Aktuelle Asterisk Versionen: 1.2.13 und Zaptel 1.2.10 veröffentlicht

Maik

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Hallo,

heute Morgen sind neue Versionen von Asterisk und Zaptel erschienen:

asterisk-dev Mailingliste schrieb:
The Asterisk Development Team is pleased to announce new releases of
Asterisk and Zaptel!

Asterisk 1.2.12 includes a number of bug fixes, including fixes for two
regressions that occurred in the 1.2.11 release. Specifically, the AGI 'GET
VARIABLE' command has now gone back to its previous behavior, and CDR
records now reflect the CallerID number instead of ANI in the situations
that this was the case in earlier 1.2 releases.

Zaptel 1.2.9 include a small number of bug fixes, an update to the embedded
Octasic API for the VPMOCT echo canceller modules, and reverts to the use of
hardware DTMF detection on VPM400M modules (as it was in pre-1.2.8 releases).

As always, the release files are available on the Digium FTP servers at
http://ftp.digium.com, in both tarball and patch file form. All of the
release files have been signed with our GPG keys and the signature files are
available in the same directories as the release files.

Thanks for using and supporting Asterisk!

Diese Asterisk-Version enthält aber immer noch mindestens einen schweren Bug, den es ab Version 1.2.11 gab. Dieser Bug kann dazu führen, dass nach einem Pickup die Gesprächsdaten im falschen Format weitergeleitet werden und dass man deshalb nur Rauschen hört. Ich gehe daher davon aus, dass in den nächsten Tagen die Version 1.2.12.1 erscheinen wird.

Wer Pickup verwendet, sollte also maximal auf Version 1.2.10 upgraden oder besser gleich abwarten.
 
Bei dem zaptel-Paket auf den Digium-Servern scheint mindestens eine Datei zu fehlen. Ich hänge mal ein Paket an, das alle Dateien enthält.
 

Anhänge

Maik schrieb:
Ich gehe daher davon aus, dass in den nächsten Tagen die Version 1.2.12.1 erscheinen wird.


Asterisk 1.2.12.1 ist draussen. Zu sehen im rechten oberen Eck.
 
Und die nächsten stehen an - zaptel 1.2.10 und Asterisk 1.2.13 - ist nicht wirklich super aktuell...nur vermisste ich den netten Thread dazu ;)

Hier die Changelogs :

Code:
2006-10-17  Kevin P. Fleming <[email protected]>

    * Zaptel 1.2.10 released
    
2006-10-11 17:22 +0000 [r1499]  Joshua Colp <[email protected]>

    * Makefile: Copy to both directories, not just the first one.

2006-10-03 04:50 +0000 [r1493-1496]  Tzafrir Cohen <[email protected]>

    * xpp/README.Astribank: Updated README.Astribank to reflect current
      hardware.

    * xpp/utils/genzaptelconf: * run config function (genconf) in a
      single pass. Bumped version to 0.5.6 . * E1 spans should not be
      configured as "national". TODO: find a better default * Add
      support for Sangoma A102 (Hopefully also all A10x) PRI cards: -d
      won't work, but will configure existing spans. * Better comments
      in genconf() * Removing update_extensions_defs: cleaning
      temporary files is only done once.

    * xpp/utils/genzaptelconf: * fixed a small syntax error * Hopefully
      correct PRI setting for country=cl

    * xpp/utils/genzaptelconf: Allow overriding the location of the
      asterisk init.d script using the variable ASTERISK_INIT_SCRIPT
      (from the config file or from the environment).

2006-09-16 07:45 +0000 [r1468]  Tilghman Lesher <[email protected]>

    * zaptel.c: Round two of the fix for "hard-safe -> hard-unsafe lock
      order detected" (Issue 7620) Only difference is that the fix is
      only defined for kernel versions 2.6.11 and above; older kernels
      keep the current (possible deadlocking) code.

2006-09-11  Kevin P. Fleming <[email protected]>

    * Zaptel 1.2.9.1 released

und

Code:
2006-10-17 Kevin P. Fleming <[email protected]>

    * Asterisk 1.2.13 released

2006-10-17 20:37 +0000 [r45380]  Joshua Colp <[email protected]>

    * channels/chan_sip.c: Don't create a "real" pvt structure for
      requests that shouldn't be able to create one. Instead use a
      temporary pvt and fill it with enough information so we can send
      a reply.

2006-10-17 17:50 +0000 [r45332]  Jason Parker <[email protected]>

    * channels/chan_skinny.c: Fix an integer signedness problem.

2006-10-17 17:22 +0000 [r45326]  Kevin P. Fleming <[email protected]>

    * LICENSE: provide licensing language for IAXy firmware file

2006-10-17 15:50 +0000 [r45306]  Olle Johansson <[email protected]>

    * channels/chan_sip.c, configs/sip.conf.sample: After some
      research, we realized that the default behaviour since a long
      time was doing the right thing, even though the change optimized
      a bit and removed a lot of potential risks. Conclusion: No need
      for a configuration option at all.

2006-10-16 19:59 +0000 [r45260-45265]  Joshua Colp <[email protected]>

    * channels/chan_sip.c, configs/sip.conf.sample: Use responses
      rather then replies even though they mean the same thing.

    * channels/chan_sip.c, configs/sip.conf.sample: Add
      'ignoreoodreplies' option which will not create a pvt structure
      on a SIP response but instead basically drop it.

2006-10-14 00:16 +0000 [r45134]  Steve Murphy <[email protected]>

    * pbx/pbx_ael.c: Made a small update to solve bug 8128; The
      switch-case fallthru goto to a pattern extension needed to
      resolved the wildcards to an appropriate digit for extension
      matching to work

2006-10-13 22:57 +0000 [r45119]  Kevin P. Fleming <[email protected]>

    * acl.c: don't drop the entire permit/deny list when an attempt is
      made to add an invalid entry (BE-92)

2006-10-13 19:27 +0000 [r45090]  Christian Richter <[email protected]>

    * channels/chan_misdn.c: avoiding warning, fixing potential bug
      (backported from 1.2)

2006-10-13 17:01 +0000 [r45060]  Joshua Colp <[email protected]>

    * apps/app_chanspy.c: Turn on volume adjustment if it needs to be
      on (issue #8136 reported by mnicholson)

2006-10-13 16:18 +0000 [r45048]  Kevin P. Fleming <[email protected]>

    * channels/chan_iax2.c: when sending a call to a peer, use the
      proper socket if we have multiple bindings (reported on
      asterisk-dev)

2006-10-13 15:49 +0000 [r45030]  Joshua Colp <[email protected]>

    * dnsmgr.c: Pass the right value to usleep for sleeping, and always
      add the background refresh item back into the scheduler if
      enabled since it is deleted during reload. (issue #8142 reported
      by p_lindheimer)

2006-10-13 13:11 +0000 [r44993-45020]  Christian Richter <[email protected]>

    * channels/chan_misdn.c, channels/misdn/isdn_lib.c: fixed some
      echocandisable issues when bridged. this caused a kernel panic
      sometimes..also some minor formatting fixes

    * channels/misdn/isdn_msg_parser.c: fixed issue, that the
      hangupcause got a wrong isdn cause at RELEASE_COMPLETE

2006-10-12 18:31 +0000 [r44955]  Kevin P. Fleming <[email protected]>

    * include/asterisk/utils.h, channels/chan_sip.c, utils.c,
      netsock.c: ensure that IAX2 and SIP sockets allow UDP
      fragmentation when running on Linux (thanks to Brian Candler on
      the asterisk-dev list for the tip)

2006-10-10 13:34 +0000 [r44785]  Christian Richter <[email protected]>

    * channels/chan_misdn.c, channels/misdn/isdn_lib.c: (re)added
      support of dynamical enabling hdlc on bchannels

2006-10-09 14:36 +0000 [r44757]  Olle Johansson <[email protected]>

    * channels/chan_sip.c: Issue #8101 - wrong parameter for screening
      in remote-party-id

2006-10-06 16:52 +0000 [r44501-44580]  Joshua Colp <[email protected]>

    * file.c: Even more frames to treat as though the remote side
      disappeared (issue #8097 reported by eldadran)

    * file.c: Treat busy control frames as hangup in the file streaming
      core (issue #8097 reported by eldadran)

2006-10-05 10:02 +0000 [r44460]  Christian Richter <[email protected]>

    * channels/chan_misdn.c: fixed segfault which happens during
      hold/transfer action

2006-10-05 01:27 +0000 [r44392-44432]  Kevin P. Fleming <[email protected]>

    * channels/chan_sip.c: fix Polycom presence notification again

    * channels/chan_sip.c: remove workaround for old Polycom firmware
      SUBSCRIBE requests add workaround for new Polycom firmware
      SUBSCRIBE requests (bug is known to exist in 2.0.1 firmware)

2006-10-04 16:02 +0000 [r44343]  Steve Murphy <[email protected]>

    * apps/app_macro.c: For bug 7776, I have inserted a warning about
      Macro nesting vs. stack limitations

2006-10-04 15:26 +0000 [r44334-44335]  Christian Richter <[email protected]>

    * channels/chan_misdn.c: if INFORMATION Message come with keypad
      instead of called party number, we just use the keypad as called
      party number.

    * channels/misdn_config.c, channels/misdn/isdn_lib.h,
      channels/chan_misdn.c, channels/misdn/chan_misdn_config.h,
      configs/misdn.conf.sample, channels/misdn/isdn_lib.c: added the
      option 'reject_cause' to make it possible to set the
      RELEASE_COMPLETE - cause on the 3. incoming PMP channel, which is
      automatically rejected because chan_misdn does not support that
      kind of callwaiting. Therefore chan_misdn supports now 3 incoming
      channels on a PMP BRI Port. misdn_lib_get_free_bc now gets the
      info if the requested channel is incoming or outgoing to make the
      3. channel possible

2006-10-03 20:14 +0000 [r44296]  Kevin P. Fleming <[email protected]>

    * apps/app_queue.c: fix a logic error in my previous fix to the
      queue reload code

2006-10-02 20:07 +0000 [r44168-44213]  Joshua Colp <[email protected]>

    * channels/chan_sip.c: Change the fd on the I/O context in case it
      changed during the reload, which is indeed possible. (issue #7943
      reported by eclubb)

    * contrib/init.d/rc.redhat.asterisk: We should be using $AST_SBIN
      instead of hardcoding the path for the error message (issue #7942
      reported by eclubb)

    * io.c: Shrink when current_ioc is unused. It is set to -1 when
      unused, not 0. (issue #7941 reported by eclubb)

2006-10-02 13:28 +0000 [r44149]  Christian Richter <[email protected]>

    * channels/misdn/isdn_lib.h, channels/chan_misdn.c,
      channels/misdn/isdn_lib.c: fixed the hold/retrieve/transfer
      issues, removed a useless bc field, added setting of
      frame.delivery fields, some minor code cleanups

2006-10-01 15:19 +0000 [r44110]  Russell Bryant <[email protected]>

    * configs/queues.conf.sample: Fix the name of the
      "eventmemberstatus" option in the sample queues.conf (issue
      #8065, adamg)

2006-09-29 13:44 +0000 [r43977]  Kevin P. Fleming <[email protected]>

    * cli.c: proper fix for ast_group_t change

2006-09-28 18:00 +0000 [r43924]  Joshua Colp <[email protected]>

    * frame.c, include/asterisk/logger.h, channels/chan_misdn.c,
      channels/chan_sip.c, channels/chan_skinny.c,
      funcs/func_timeout.c, apps/app_festival.c, res/res_features.c,
      apps/app_hasnewvoicemail.c, apps/app_alarmreceiver.c,
      channels/iax2-provision.c, res/res_musiconhold.c,
      res/res_monitor.c: Put in missing \ns on the end of ast_logs
      (issue #7936 reported by wojtekka)

2006-09-28 17:31 +0000 [r43916]  Kevin P. Fleming <[email protected]>

    * apps/app_queue.c: fix buggy (and overly complex) loop used during
      reload of app_queue for static member list updating

2006-09-28 16:37 +0000 [r43897]  BJ Weschke <[email protected]>

    * apps/app_queue.c: app_queue is comparing the device names
      incorrectly while checking their statuses. It's internal list of
      interfaces includes the dial string, while the argument passed to
      this function does not have the dial string (/n for a local
      channel). This causes it to ignore the device state changes
      because it thinks it belongs to none of its members. (#8040
      reported and patch by tim_ringenbach)

2006-09-28 16:32 +0000 [r43895]  Kevin P. Fleming <[email protected]>

    * cli.c: eliminate compiler warning introduced by recent changes

2006-09-28 16:13 +0000 [r43891]  Joshua Colp <[email protected]>

    * apps/app_meetme.c: Stop the stream after waitstream returns so
      that our formats get restored. (issue #7370 reported by
      kryptolus)

2006-09-28 15:18 +0000 [r43871]  BJ Weschke <[email protected]>

    * apps/app_queue.c: Fix race condion crash with get_member_status
      (#7864 - tim_ringenbach reported and patched)

2006-09-27 20:20 +0000 [r43815]  Tilghman Lesher <[email protected]>

    * apps/app_voicemail.c: Avoid inability to lock directory log
      message by creating the directory ahead of time. (Issue 7631)

2006-09-27 19:35 +0000 [r43800]  Jason Parker <[email protected]>

    * apps/app_playback.c, pbx.c: Playback() wasn't setting
      PLAYBACKSTATUS under several circumstances. Playback() returns -1
      on missing args - so should Background()

2006-09-27 16:54 +0000 [r43778]  Russell Bryant <[email protected]>

    * res/res_features.c, channel.c: Fix a problem that occurred if a
      user entered a digit that matched a bridge feature that was
      configured using multiple digits, and the digit that was pressed
      timed out in the feature digit timeout period. For example, if
      blind transfer is configured as '##', and a user presses just
      '#'. In this situation, the call would lock up and no longer pass
      any frames. (issue #7977 reported by festr, and issue #7982
      reported by michaels and valuable input provided by mneuhauser
      and kuj. Fixed by me, with testing help and peer review from
      Joshua Colp). There are a couple of issues involved in this fix:
      1) When ast_generic_bridge determines that there has been a
      timeout, it returned AST_BRIDGE_RETRY. Then, when
      ast_channel_bridge gets this result, it calls ast_generic_bridge
      over again with the same timestamp for the next event. This
      results in an endless loop of nothing until the call is
      terminated. This is resolved by simply changing
      ast_generic_bridge to return AST_BRIDGE_COMPLETE when it sees a
      timeout. 2) I also changed ast_channel_bridge such that if in the
      process of calculating the time until the next event, it knows a
      timeout has already occured, to immediately return
      AST_BRIDGE_COMPLETE instead of attempting to bridge the channels
      anyway. 3) In the process of testing the previous two changes, I
      ran into a problem in res_features where ast_channel_bridge would
      return because it determined that there was a timeout. However,
      ast_bridge_call in res_features would then determine by its own
      calculation that there was still 1 ms before the timeout really
      occurs. It would then proceed, and since the bridge broke out and
      did *not* return a frame, it interpreted this as the call was
      over and hung up the channels. The reason for this was because
      ast_bridge_call in res_features and ast_channel_bridge in
      channel.c were using different times for their calculations.
      channel.c uses the start_time on the bridge config, which is the
      time that the feature digit was recieved. However, res_features
      had another time, 'start', which was set right before calling
      ast_channel_bridge. 'start' will always be slightly after
      start_time in the bridge config, and sometimes enough to round up
      to one ms. This is fixed by making ast_bridge_call use the same
      time as ast_channel_bridge for the timeout calculation.

2006-09-27 12:51 +0000 [r43764]  Christian Richter <[email protected]>

    * channels/misdn/isdn_lib.h, channels/chan_misdn.c,
      channels/misdn/isdn_lib.c: fixed a bug which led to chan_list
      zombies, when the call could not be properly established in
      misdn_call. also removed the ACK_HDLC stuff which is not really
      needed.

2006-09-26 20:49 +0000 [r43708]  Russell Bryant <[email protected]>

    * asterisk.c: Back in revision 4798, this message was changed from
      using ast_cli() to directly calling write(). During this change,
      checking if this was a remote console was removed. This caused
      this message about using "exit" or "quit" to exit an Asterisk
      console to come up in times where it did not make sense. This
      change restores the check to see if this is a remote console
      before printing the message. (fixes BE-4)

2006-09-26 20:38 +0000 [r43705-43706]  Joshua Colp <[email protected]>

    * .cleancount: I changed the channel structure... let's be sure
      this gets updated!

    * channels/chan_sip.c, include/asterisk/channel.h: Use proper type
      to represent the group variable (issue #8025 reported by makoto)

2006-09-26 20:23 +0000 [r43699]  Russell Bryant <[email protected]>

    * apps/app_voicemail.c: When parsing the sections of voicemail.conf
      that contain mailbox definitions, don't introduce a length limit
      on the definition by using a 256 byte temporary storage buffer.
      Instead, make the temporary buffer just as big as it needs to be
      to hold the entire mailbox definition. (fixes BE-68)

2006-09-25 21:14 +0000 [r43634]  Tilghman Lesher <[email protected]>

    * apps/app_voicemail.c: Two bugs when forwarding voicemail (Issue
      7824): 1) delete=yes was ignored 2) maxmessages was ignored

2006-09-24 13:50 +0000 [r43552]  Russell Bryant <[email protected]>

    * channels/chan_iax2.c: Check to see if the channel that is
      activating the IAXPEER function is actually an IAX2 channel
      before proceeding to process it to avoid crashing. (issue #8017,
      reported by admott, fixed by myself)

2006-09-22 21:53 +0000 [r43509]  Joshua Colp <[email protected]>

    * apps/app_chanspy.c, channel.c: Yay another 'round of spy fixes!
      This fixes a small logic flaw with the cleanup function and a
      memory allocation issue. (issue #7960 reported by jojo & issue
      #7999 reported by aster1) Special thanks to csum77 for letting me
      into a box where this issue was happening.

2006-09-21 17:01 +0000 [r43409-43420]  Tilghman Lesher <[email protected]>

    * apps/app_rpt.c: Whitespace change... really just an excuse to
      test repotools

    * cdr/cdr_tds.c, cdr/Makefile: TDS 0.64 updates

2006-09-20 05:08 +0000 [r43314]  Kevin P. Fleming <[email protected]>

    * channels/chan_misdn.c, channels/chan_sip.c,
      channels/chan_skinny.c: make some more functions static

2006-09-19 16:21 +0000 [r43269]  Matt O'Gorman <[email protected]>

    * pbx/pbx_gtkconsole.c, apps/app_dial.c, channels/chan_sip.c,
      apps/app_macro.c, asterisk.c, config.c, apps/app_queue.c, pbx.c:
      fixes some verbose vs debug issues. patch from bug 2617

2006-09-19 12:28 +0000 [r43248]  Tilghman Lesher <[email protected]>

    * apps/app_voicemail.c: cid is passed to a destructive function;
      thus a copy is needed (issue 7961)

2006-09-18 20:08 +0000 [r43220]  Olle Johansson <[email protected]>

    * channels/chan_sip.c: Issue #7682 - don't add contacts to 4xx
      responses. (Ugly fix, not proud at all)

2006-09-18 15:30 +0000 [r43163]  Joshua Colp <[email protected]>

    * apps/app_math.c: Add deprecation notice about app_math (issue
      #7957 reported by k-egg)

2006-09-18 15:05 +0000 [r43160]  Steve Murphy <[email protected]>

    * configs/zapata.conf.sample: Clarified what "callwaiting" does in
      zapata.conf.

2006-09-18 15:05 +0000 [r43159]  Joshua Colp <[email protected]>

    * configs/indications.conf.sample: Add number unobtainable tone for
      New Zealand (issue #7969 reported by nic_bellamy)

2006-09-17 13:54 +0000 [r43072]  Tilghman Lesher <[email protected]>

    * apps/app_directory.c: Directory used the wrong context for
      delivery of 0- and *- keypresses (according to Directory's own
      documentation) - Issue 7965

2006-09-16 07:57 +0000 [r43003-43019]  Tilghman Lesher <[email protected]>

    * channels/chan_iax2.c: When a realtime peer expires, reset the
      ipaddress in the realtime database back to 0 (Issue 6656)

    * apps/app_meetme.c: When the marked user enters the conference, we
      should no longer timeout

2006-09-14 22:16 +0000 [r42946]  Tilghman Lesher <[email protected]>

    * channels/chan_zap.c: Error message references wrong argument
      (Issue 7951)

2006-09-13 19:51 +0000 [r42892]  Tilghman Lesher <[email protected]>

    * apps/app_voicemail.c: Backport bugfix patch from 7918 to 1.2 -
      msg_cfg destroyed before used

2006-09-11 Kevin P. Fleming <[email protected]>

Mit dem Hinweis :

Code:
The Asterisk Development team has released an update to Asterisk 1.2, Asterisk 1.2.13.
 This release contains a fix for a security vulnerability recently found in the chan_skinny channel driver (for Cisco SCCP phones). This vulnerability would enable an attacker to remotely execute code as the system user running Asterisk (frequently 'root'). The exploit does not require that the skinny.conf contain any valid phone entries, only that chan_skinny is loaded and operational.
 This release also contains a number of bug fixes, and some improvements to the chan_sip channel driver (for SIP devices) to mitigate the impacts of a certain class of denial-of-service attacks that have recently been published.
 
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