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Вышла новая версия Asterisk 15.1.0

Выпуск Asterisk 15.1.0 содержит много исправлений, сделанных сообществом разработчиков Asterisk и это благодаря участию активных пользователей и фанатов Asterisk!

Скачать Asterisk 15.1.0

В этом выпуске разрешены следующие проблемы:

Исправления, сделанные в этом выпуске:
-----------------------------------

[ASTERISK-27278] -
   [patch] chan_sip: предоставить доступ для чтения полного SIP Request-URI от INVITE
(Сообщается Дэвидом Дж. Прике)
[ASTERISK-27255] -
   alembic: добавление поддержки для сервера Microsoft SQL
(Сообщается Флорианом Фломайром)
[ASTERISK-27253] -
   [patch] Поддержка libsrtp-2.1.x
(Сообщается Александром Траудом)
[ASTERISK-27220] -
   Включить функцию CHANNEL, чтобы получать и тегировать из заголовков SIP
(Сообщается Андре Назарио)
[ASTERISK-27169] -
   Поддержка Google OAuth 2.0 для XMPP / Motif
(Сообщается Андреем)
[ASTERISK-27173] -
   Поддержка GMIME 3.0
(Сообщается Цафриром Коэном)
[ASTERISK-27085] -
   [patch] chan_pjsip: Порт SIPDtmfMode для chan_pjsip
(Сообщается Торри Сирл)

Bugs fixed in this release:
-----------------------------------

[ASTERISK-27346] -
res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
(Reported by Ronald Raikes)
[ASTERISK-27372] -
ARI: Node ARI client broken in latest versions of 13 and 14
(Reported by Benjamin Keith Ford)
[ASTERISK-27047] -
res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
(Reported by dtryba)
[ASTERISK-26988] -
res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
(Reported by dtryba)
[ASTERISK-27270] -
cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
(Reported by Tzafrir Cohen)
[ASTERISK-25266] -
Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
(Reported by Allen Ford)
[ASTERISK-27192] -
res_pjsip: Loss of SIP registrations causing unavailable endpoints
(Reported by Richard Mudgett)
[ASTERISK-27305] -
res_ari: Memory leaks in ARI when using Content-Type: application/json
(Reported by David Hajek)
[ASTERISK-26922] -
chan_sip: tcpbind uses wrong source address
(Reported by Ksenia)
[ASTERISK-27324] -
[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
(Reported by Alexander Traud)
[ASTERISK-27317] -
vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
(Reported by Corey Farrell)
[ASTERISK-27301] -
[patch] app_queue: Music On Hold for real-time queues is not reset to default
(Reported by Nathan Bruning)
[ASTERISK-27318] -
res_pjsip_mwi: uninitialized value from ast_strings_match
(Reported by Corey Farrell)
[ASTERISK-27284] -
Status of RFC 3323 and PJSIP
(Reported by dtryba)
[ASTERISK-27296] -
[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
(Reported by Benoît Dereck-Tricot)
[ASTERISK-27216] -
app_queue: does its check-makeannouncement-logic twice each head-caller-loop
(Reported by Stefan Engström)
[ASTERISK-27298] -
Problem with expires on pjsip / outbound-publish
(Reported by Cyrille Demaret)
[ASTERISK-27295] -
Contact is improperly translated after d178f497
(Reported by Sean Bright)
[ASTERISK-27292] -
Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
(Reported by Ross Beer)
[ASTERISK-27259] -
chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
(Reported by lvl)
[ASTERISK-27289] -
A codeblock that maintains a bug,but maybe the codeblock will never run
(Reported by Huangyx)
[ASTERISK-27277] -
bridge: Renegotiate if source stream changes.
(Reported by Joshua Colp)
[ASTERISK-27283] -
Realtime config fail with PostgreSQL version before 9.1
(Reported by Rodrigo Ramirez Norambuena)
[ASTERISK-27264] -
res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
(Reported by Daniel Heckl)
[ASTERISK-27260] -
[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
(Reported by Daniel Heckl)
[ASTERISK-27257] -
bridge_native_rtp: half-way direct media when using early bridging
(Reported by Jean Aunis — Prescom)
[ASTERISK-16898] -
SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -> 0
(Reported by Marcello Ceschia)
[ASTERISK-27274] -
RTCP needs better packet validation to resist port scans.
(Reported by Richard Mudgett)
[ASTERISK-27252] -
RTP: One way audio with direct media and strictrtp=yes.
(Reported by Richard Mudgett)
[ASTERISK-27279] -
Crash in pubsub_on_rx_request NULL pointer — Possible PJSIP Vulnerability
(Reported by Ross Beer)
[ASTERISK-25524] -
module reload res_calendar.so does not reload everything in calendar.conf
(Reported by Jesper)
[ASTERISK-24588] -
res_calendar does not process CalDAV from Owncloud [fix included]
(Reported by Stefan Gofferje)
[ASTERISK-25523] -
res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
(Reported by Jesper)
[ASTERISK-21399] -
RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
(Reported by Tzafrir Cohen)
[ASTERISK-27248] -
[patch]external_media_address and external_signaling_address don't always honor localnet
(Reported by Walter Doekes)
[ASTERISK-24066] -
res_smdi: convert to astobj2
(Reported by Corey Farrell)
[ASTERISK-27217] -
chan_sip: Asterisk crashing when subscription doesn't get set
(Reported by Bryan Walters)
[ASTERISK-17540] -
SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
(Reported by saghul)
[ASTERISK-27165] -
CDR: CDR (start,u) function won't work in cdr_custom config
(Reported by Jacek Konieczny)
[ASTERISK-27254] -
alembic: prune_on_boot fix erroneous
(Reported by Florian Floimair)
[ASTERISK-27232] -
When in queue on g722 with interruptions, music on hold can get stuck and no longer play
(Reported by Jens T.)
[ASTERISK-27024] -
nat/external_media settings ignored in 14.4.1
(Reported by Christopher van de Sande)
[ASTERISK-26879] -
PJSIP external_media_address ignored if no local_net options are provided
(Reported by Matt Jordan)
[ASTERISK-27236] -
Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
(Reported by Ross Beer)
[ASTERISK-27225] -
Crash when freeing dtls_cfg->cafile
(Reported by Richard Kenner)
[ASTERISK-27177] -
ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
(Reported by Tzafrir Cohen)
[ASTERISK-27241] -
libc segfault upon entry into app_directory
(Reported by David Moore)
[ASTERISK-27152] -
Sending a «tel» uri in a From or To header in an unauthenticated message causes asterisk to crash
(Reported by Ross Beer)
[ASTERISK-27103] -
core: ast_safe_system command injection possible.
(Reported by Corey Farrell)
[ASTERISK-27013] -
res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
(Reported by Joshua Colp)
[ASTERISK-27231] -
res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
(Reported by Joshua Colp)
[ASTERISK-26994] -
Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
(Reported by James Terhune)
[ASTERISK-27222] -
core: Don't queue up multiple video update frames.
(Reported by Joshua Colp)
[ASTERISK-20858] -
app_minivm fails to clean up mkstemp files
(Reported by Walter Doekes)
[ASTERISK-16777] -
several filename bugs in Record () application
(Reported by klaus3000)
[ASTERISK-27168] -
alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
(Reported by Florian Floimair)
[ASTERISK-27209] -
Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
(Reported by Torrey Searle)
[ASTERISK-19103] -
When using realtime queues, function QUEUE_MEMBER_LIST () will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
(Reported by Jim Van Meggelen)
[ASTERISK-21241] -
When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
(Reported by Eelco Brolman)
[ASTERISK-27212] -
bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
(Reported by Richard Mudgett)
[ASTERISK-27204] -
[patch] app_queue: Wrong queue stat calculation
(Reported by sungtae kim)
[ASTERISK-27207] -
XMPP OAuth not working due to inverted logic
(Reported by Michael Kuron)
[ASTERISK-27174] -
res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
(Reported by Mark Thompson)
[ASTERISK-27202] -
If wget is not installed and «or» is not available, external components (excluding pjsip) are not installed
(Reported by Seán C. McCord)
[ASTERISK-27200] -
manager: hook event is not being raised
(Reported by Kevin Harwell)
[ASTERISK-27147] -
Either asterisk or pjproject isn't re-using tcp connections (again)
(Reported by George Joseph)
[ASTERISK-27193] -
IPv6 receive address in message doesn't include brackets
(Reported by Scott Griepentrog)
[ASTERISK-27158] -
[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
(Reported by Torrey Searle)
[ASTERISK-26745] -
Asymmetric codecs when asymmetric_rtp_codec=no
(Reported by Jesse Ross)
[ASTERISK-27189] -
Make --with-pjproject-bundled the default for Asterisk 15
(Reported by George Joseph)
[ASTERISK-27110] -
RTP session is not fully destroyed on channel hangup
(Reported by Matt Jordan)
[ASTERISK-27182] -
bridge: Crash when mapping streams
(Reported by Joshua Colp)
[ASTERISK-27180] -
channel: requester leaks joint_cap on success.
(Reported by Corey Farrell)
[ASTERISK-27179] -
res_pjsip_session: Handling of 'msid' is incorrect
(Reported by Kevin Harwell)
[ASTERISK-27119] -
res_pjsip: parse/add msid attribute when webrtc is enabled
(Reported by Kevin Harwell)
[ASTERISK-27171] -
Asterisk 15.0.0-Beta1 does not compile
(Reported by Ira Emus)
[ASTERISK-26659] -
res_pjsip: PJSIP presence — missing braces around the status element in XML
(Reported by Abraham Liebsch)
[ASTERISK-27156] -
Asterisk won't compile on Fedora 26 with devmode enabled.
(Reported by Corey Farrell)

Новые функции, добавленные в этому версию:
-----------------------------------

[ASTERISK-27215] -
[patch]AMI: Add CancelAtxfer Action
(Reported by Thomas Sevestre)

Полный список изменений в этом выпуске см. в ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0

Благодарим Вас за интерес к продукту Asterisk!

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07.11.2017 г.
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