     README: use file time for reproducable build

     xpp: USB_FW*: fix incorrect chan num with 2FXS6FXO

Shortlog of dahdi-tools changes since v2.10.1:

Oron Peled (8):

     xpp: revert USB “clear_halt” change and better overrides.

     xpp: astribank_is_starting: improve ‘-v’ output

     xpp: waitfor_xpds: expansion error with no ABs

     xpp: waitfor_xpds: assume astribank_is_starting exists

     xpp: can use modern Asterisk hotplug support

     xpp: waitfor_xpds: documentation

     xpp/astribank_hook: remove Astribank initialization

     xpp: waitfor_xpds: Always remove Astribank semaphore

Russ Meyerriecks (1):

     tonezone: Fix regression in Australian tone patch

Tzafrir Cohen (3):

     astribank_hook: remove useless ‘time’

     no astribank_is_starting with hotplug asterisk

     xpp_fxloader: handle empty span-type.conf

The diffstat from the dahdi-linux v2.10.1 release:

Makefile                                   |  3 +-

drivers/dahdi/firmware/Makefile            |  8 ++—

drivers/dahdi/pciradio.c                   |  3 +-

drivers/dahdi/tor2.c                       |  7 +++-

drivers/dahdi/voicebus/voicebus.c          |  2 +-

drivers/dahdi/wcaxx-base.c                 |  4 +–

drivers/dahdi/wcb4xxp/base.c               |  7 +++-

drivers/dahdi/wcfxo.c                      |  2 +-

drivers/dahdi/wct1xxp.c                    | 10 ++++–

drivers/dahdi/wct4xxp/base.c               |  4 +–

drivers/dahdi/wctc4xxp/base.c              | 14 +++++—

drivers/dahdi/wctdm.c                      |  3 +-

drivers/dahdi/wcte11xp.c                   | 10 ++++–

drivers/dahdi/wcte13xp-base.c              |  2 +-

drivers/dahdi/wcte43x-base.c               |  2 +-

drivers/dahdi/wcxb.c                       | 56 ++++++++++——————–

drivers/dahdi/wcxb.h                       |  7 ++–

drivers/dahdi/xpp/card_pri.c               |  6 ++–

drivers/dahdi/xpp/firmwares/USB_FW.201.hex |  6 ++–

drivers/dahdi/xpp/firmwares/USB_FW.hex     |  6 ++–

drivers/dahdi/xpp/xproto.c                 |  3 +-

include/dahdi/kernel.h                     | 12 +++—-

22 files changed, 93 insertions(+), 84 deletions(-)

The diffstat from the dahdi-tools v2.10.1 release:

xpp/xpp_fxloader | 5 +++–

1 file changed, 3 insertions(+), 2 deletions(-)

For a full list of changes in these releases, please see the shortlog at:

http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=shortlog;h=refs/tags/v2.10.2

http://git.asterisk.org/gitweb/?p=dahdi/tools.git;a=shortlog;h=refs/tags/v2.10.2

Issues found in this release can be reported in the DAHDI-Linux [1] and

DAHDI-Tools [2] projects at https://issues.asterisk.org/jira

[1] https://issues.asterisk.org/jira/browse/DAHLIN

[2] https://issues.asterisk.org/jira/browse/DAHTOOL

[asterisk-dev] DAHDI-Linux and DAHDI-Tools 2.10.2-rc1 Now Available

The Asterisk Development Team has announced the releases of:

DAHDI-Linux-v2.10.2-rc1

DAHDI-Tools-v2.10.2-rc1

dahdi-linux-complete-2.10.2-rc1+2.10.2-rc1

This release is available for immediate download at:

http://downloads.asterisk.org/pub/telephony/dahdi-linux

http://downloads.asterisk.org/pub/telephony/dahdi-tools

http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete

* Fixes compile issues with linux kernel source up to 4.1

* Fixes “I/O error reported by firmware” issue on TE131,TE133,TE235,TE435,A4B,and A8B

Shortlog of dahdi-linux changes since v2.10.1:

John Sloan (1):

     wcxb: Fix “I/O error reported by firmware” followed by underruns

Michael Walton (1):

     dynamic: Prevent oops due to inverted compile flag

Russ Meyerriecks (1):

     wctc4xxp: Fix continuous “errored receive packets” with 2+ transcoders

Shaun Ruffell (10):

     dahdi: Fix failure to read / write on kernel 3.16+

     dahdi: smp_mb_{before,after}_clear_bit -> smp_mb_{before,after}_atomic.

     build_tools/dkms-helper: Use bash to process dkms-helper script.

     dahdi_dynamic: Release reference count on network device when destroying dynamic spans.

     dahdi: struct file.f_dentry macro was removed in kernel 3.19

     dahdi: Fix “void value not ignored…” error when compiling against kernel 4.0.

     dahdi: strnicmp() -> strncasecmp()

     Remove DAHDI_IRQF_[SHARED|DISABLED] flags.

     dahdi: Remove IRQF_DISABLED.

     wctc4xxp: Clear packet error count when reloading firmware.

Tzafrir Cohen (6):

     xpp: FPGA_1161.201.hex: module types detection

     xpp: firmware: 203 as alias to (newer) 201

     xpp: firmware: a stray ^Z in FPGA_1161.201.hex

     xpp: module_refcount is back to int on 3.19

     README: use file time for reproducable build

     xpp: USB_FW*: fix incorrect chan num with 2FXS6FXO

Shortlog of dahdi-tools changes since v2.10.1:

Oron Peled (8):

     xpp: revert USB “clear_halt” change and better overrides.

     xpp: astribank_is_starting: improve ‘-v’ output

     xpp: waitfor_xpds: expansion error with no ABs

     xpp: waitfor_xpds: assume astribank_is_starting exists

     xpp: can use modern Asterisk hotplug support

     xpp: waitfor_xpds: documentation

     xpp/astribank_hook: remove Astribank initialization

     xpp: waitfor_xpds: Always remove Astribank semaphore

Russ Meyerriecks (1):

     tonezone: Fix regression in Australian tone patch

Tzafrir Cohen (3):

     astribank_hook: remove useless ‘time’

     no astribank_is_starting with hotplug asterisk

     xpp_fxloader: handle empty span-type.conf

The diffstat from the dahdi-linux v2.10.1 release:

Makefile                                   |  3 +-

drivers/dahdi/firmware/Makefile            |  8 ++—

drivers/dahdi/pciradio.c                   |  3 +-

drivers/dahdi/tor2.c                       |  7 +++-

drivers/dahdi/voicebus/voicebus.c          |  2 +-

drivers/dahdi/wcaxx-base.c                 |  4 +–

drivers/dahdi/wcb4xxp/base.c               |  7 +++-

drivers/dahdi/wcfxo.c                      |  2 +-

drivers/dahdi/wct1xxp.c                    | 10 ++++–

drivers/dahdi/wct4xxp/base.c               |  4 +–

drivers/dahdi/wctc4xxp/base.c              | 14 +++++—

drivers/dahdi/wctdm.c                      |  3 +-

drivers/dahdi/wcte11xp.c                   | 10 ++++–

drivers/dahdi/wcte13xp-base.c              |  2 +-

drivers/dahdi/wcte43x-base.c               |  2 +-

drivers/dahdi/wcxb.c                       | 56 ++++++++++——————–

drivers/dahdi/wcxb.h                       |  7 ++–

drivers/dahdi/xpp/card_pri.c               |  6 ++–

drivers/dahdi/xpp/firmwares/USB_FW.201.hex |  6 ++–

drivers/dahdi/xpp/firmwares/USB_FW.hex     |  6 ++–

drivers/dahdi/xpp/xproto.c                 |  3 +-

include/dahdi/kernel.h                     | 12 +++—-

22 files changed, 93 insertions(+), 84 deletions(-)

The diffstat from the dahdi-tools v2.10.1 release:

xpp/xpp_fxloader | 5 +++–

1 file changed, 3 insertions(+), 2 deletions(-)

For a full list of changes in these releases, please see the shortlog at:

http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=shortlog;h=refs/tags/v2.10.2-rc1

http://git.asterisk.org/gitweb/?p=dahdi/tools.git;a=shortlog;h=refs/tags/v2.10.2-rc1

Issues found in this release can be reported in the DAHDI-Linux [1] and DAHDI-Tools [2] projects at https://issues.asterisk.org/jira

[1] https://issues.asterisk.org/jira/browse/DAHLIN

[2] https://issues.asterisk.org/jira/browse/DAHTOOL

Thank you for your continued support of Asterisk!

Asterisk 13.4.0 Now Available

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:

New Features made in this release:

———————————–

* ASTERISK-24922 – ARI: Add the ability to intercept hold and

     raise an event (Reported by Matt Jordan)

Bugs fixed in this release:

———————————–

* ASTERISK-25112 – Logger: Configuration settings are not reset to

     default during reload. (Reported by Corey Farrell)

* ASTERISK-24944 – main/audiohook.c change prevents G722 call

     recording (Reported by Ronald Raikes)

* ASTERISK-24887 – tags in a=crypto lines do not accept 2

     or more digits (Reported by Makoto Dei)

* ASTERISK-25086 – PJSIP crashes if endpoint missing in

     Dial() (Reported by snuffy)

* ASTERISK-25089 – res_pjsip_config_wizard: Variable specified in

     templates aren’t being processed correctly (Reported by George

     Joseph)

* ASTERISK-25090 – CLI core show channel truncates cdr variables

     (Reported by snuffy)

* ASTERISK-25085 – Potential crash after unload of

     func_periodic_hook or test_message (Reported by Corey Farrell)

* ASTERISK-25083 – Message.c: Message channel becomes saturated

     with frames leading to spammy log messages (Reported by Jonathan

     Rose)

* 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. (Reported by Jonathan Rose)

* ASTERISK-25041 – Broken column type checking in

     res_config_mysql addon (Reported by Alexandre Fournier)

* ASTERISK-21893 – Segfault after call hangup, in

     ast_channel_hangupcause_set, at channel_internal_api.c (Reported

     by Alexandr Gordeev)

* ASTERISK-25074 – Regression: Recent clang-related change broke

     cross compiling of Asterisk (Reported by Sebastian Kemper)

* ASTERISK-25042 – asterisk.conf options override command-line

     options. (Reported by Corey Farrell)

* ASTERISK-24442 – Outgoing call files don’t work properly when

     set in the future (Reported by tootai)

* ASTERISK-25057 – res_pjsip_pubsub: Crash in send_notify due to

     invalid root pointer in sub_tree (Reported by Matt Jordan)

* ASTERISK-24938 – ARI Snoop Channel results in excessive

     escalating CPU usage (Reported by George Ladoff)

* ASTERISK-25034 – chan_dahdi: Some telco switches occasionally

     ignore ISDN RESTART requests. (Reported by Richard Mudgett)

* ASTERISK-25003 – Asterisk crashes on attended transfer (using

     feature) (Reported by Artem Volodin)

* ASTERISK-25038 – Queue log “EXITWITHTIMEOUT” does not always

     contain waiting time (Reported by Etienne Lessard)

* ASTERISK-25027 – Build System: Many ARI modules are missing

     dependencies. (Reported by Corey Farrell)

* ASTERISK-25061 – pbx_config: Register manager actions with

     module version of macro. (Reported by Corey Farrell)

* ASTERISK-25025 – Periodic crashes (in

     ast_channel_snapshot_create at stasis_channels.c) with Certified

     Asterisk 13. (Reported by Chet Stevens)

* ASTERISK-25053 – Unit test category /main/presence missing

     trailing slash. (Reported by Corey Farrell)

* ASTERISK-22708 – res_odbc.conf negative_connection_cache option

     not respected, failover between DSNs doesn’t work (Reported by

     JoshE)

* ASTERISK-25054 – Formats interface’s cannot be unregistered,

     needs to hold modules until shutdown. (Reported by Corey

     Farrell)

* ASTERISK-24896 – Using force black background leads to

     colours not being reset (Reported by dant)

* ASTERISK-25033 – Asterisk 13 (branch head) won’t compile without

     PJSip (Reported by Peter Whisker)

* ASTERISK-25028 – Build System: Unneeded defines in

     asterisk/buildopts.h (Reported by Corey Farrell)

* ASTERISK-25048 – Astobj2: Initialization order wrong when both

     refdebug and AO2_DEBUG are both enabled. (Reported by Corey

     Farrell)

* ASTERISK-19608 – Asterisk-1.8.x  starts rejecting calls with

     cause code 44 after some time. (Reported by Denis Alberto

     Martinez)

* ASTERISK-24976 – cdr_odbc not include new columns added on 1.8

     (Reported by Rodrigo Ramirez Norambuena)

* ASTERISK-25037 – res_pjsip_outbound_registration: Potential

     crash in off-nominal failure case when sending message (Reported

     by Joshua Colp)

* ASTERISK-25022 – Memory leak setting up DTLS/SRTP calls

     (Reported by Steve Davies)

* ASTERISK-22790 – check_modem_rate() may return incorrect rate

     for V.27 (Reported by not here)

* ASTERISK-23231 – Since 405693 If we have res_fax.conf file set

     to minrate=2400, then res_fax refuse to load (Reported by David

     Brillert)

* ASTERISK-24955 – res_fax: v.27ter support baud rate of 2400,

     which is disallowed in res_fax’s check_modem_rate (Reported by

     Matt Jordan)

* ASTERISK-24996 – chan_pjsip: Creating Channel Causes Asterisk to

     Crash When Duplicate AOR Sections Exist in pjsip.conf (Reported

     by Ashley Sanders)

* ASTERISK-25020 – Mismatched response to outgoing REGISTER

     request (Reported by Mark Michelson)

* ASTERISK-25018 – pjsip show endpoints crashes asterisk when

     qualified aors present (Reported by Ivan Poddubny)

* ASTERISK-24749 – ConfBridge: Wrong language on playing

     conf-hasjoin and conf-hasleft when played to bridge (Reported by

     Philippe Bolduc)

* ASTERISK-24845 – pjsip send notify not working with Cisco phone

     (Reported by Carl Fortin)

* ASTERISK-25004 – Crash in authenticated reinvite after

     originated T.38 FAX (Reported by Mark Michelson)

* ASTERISK-24999 – PJSIP crashes with malformed contact line

     (Reported by snuffy)

* ASTERISK-24998 – res_corosync:  res_corosync tries to load even

     if res_corosync.conf is missing (Reported by George Joseph)

* ASTERISK-24997 – Astobj2: Some callers of __adjust_lock do not

     pre-check the object (Reported by Corey Farrell)

* ASTERISK-24982 – res_pjsip_mwi: Unsolicited MWI NOTIFY only sent

     on mailbox changes (Reported by Joshua Colp)

* ASTERISK-24991 – Check for ao2_alloc failure in

     __ast_channel_internal_alloc (Reported by Corey Farrell)

* ASTERISK-24895 – After hangup on the side of the ISDN network no

     HangupRequest event comes for the dahdi channel. (Reported by

     Andrew Zherdin)

* ASTERISK-24977 – Contacts that don’t use qualify are being

     marked as unavailable (Reported by George Joseph)

* ASTERISK-24774 – Segfault in ast_context_destroy with

     extensions.ael and extensions.conf (Reported by Corey Farrell)

* ASTERISK-24841 – ConfBridge: Strange sampling rates chosen when

     channels have multiple native formats (Reported by Matt Jordan)

* ASTERISK-24975 – Enabling ‘DEBUG_THREADLOCALS’ Causes the Build

     to Fail (Reported by Ashley Sanders)

* ASTERISK-24958 – Forwarding loop detection inhibits certain

     desirable scenarios (Reported by Mark Michelson)

* ASTERISK-24863 – res_pjsip: No endpoint events raised via AMI

     when contacts cannot be reached/qualified (Reported by Dmitriy

     Serov)

* ASTERISK-24869 – Asterisk segfaults on DAHDI attended transfer

     due to application (appl) being NULL on unbridged channel

     (Reported by viniciusfontes)

* ASTERISK-24970 – Crash in res_pjsip_pubsub handling of failed

     notify (Reported by Scott Griepentrog)

* ASTERISK-24959 – CLI command cdr show pgsql status

     (Reported by Rodrigo Ramirez Norambuena)

* ASTERISK-24954 – Git migration: Asterisk version numbers are

     incompatible with the Test Suite (Reported by Matt Jordan)

* ASTERISK-17608 – func_aes.so cannot be loaded if res_crypto /

     openssl not compiled (Reported by Warren Selby)

* ASTERISK-24928 – t38_udptl_maxdatagram in pjsip.conf not

     honored (Reported by Juergen Spies)

* ASTERISK-24835 – Early Media Not working with Chan SIP and

     Asterisk 13 (Reported by Andrew Nagy)

* ASTERISK-21777 – Asterisk tries to transcode video instead of

     audio (Reported by Nick Ruggles)

* ASTERISK-24380 – core: Native formats are set to h264 with

     certain audio/video codec configuration, resulting in path

     translation WARNINGs (Reported by Matt Jordan)

* ASTERISK-22352 – IAX2 custom qualify timer is not taken

     into account (Reported by Frederic Van Espen)

* ASTERISK-24894 – iax2_poke_noanswer expiration timer too

     short (Reported by Y Ateya)

* ASTERISK-24935 – res_pjsip_phoneprov_provider: Fix leaked

     OBJ_MULTIPLE iterator. (Reported by Corey Farrell)

* ASTERISK-23319 – Segmentation fault in queue_exec at app_queue.c

     (Reported by Vadim)

* ASTERISK-24933 – T38 fails negotiation (Reported by Jonathan

     Rose)

* ASTERISK-24847 – [security] tcptls: certificate CN NULL

     byte prefix bug (Reported by Matt Jordan)

* ASTERISK-21211 – chan_iax2 – unprotected access of

     iaxs[peer->callno] potentially results in segfault (Reported by

     Jaco Kroon)

* ASTERISK-18032 – – IPv6 and IPv4 NAT not working

     (Reported by Christoph Timm)

* ASTERISK-24782 – StasisEnd event not present for channel that

     was swapped out for another after completing attended transfer

     (Reported by John Bigelow)

* ASTERISK-24910 – “timer=no” and “timer=required” settings in

     pjsip.conf fail (Reported by Ray Crumrine)

* ASTERISK-24932 – Asterisk 13.x does not build with GCC 5.0

     (Reported by Jeffrey C. Ollie)

* ASTERISK-24914 – Division by zero in file.c when playback of

     voicemail with video as h264 (Reported by Marcello Ceschia)

* ASTERISK-24899 – Parking fall-through behavior different in 13

     (Reported by Malcolm Davenport)

* ASTERISK-24937 – res_pjsip_messaging: Messages may be

     sent out of order (Reported by Mark Michelson)

* ASTERISK-24920 – Asterisk handles duplicate SIP requests as if

     they were each a new request (Reported by Mark Michelson)

* ASTERISK-24857 – “timing test”, pjsip incoming/outgoing

     calls, voicemail prompts and recordings all fail when using the

     kqueue timer source on FreeBSD 10.x (Reported by Justin T.

     Gibbs)

* ASTERISK-24155 – Non-portable and non-reliable recursion

     detection in ast_malloc (Reported by Timo Teräs)

* ASTERISK-24142 – CCSS: crash during shutdown due to device

     lookup in destroyed container (Reported by David Brillert)

* ASTERISK-24683 – Crash in PBX ast_hashtab_lookup_internal during

     core restart now (Reported by Peter Katzmann)

* ASTERISK-24805 – – ASAN: Race condition

     (heap-use-after-free) on asterisk closing (Reported by Badalian

     Vyacheslav)

* ASTERISK-24881 – ast_register_atexit should only be used when

     absolutely needed (Reported by Corey Farrell)

* ASTERISK-24731 – res_pjsip_session cannot be unloaded (Reported

     by Corey Farrell)

* ASTERISK-24864 – app_confbridge: file playback blocks dtmf

     (Reported by Kevin Harwell)

* ASTERISK-14233 – Buddies are always auto-registered when

     processing the roster (Reported by Simon Arlott)

* ASTERISK-24780 – – Buddies are always auto-registered

     when processing the roster (Reported by Simon Arlott)

* ASTERISK-24781 – PJSIP: Unnecessary 180 Ringing messages sent

     with undesireabe consequences. (Reported by Richard Mudgett)

Improvements made in this release:

———————————–

* ASTERISK-25044 – sorcery:  Add ability to insert a new wizard

     into an object type’s list (Reported by George Joseph)

* ASTERISK-24892 – Super Awesome Company sound prompts (Reported

     by Rusty Newton)

* ASTERISK-24744 – Swedish Core Voice prompts (Reported by Tove

     Hjelm)

* ASTERISK-25043 – Avoiding ERR_remove_state in OpenSSL

     (Reported by Alexander Traud)

* ASTERISK-25045 – vector:  Add new capabilities and unit tests

     (Reported by George Joseph)

* ASTERISK-24706 – add auto-dtmf mode for pjsip (Reported

     by yaron nahum)

* ASTERISK-25051 – Remove unneeded uses of optional_api providers.

     (Reported by Corey Farrell)

* ASTERISK-25040 – pbx: Improve performance of reloads by making

     hint destruction more performant (Reported by Matt Jordan)

* ASTERISK-24917 – clang compilation warnings (Reported by

     Diederik de Groot)

* ASTERISK-24949 – res_pjsip_outbound_registration: Backport line

     functionality (Reported by Joshua Colp)

* ASTERISK-24965 – cel_pgsql – log_error string references CDR

     instead of CEL (Reported by Rodrigo Ramirez Norambuena)

* ASTERISK-24918 – pjsip: add CLI options to display global and

     system configuration (Reported by Scott Griepentrog)

* ASTERISK-24862 – Support in-dialog OPTIONS (Reported by

     yaron nahum)

* ASTERISK-24802 – stasis: set a channel variable on websocket

     disconnect error (Reported by Kevin Harwell)

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

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

Asterisk 11.18.0 Now Available

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:

Bugs fixed in this release:

———————————–

* ASTERISK-25112 – Logger: Configuration settings are not reset to

     default during reload. (Reported by Corey Farrell)

* ASTERISK-24887 –

tags in a=crypto lines do not accept 2

     or more digits (Reported by Makoto Dei)

* ASTERISK-24944 – main/audiohook.c change prevents G722 call

     recording (Reported by Ronald Raikes)

* ASTERISK-25083 – Message.c: Message channel becomes saturated

     with frames leading to spammy log messages (Reported by Jonathan

     Rose)

* ASTERISK-25041 –

Broken column type checking in

     res_config_mysql addon (Reported by Alexandre Fournier)

* ASTERISK-21893 – Segfault after call hangup, in

     ast_channel_hangupcause_set, at channel_internal_api.c (Reported

     by Alexandr Gordeev)

* ASTERISK-25074 – Regression: Recent clang-related change broke

     cross compiling of Asterisk (Reported by Sebastian Kemper)

* ASTERISK-25042 – asterisk.conf options override command-line

     options. (Reported by Corey Farrell)

* ASTERISK-24442 – Outgoing call files don’t work properly when

     set in the future (Reported by tootai)

* ASTERISK-25034 – chan_dahdi: Some telco switches occasionally

     ignore ISDN RESTART requests. (Reported by Richard Mudgett)

* ASTERISK-25038 – Queue log “EXITWITHTIMEOUT” does not always

     contain waiting time (Reported by Etienne Lessard)

* ASTERISK-22708 – res_odbc.conf negative_connection_cache option

     not respected, failover between DSNs doesn’t work (Reported by

     JoshE)

* ASTERISK-25028 – Build System: Unneeded defines in

     asterisk/buildopts.h (Reported by Corey Farrell)

* ASTERISK-19608 – Asterisk-1.8.x  starts rejecting calls with

     cause code 44 after some time. (Reported by Denis Alberto

     Martinez)

* ASTERISK-24976 – cdr_odbc not include new columns added on 1.8

     (Reported by Rodrigo Ramirez Norambuena)

* ASTERISK-25022 – Memory leak setting up DTLS/SRTP calls

     (Reported by Steve Davies)

* ASTERISK-22790 – check_modem_rate() may return incorrect rate

     for V.27 (Reported by not here)

* ASTERISK-23231 – Since 405693 If we have res_fax.conf file set

     to minrate=2400, then res_fax refuse to load (Reported by David

     Brillert)

* ASTERISK-24955 – res_fax: v.27ter support baud rate of 2400,

     which is disallowed in res_fax’s check_modem_rate (Reported by

     Matt Jordan)

* ASTERISK-24916 – Increasing memory usage when multiple reinvite

     during call (Reported by Christophe Osuna)

* ASTERISK-19538 – Asterisk segfaults on sippeers realtime

     redundancy (Reported by Alex)

* ASTERISK-24749 – ConfBridge: Wrong language on playing

     conf-hasjoin and conf-hasleft when played to bridge (Reported by

     Philippe Bolduc)

* ASTERISK-24991 – Check for ao2_alloc failure in

     __ast_channel_internal_alloc (Reported by Corey Farrell)

* ASTERISK-24895 – After hangup on the side of the ISDN network no

     HangupRequest event comes for the dahdi channel. (Reported by

     Andrew Zherdin)

* ASTERISK-24774 – Segfault in ast_context_destroy with

     extensions.ael and extensions.conf (Reported by Corey Farrell)

* ASTERISK-24975 – Enabling ‘DEBUG_THREADLOCALS’ Causes the Build

     to Fail (Reported by Ashley Sanders)

* ASTERISK-24959 –

CLI command cdr show pgsql status

     (Reported by Rodrigo Ramirez Norambuena)

* ASTERISK-24954 – Git migration: Asterisk version numbers are

     incompatible with the Test Suite (Reported by Matt Jordan)

* ASTERISK-21777 – Asterisk tries to transcode video instead of

     audio (Reported by Nick Ruggles)

* ASTERISK-24380 – core: Native formats are set to h264 with

     certain audio/video codec configuration, resulting in path

     translation WARNINGs (Reported by Matt Jordan)

* ASTERISK-22352 –

IAX2 custom qualify timer is not taken

     into account (Reported by Frederic Van Espen)

* ASTERISK-24894 –

iax2_poke_noanswer expiration timer too

     short (Reported by Y Ateya)

* ASTERISK-23319 – Segmentation fault in queue_exec at app_queue.c

     (Reported by Vadim)

* ASTERISK-24847 – [security]

tcptls: certificate CN NULL

     byte prefix bug (Reported by Matt Jordan)

* ASTERISK-21211 – chan_iax2 – unprotected access of

     iaxs[peer->callno] potentially results in segfault (Reported by

     Jaco Kroon)

* ASTERISK-18032 –

– IPv6 and IPv4 NAT not working

     (Reported by Christoph Timm)

* ASTERISK-24942 – Voicemail API: message is deleted when

     destination mailbox is at maxmsg (Reported by Scott Griepentrog)

* ASTERISK-24932 – Asterisk 13.x does not build with GCC 5.0

     (Reported by Jeffrey C. Ollie)

* ASTERISK-21854 – Long Asterisk-version strings display

     improperly in the ‘Connected to …’ line upon remote console

     connection (Reported by klaus3000)

* ASTERISK-24155 –

Non-portable and non-reliable recursion

     detection in ast_malloc (Reported by Timo Teräs)

* ASTERISK-24142 – CCSS: crash during shutdown due to device

     lookup in destroyed container (Reported by David Brillert)

* ASTERISK-24683 – Crash in PBX ast_hashtab_lookup_internal during

     core restart now (Reported by Peter Katzmann)

* ASTERISK-24805 –

– ASAN: Race condition

     (heap-use-after-free) on asterisk closing (Reported by Badalian

     Vyacheslav)

* ASTERISK-24881 – ast_register_atexit should only be used when

     absolutely needed (Reported by Corey Farrell)

* ASTERISK-24864 – app_confbridge: file playback blocks dtmf

     (Reported by Kevin Harwell)

* ASTERISK-14233 –

Buddies are always auto-registered when

     processing the roster (Reported by Simon Arlott)

* ASTERISK-24780 –

– Buddies are always auto-registered

     when processing the roster (Reported by Simon Arlott)

Improvements made in this release:

———————————–

* ASTERISK-24744 – Swedish Core Voice prompts (Reported by Tove

     Hjelm)

* ASTERISK-25043 –

Avoiding ERR_remove_state in OpenSSL

     (Reported by Alexander Traud)

* ASTERISK-24917 –

clang compilation warnings (Reported by

     Diederik de Groot)

* ASTERISK-25040 – pbx: Improve performance of reloads by making

     hint destruction more performant (Reported by Matt Jordan)

* ASTERISK-24965 – cel_pgsql – log_error string references CDR

     instead of CEL (Reported by Rodrigo Ramirez Norambuena)

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

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

Asterisk 13.3.0 Now Available

The Asterisk Development Team has announced the release of Asterisk 13.3.0.

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

The release of Asterisk 13.3.0 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:

New Features made in this release:

———————————–

* ASTERISK-24703 – ARI: Add the ability to “transfer” (redirect) a

     channel (Reported by Matt Jordan)

* ASTERISK-17899 – Handle crypto lifetime in SDES-SRTP negotiation

     (Reported by Dwayne Hubbard)

Bugs fixed in this release:

———————————–

* ASTERISK-24616 – Crash in res_format_attr_h264 due to invalid

     string copy (Reported by Yura Kocyuba)

* ASTERISK-24748 – res_pjsip: If wizards explicitly configured in

     sorcery.conf false ERROR messages may occur (Reported by Joshua

     Colp)

* ASTERISK-24769 – res_pjsip_sdp_rtp: Local ICE candidates leaked

     (Reported by Matt Jordan)

* ASTERISK-24742 – Fix ast_odbc_find_table function in

     res_odbc (Reported by ibercom)

* ASTERISK-24479 – Enable REF_DEBUG for module references

     (Reported by Corey Farrell)

* ASTERISK-24701 – Stasis: Write timeout on WebSocket fails to

     fully disconnect underlying socket, leading to events being

     dropped with no additional information (Reported by Matt Jordan)

* ASTERISK-24772 – ODBC error in realtime sippeers when device

     unregisters under MariaDB (Reported by Richard Miller)

* ASTERISK-24752 – Crash in bridge_manager_service_req when bridge

     is destroyed by ARI during shutdown (Reported by Richard

     Mudgett)

* ASTERISK-24741 – dtls_handler causes Asterisk to crash (Reported

     by Zane Conkle)

* ASTERISK-24015 – app_transfer fails with PJSIP channels

     (Reported by Private Name)

* ASTERISK-24727 – PJSIP: Crash experienced during multi-Asterisk

     transfer scenario. (Reported by Mark Michelson)

* ASTERISK-24771 – ${CHANNEL(pjsip)} – segfault (Reported by

     Niklas Larsson)

* ASTERISK-24716 – Improve pjsip log messages for presence

     subscription failure (Reported by Rusty Newton)

* ASTERISK-24612 – res_pjsip: No information if a required sorcery

     wizard is not loaded (Reported by Joshua Colp)

* ASTERISK-24768 – res_timing_pthread: file descriptor leak

     (Reported by Matthias Urlichs)

* ASTERISK-24685 – “pjsip show version” CLI command (Reported by

     Joshua Colp)

* ASTERISK-24632 – install_prereq script installs pjproject

     without IPv6 support (Reported by Rusty Newton)

* ASTERISK-24085 – Documentation – We should remove or further

     document the ‘contact’ section in pjsip.conf (Reported by Rusty

     Newton)

* ASTERISK-24791 – Crash in ast_rtcp_write_report (Reported by

     JoshE)

* ASTERISK-24700 – CRASH: NULL channel is being passed to

     ast_bridge_transfer_attended() (Reported by Zane Conkle)

* ASTERISK-24451 – chan_iax2: reference leak in sched_delay_remove

     (Reported by Corey Farrell)

* ASTERISK-24799 – make fails with undefined reference to

     SSLv3_client_method (Reported by Alexander Traud)

* ASTERISK-22670 – Asterisk crashes when processing ISDN AoC

     Events (Reported by klaus3000)

* ASTERISK-24689 – Segfault on hangup after outgoing PRI-Euroisdn

     call (Reported by Marcel Manz)

* ASTERISK-24740 – Segmentation fault on aoc-e event

     (Reported by Panos Gkikakis)

* ASTERISK-24787 – – Microsoft exchange incompatibility

     for playing back messages stored in IMAP – play_message: No

     origtime (Reported by Graham Barnett)

* ASTERISK-24814 – asterisk/lock.h: Fix syntax errors for non-gcc

     OSX with 64 bit integers (Reported by Corey Farrell)

* ASTERISK-24796 – Codecs and bucket schema’s prevent module

     unload (Reported by Corey Farrell)

* ASTERISK-24724 – ‘httpstatus’ Web Page Produces Incomplete HTML

     (Reported by Ashley Sanders)

* ASTERISK-24499 – Need more explicit debug when PJSIP dialstring

     is invalid (Reported by Rusty Newton)

* ASTERISK-24785 – ‘Expires’ header missing from 200 OK on

     REGISTER (Reported by Ross Beer)

* ASTERISK-24677 – ARI GET variable on channel provides unhelpful

     response on non-existent variable (Reported by Joshua Colp)

* ASTERISK-24797 – bridge_softmix: G.729 codec license held

     (Reported by Kevin Harwell)

* ASTERISK-24812 – ARI: Creating channels through /channels

     resource always uses SLIN, which results in unneeded transcoding

     (Reported by Matt Jordan)

* ASTERISK-24800 – Crash in __sip_reliable_xmit due to invalid

     thread ID being passed to pthread_kill (Reported by JoshE)

* ASTERISK-17721 – Incoming SRTP calls that specify a key lifetime

     fail (Reported by Terry Wilson)

* ASTERISK-23214 – chan_sip WARNING message ‘We are requesting

     SRTP for audio, but they responded without it’ is ambiguous and

     wrong in some cases (Reported by Rusty Newton)

* ASTERISK-15434 – When ast_pbx_start failed, both an

     error response and BYE are sent to the caller (Reported by

     Makoto Dei)

* ASTERISK-18105 – most of asterisk modules are unbuildable in

     cygwin environment (Reported by feyfre)

* ASTERISK-24828 – Fix Frame Leaks (Reported by Kevin Harwell)

* ASTERISK-24751 – Integer values in json payload to ARI cause

     asterisk to crash (Reported by jeffrey putnam)

* ASTERISK-24838 – chan_sip: Locking inversion occurs when

     building a peer causes a peer poke during request handling

     (Reported by Richard Mudgett)

* ASTERISK-24825 – Caller ID not recognized using

     Centrex/Distinctive dialing (Reported by Richard Mudgett)

* ASTERISK-24830 – res_rtp_asterisk.c checks USE_PJPROJECT not

     HAVE_PJPROJECT (Reported by Stefan Engström)

* ASTERISK-24840 – res_pjsip: conflicting endpoint identifiers

     (Reported by Kevin Harwell)

* ASTERISK-24755 – Asterisk sends unexpected early BYE to

     transferrer during attended transfer when using a Stasis bridge

     (Reported by John Bigelow)

* ASTERISK-24739 – – Out of files — call fails —

     numerous files with inodes from under /usr/share/zoneinfo,

     mostly posixrules (Reported by Ed Hynan)

* ASTERISK-23390 – NewExten Event with application AGI shows up

     before and after AGI runs (Reported by Benjamin Keith Ford)

* ASTERISK-24786 – – Asterisk terminates when playing a

     voicemail stored in LDAP (Reported by Graham Barnett)

* ASTERISK-24808 – res_config_odbc: Improper escaping of

     backslashes occurs with MySQL (Reported by Javier Acosta)

* ASTERISK-24807 – Missing mandatory field Max-Forwards (Reported

     by Anatoli)

* ASTERISK-20850 – Nested functions aren’t portable.

     Adapting RAII_VAR to use clang/llvm blocks to get the

     same/similar functionality. (Reported by Diederik de Groot)

* ASTERISK-24872 – AMI PJSIPShowEndpoint closes AMI

     connection on error (Reported by Dmitriy Serov)

* ASTERISK-19470 – Documentation on app_amd is incorrect (Reported

     by Frank DiGennaro)

* ASTERISK-21038 – Bad command completion of “core set debug

     channel” (Reported by Richard Kenner)

* ASTERISK-18708 – func_curl hangs channel under load (Reported by

     Dave Cabot)

* ASTERISK-16779 – Cannot disallow unknown format ” (Reported by

     Atis Lezdins)

* ASTERISK-24876 – Investigate reference leaks from

     tests/channels/local/local_optimize_away (Reported by Corey

     Farrell)

* ASTERISK-24882 – chan_sip: Improve usage of REF_DEBUG (Reported

     by Corey Farrell)

* ASTERISK-24817 – init_logger_chain: unreachable code block

     (Reported by Corey Farrell)

* ASTERISK-24880 – Compilation under OpenBSD  (Reported by

     snuffy)

* ASTERISK-24879 – Compilation fails due to 64bit time

     under OpenBSD (Reported by snuffy)

Improvements made in this release:

———————————–

* ASTERISK-24745 – Add no_answer to ARI hangup causes

     (Reported by Ben Merrills)

* ASTERISK-24811 – asterisk-publication sorcery object does not

     use realtime (Reported by Matt Hoskins)

* ASTERISK-24790 – Reduce spurious noise in logs from voicemail –

     Couldn’t find mailbox %s in context (Reported by Graham Barnett)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.3.0