[asterisk-users] Asterisk 13.8.0 Now Available
Marek Červenka
cervajs at fpf.slu.cz
Wed Mar 30 06:51:53 CDT 2016
there is no info about --with-pjproject-bundled
i tried it (centos6 32bit)
./configure --with-pjproject-bundled
checking for SSL_library_init in -lssl... yes
OpenSSL library found, SSL support enabled
./aconfigure: line 14995: syntax error near unexpected token `fi'
./aconfigure: line 14995: `fi'
make: *** [build.mak] Error 2
make: Leaving directory
`/root/rpmbuild/SOURCES/asterisk-13.8.0/third-party/pjproject'
vim ./third-party/pjproject/source/aconfigure
# Check whether --with-opencore-amrnb was given.
if test "${with_opencore_amrnb+set}" = set; then
withval=$with_opencore_amrnb; { { $as_echo "$as_me:$LINENO: error:
This option is obsolete and replaced by --with-opencore-amr=DIR" >&5
$as_echo "$as_me: error: This option is obsolete and replaced by
--with-opencore-amr=DIR" >&2;}
{ (exit 1); exit 1; }; }
else
???missing something???
fi
Dne 30.3.2016 v 0:12 Asterisk Development Team napsal(a):
> The Asterisk Development Team has announced the release of Asterisk 13.8.0.
> This release is available for immediate download at
> http://downloads.asterisk.org/pub/telephony/asterisk
>
> The release of Asterisk 13.8.0 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-24919 - res_pjsip_config_wizard: Ability to write
> contents to file (Reported by Ray Crumrine)
> * ASTERISK-25670 - Add regcontext to PJSIP (Reported by Daniel
> Journo)
> * ASTERISK-25480 - [patch]Add field PauseReason on
> QueueMemberStatus (Reported by Rodrigo Ramirez Norambuena)
>
> Bugs fixed in this release:
> -----------------------------------
> * ASTERISK-25849 - chan_pjsip: transfers with direct media
> sometimes drops audio (Reported by Kevin Harwell)
> * ASTERISK-25113 - install_prereq in Debian 8 without "standard
> system utilities" (Reported by Rodrigo Ramirez Norambuena)
> * ASTERISK-25814 - Segfault at f ip in res_pjsip_refer.so
> (Reported by Sergio Medina Toledo)
> * ASTERISK-25023 - Deadlock in chan_sip in
> update_provisional_keepalive (Reported by Arnd Schmitter)
> * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
> channel (Reported by Filip Frank)
> * ASTERISK-25829 - res_pjsip: PJSIP does not accept spaces when
> separating multiple AORs (Reported by Mateusz Kowalski)
> * ASTERISK-25771 - ARI:Crash - Attended transfers of channels into
> Stasis application. (Reported by Javier Riveros )
> * ASTERISK-25830 - Revision 2451d4e breaks NAT (Reported by Sean
> Bright)
> * ASTERISK-25582 - Testsuite: Reactor timeout error in
> tests/fax/pjsip/directmedia_reinvite_t38 (Reported by Matt
> Jordan)
> * ASTERISK-25811 - Unable to delete object from sorcery cache
> (Reported by Ross Beer)
> * ASTERISK-25800 - [patch] Calculate talktime when is first call
> answered (Reported by Rodrigo Ramirez Norambuena)
> * ASTERISK-25727 - RPM build requires OPTIONAL_API cflag due to
> PJSIP requirement (Reported by Gergely Dömsödi)
> * ASTERISK-25337 - Crash on PJSIP_HEADER Add P-Asserted-Identity
> when calling from Gosub (Reported by Jacques Peacock)
> * ASTERISK-25738 - res_pjsip_pubsub: Crash while executing
> OutboundSubscriptionDetail ami action (Reported by Kevin
> Harwell)
> * ASTERISK-25721 - [patch] res_phoneprov: memory leak and
> heap-use-after-free (Reported by Badalian Vyacheslav)
> * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
> returns garbage (Reported by Etienne Lessard)
> * ASTERISK-25751 - res_pjsip: Support
> pjsip_dlg_create_uas_and_inc_lock (Reported by Joshua Colp)
> * ASTERISK-25606 - Core dump when using transports in sorcery
> (Reported by Martin MouÄka)
> * ASTERISK-20987 - non-admin users, who join muted conference are
> not being muted (Reported by hristo)
> * ASTERISK-25737 - res_pjsip_outbound_registration: line option
> not in Alembic (Reported by Joshua Colp)
> * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
> udptl_rx_packet cause ast_frdup crash (Reported by Walter
> Doekes)
> * ASTERISK-25742 - Secondary IFP Packets can result in accessing
> uninitialized pointers and a crash (Reported by Torrey Searle)
> * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
> Vulnerability - Investigate vulnerability of HTTP server
> (Reported by Alex A. Welzl)
> * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
> non-default timert1 (Reported by Alexander Traud)
> * ASTERISK-25702 - PjSip realtime DB and Cache Errors since
> upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2 (Reported by
> Nic Colledge)
> * ASTERISK-25730 - build: make uninstall after make distclean
> tries to remove root (Reported by George Joseph)
> * ASTERISK-25725 - core: Incorrect XML documentation may result in
> weird behavior (Reported by Joshua Colp)
> * ASTERISK-25722 - ASAN & testsute: stack-buffer-overflow in
> sip_sipredirect (Reported by Badalian Vyacheslav)
> * ASTERISK-25709 - ARI: Crash can occur due to race condition when
> attempting to operate on a hung up channel (Reported by Mark
> Michelson)
> * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
> by Badalian Vyacheslav)
> * ASTERISK-25685 - infrastructure: Run alembic in Jenkins build
> script (Reported by Joshua Colp)
> * ASTERISK-25712 - Second call to already-on-call phone and
> Asterisk sends "Ready" (Reported by Richard Mudgett)
> * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
> (Reported by Badalian Vyacheslav)
> * ASTERISK-25179 - CDR(billsec,f) and CDR(duration,f) report
> incorrect values (Reported by Gianluca Merlo)
> * ASTERISK-25611 - core: threadpool thread_timeout_thrash unit
> test sporadically failing (Reported by Joshua Colp)
> * ASTERISK-24097 - Documentation - CHANNEL function help text
> missing 'linkedid' argument (Reported by Steven T. Wheeler)
> * ASTERISK-25700 - main/config: Clean config maps on shutdown.
> (Reported by Corey Farrell)
> * ASTERISK-25696 - bridge_basic: don't cache xferfailsound during
> a transfer (Reported by Kevin Harwell)
> * ASTERISK-25697 - bridge_basic: don't play an attended transfer
> fail sound after target hangs up (Reported by Kevin Harwell)
> * ASTERISK-25683 - res_ari: Asterisk fails to start if compiled
> with MALLOC_DEBUG (Reported by yaron nahum)
> * ASTERISK-25686 - PJSIP: qualify_timeout is a double, database
> schema is an integer (Reported by Marcelo Terres)
> * ASTERISK-25690 - Hanging up when executing connected line sub
> does not cause hangup (Reported by Joshua Colp)
> * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
> reload' cause a crash (Reported by Sean Bright)
> * ASTERISK-25632 - res_pjsip_sdp_rtp: RTP is sent from wrong IP
> address when multihomed (Reported by Olivier Krief)
> * ASTERISK-25637 - Multi homed server using wrong IP (Reported by
> Daniel Journo)
> * ASTERISK-25394 - pbx: Incorrect device and presence state when
> changing hint details (Reported by Joshua Colp)
> * ASTERISK-25640 - pbx: Deadlock on features reload and state
> change hint. (Reported by Krzysztof Trempala)
> * ASTERISK-25681 - devicestate: Engine thread is not shut down
> (Reported by Corey Farrell)
> * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
> shutdown (Reported by Corey Farrell)
> * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
> Corey Farrell)
> * ASTERISK-25675 - Endpoint not listed as Unreachable (Reported by
> Daniel Journo)
> * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
> by Corey Farrell)
> * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
> Farrell)
> * ASTERISK-25668 - res_pjsip: Deadlock in distributor (Reported by
> Mark Michelson)
> * ASTERISK-25664 - ast_format_cap_append_by_type leaks a reference
> (Reported by Corey Farrell)
> * ASTERISK-25647 - bug of cel_radius.c: wrong point of
> ADD_VENDOR_CODE (Reported by Aaron An)
> * ASTERISK-25317 - asterisk sends too many stun requests (Reported
> by Stefan Engström)
> * 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-25641 - bridge: GOTO_ON_BLINDXFR doesn't work on
> transfer initiated channel (Reported by Dmitry Melekhov)
> * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
> Brandon)
> * ASTERISK-25442 - using realtime (mysql) queue members are never
> updated in wait_our_turn function (app_queue.c) (Reported by
> Carlos Oliva)
> * ASTERISK-25625 - res_sorcery_memory_cache: Add full backend
> caching (Reported by Joshua Colp)
> * ASTERISK-25601 - json: Audit reference usage and thread safety
> (Reported by Joshua Colp)
> * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
> sungtae kim)
>
> Improvements made in this release:
> -----------------------------------
> * ASTERISK-25495 - [patch] Prevent old-update packages on
> repository Debian systems (Reported by Rodrigo Ramirez
> Norambuena)
> * ASTERISK-25846 - Gracefully deal with Absent Stasis Apps
> (Reported by Andrew Nagy)
> * ASTERISK-25791 - res_pjsip_caller_id: Lack of support for
> Anonymous <anonymous at anonymous.invalid> (Reported by Anthony
> Messina)
> * ASTERISK-24813 - asterisk.c: #if statement in listener()
> confuses code folding editors (Reported by Corey Farrell)
> * ASTERISK-25767 - [patch] Add check to configure for sanitizes
> (Reported by Badalian Vyacheslav)
> * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
> core set (Reported by Rusty Newton)
>
> For a full list of changes in this release, please see the ChangeLog:
>
> http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.8.0
>
> Thank you for your continued support of Asterisk!
>
>
>
--
---------------------------------------
Marek Cervenka
=======================================
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