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Door , , 3 reacties
Bron: Digium

Asterisk is een uitgebreide PBX voor het BSD-, Linux- en Mac OS X-platform. Het programma biedt alle functies die je van een telefooncentrale mag verwachten. Het beschikt onder andere over mogelijkheden voor voicemail, conferencing en call queuing. Daarnaast is er ondersteuning voor caller ID services, ADSI, SIP en H.323 aanwezig. Voor een compleet overzicht van alle mogelijkheden verwijzen we jullie door naar deze pagina. De ontwikkelaars hebben een nieuwe versie uitgebracht die 1.2.11 als het versienummer heeft meegekregen. De lijst met veranderingen ziet er als volgt uit:

Version 1.2.11:
  • apps/app_random.c: Bug 7779 - Using initstate(3) means that we cannot unload this module once loaded.
  • asterisk.c: Move the load_modules call so that if a module needs realtime support it will work, none do currently but a good move none the less.
  • CREDITS: Reformat to match the contribution style of other contributors
  • channels/chan_sip.c: Turn media level c= parsing on by default (issue #7725 reported by psm)
  • apps/app_voicemail.c, apps/app_directory.c: Fix a bug with app_voicemail when trying to use app_directory to leave messages to another user (options 3, 5, 2). If the context/extension didn't exist in the dialplan (and why should it have to?), it would fail, saying that it's an "invalid extension". Fix was different in svn trunk. (issue BE-71)
  • configs/zapata.conf.sample: make a feeble attempt to avoid the 'how do I enable my hardware echo canceler' questions
  • channels/misdn_config.c (added), channels/chan_misdn_config.c (removed): rename file per crichter's request
  • doc/README.misdn, channels/misdn/mISDN.patch (removed), channels/misdn/isdn_lib.h, channels/chan_misdn.c, channels/misdn/fac.c (added), channels/misdn/Makefile, channels/misdn/chan_misdn_config.h, channels/misdn/ie.c, channels/misdn/fac.h (added), channels/misdn/portinfo.c (removed), channels/misdn/isdn_lib_intern.h, channels/chan_misdn_config.c, channels/misdn/isdn_msg_parser.c, configs/misdn.conf.sample, channels/Makefile, channels/misdn/isdn_lib.c: This rather small ;-) commit merges the changes from my team branch 0.3.0 into t he 1.2 branch. These changes include the new mISDN mqueue interface which makes it possible to compile chan_misdn against the current cvs version of mISDN/mISDNuser. These changes also contain various additions and numerous bugfixes to chan_misdn . Each change is documented in the commit logs in the team/crichter/0.3.0 branch.
  • channel.c: revert bogus change to attempt to fix bug 7506 which actually causes half of the channels not to get "Newchannel" events at all (issue #7745)
  • funcs/func_cdr.c: Use the last CDR entry instead of the first CDR entry for variable retrieving variables using the CDR dialplan function. (issue #7689 reported by voipgate)
  • apps/app_macro.c: Make app_macro compile again
  • apps/app_macro.c: In app_macro, changed the previously changed upper recursion depth limit to a variable, default of the original val of 7. MACRO_RECURSION is a channel variable that will override the limit, but until I can understand and fix why this limit is neccessary, I am not advertising this variable in the docs. This fix mirrors the changes made in r40200 in trunk.
  • channels/chan_mgcp.c: don't allow AUEP responses to overflow the stack during a string copy (reported by Mu Security)
  • res/res_agi.c: use pbx_builtin_getvar_helper() so that GET VARIABLE can retrieve global variables (issue #7609)
  • apps/app_macro.c: This revision fixes bug 7731, the inability for macros to be called more than one level deep in the 'h' extension. It also pushes up the limit of recursion depth from 7 to 20.
  • CREDITS: add explicit listing of anthm's contributions (issue #7683)
  • channels/chan_sip.c: Increase the buffer size for the callid (issue #7675, reported by pssatcs)
  • channels/chan_zap.c: Fix a crash reported to me by hads on IRC. This crash would occur with the use of the "distinctiveringaftercid" option. Also, on this user's system, the crash would only occur when built without optimizations. This is because the bug is that the code would write past the end of an array that was allocated on the stack, and the structure of the stack is different with or without optimizations enabled.
  • channel.c: Reset our stream and vstream pointers back to NULL so that any generator that uses them (file based MOH) will not try to close them again. (issue #7668 reported by jmls)
  • channel.c: Always generate a Newstate event in ast_setstate() instead of making it a Newchannel event if the state was AST_STATE_DOWN. The Newchannel event will always be generated in ast_request(), so this just causes a duplicated Newchannel event in some cases. (issue #7506, repoted by capouch, fixed by me)
  • apps/app_queue.c: remove duplicate queue log entry when the caller exits on a timeout (issue #7616, ppyy)
  • channels/chan_sip.c: don't advertise that this function can set a SIP header when it can only do reads
  • apps/app_dial.c: make sure the priv-callerintros directory exists before trying to create a file there (issue #7659, patch by hads, with some modifications by me)
  • channels/chan_mgcp.c, channels/chan_vpb.c, channels/chan_phone.c, channels/chan_misdn.c, channels/chan_zap.c, channels/chan_sip.c, channels/chan_skinny.c, channels/chan_h323.c, channels/chan_modem.c, channels/chan_iax2.c: Fix an issue that would cause a NewCallerID manager event to be generated before the channel's NewChannel event. This was due to a somewhat recent change that included using ast_set_callerid() where it wasn't before. This function should not be used in the channel driver "new" functions. (issue #7654, fixed by me) Also, fix a couple minor bugs in usecount handling. chan_iax2 could have increased the usecount but then returned an error. The place where chan_sip increased the usecount did not call ast_update_usecount()
  • channel.c: suppress a compiler warning about the usage of a potentially uninitialized variable
  • res/res_musiconhold.c: Treat the file as invalid if we have no valid formats for it (issue #7643 reported by KNK)
  • apps/app_voicemail.c: Bug 7648 - Checking wrong count for plurality on new messages for Dutch language
  • channels/chan_sip.c: fix brain-damage I introduced when trying to fix the CANCEL/BYE sending mechanism for pending INVITES accept unknown 1xx responses as 183 responses (as RFC3261 mandates we should do)
  • res/res_features.c, channel.c: ensure that the 'feature digit timeout' value is taken into account when deciding how long the bridge should run (this fixes a problem report where a digit press that did not invoke a feature is never passed across the bridge)
  • res/res_musiconhold.c: Close the stream when file based MOH stop. This won't get rid of their position in the file but it will cause the translation path to be setup again. (issue #7634 reported by asimpson)
  • channels/chan_sip.c: don't reissue hangup requests for SIP channels that have expired their RTP timeouts (one time is enough) don't rescan the SIP private structure list too fast, it can cause channels to not be able to hang up (issue #7495, and probably others) use ast_softhangup_nolock() since we already hold the channel's lock
  • res/res_features.c: Add missing code to bring transferee channel out of MOH/autoservice under certain circumstance (issue #7611 reported by guillecabeza with minor mods by myself)
  • frame.c: one more small tweak for thread-safety of TRACE_FRAMES
  • frame.c: Make the frame counting done with TRACE_FRAMES defined thread-safe
  • channels/chan_sip.c: How many attempts does it take to make a SIP URI parser that works well? I'm up to 5 personally. On to the good stuff - parse the domain first, user second, and get rid of port & options/params last. (issue #7616 reported by andrew)
  • channels/chan_sip.c: Make a copy of the request URI in check_user_full instead of modifying the one on the structure, and also strip params properly from the user portion of the SIP URI so as to preserve the domain (issue #7552 reported by dan42)
  • apps/app_chanspy.c: use the enum that defines the option arguments, so that the likelihood of mismatched option indexes is reduced (which in this case was a bug, the volume argument was not checked properly)
  • channel.c: do a better job avoiding translation path teardown/setup when not needed
  • channels/chan_iax2.c: Fix crash when using the "regexten" option with MALLOC_DEBUG enabled. This was not reported in the bug tracker but the same bug has been demonstrated in other places in the code.
  • channel.c: don't do useless translation destroy/build when the channel is already in the correct format
  • channels/chan_sip.c: fix a crash when MALLOC_DEBUG is enabled and the regexten is enabled. The crash would occur when the extension got removed. (fixes issue #7484)
  • channels/chan_sip.c: Put default callerid into contact when the one specified is either NULL or has a zero string length. (issue #7590 reported by key2)
  • channels/chan_zap.c: This resolves a deadlock that a tech support customer was getting frequently when his users would answer call waiting. If another thread is currently holding the zt_pvt lock for the first channel, unlock both channels and let asterisk retry the native bridge, just like what is done for the second channel directly below these changes.
  • codecs/gsm/Makefile: This fixes a compile problem for s390 as reported in bug 7253. Tested on both an s390 and non-s390 machine.
  • channels/chan_iax2.c: ensure that global 'maxauthreq' is reset to zero during 'reload'
  • frame.c: don't crash if the frame has no data, but has a src
  • frame.c: if asked to duplicate a frame that has no data, don't set the frame's data pointer past the end of the allocatted buffer for the new frame
  • formats/format_h263.c: Backport buffer increase to 1.2
  • formats/format_h263.c: Overflow bad
  • enum.c: Bug 7513 - ensure that each time we do a query, the results are returned in the same logical order, so that when we iterate over the list, we get all results, not some results repeated, due to insufficient sorting.
Version 1.2.10:
  • apps/app_sms.c: Bug 7526 - previous commit broke app_sms
  • apps/app_voicemail.c: don't fail/abort if the message category sound file cannot be played, just generate a warning message and continue message playback
  • rtp.c: yeah, ummm... This frame pointer should not be static. This situation only exists in 1.2 (pointed out by Constantine Filin on the asterisk-dev mailing list)
  • channels/chan_sip.c: report address of peer trying to subscribe to unknown hint
  • doc/README.enum: Bug 7532 - Typo in enum example
  • contrib/init.d/rc.mandrake.zaptel: Merge fixup for asterisk startup script to zaptel startup script
  • apps/app_voicemail.c: fix a weird case where a lock file could be left (but would happen almost never)
  • app.c: fix a case where ast_lock_path() could leave a randomly-named lock file hanging around make ast_unlock_path actually report when unlocking fails
  • channels/chan_iax2.c: Add support to have maxauthreq as a global option
  • channels/chan_zap.c, utils.c, res/res_agi.c, apps/app_zapras.c, asterisk.c, channels/chan_modem.c, channels/chan_iax2.c: remove some more bad examples of using printf
  • enum.c, pbx/pbx_config.c: get rid of some more printf's (although most of these were ifdef-ed out)
  • app.c: GRRR no fprintf!
  • configs/iax.conf.sample, channels/chan_iax2.c: Add configuration option for IAX2 users that will limit the amount of outstanding AUTHREQs we are waiting for replies on.
  • channel.c: do masquerade-behind-proxy checking with better control over locks
  • rtp.c: Change message regarding marker bit forcing when SSRC changes to be shown only during debug so it doesn't overload high capacity systems
  • channel.c: patch resolves issue with when to decide if its right time to native bridge, feature redirect was not being checked. patch from bug #7296
  • channels/chan_agent.c: Don't do weird things on a callback agent that has attempted logoff while still on the phone.
  • channels/chan_sip.c: Instead of giving the scheduled item ID on a peer expiration, give the time until they expire (issue #7455 reported by slavon)
  • funcs/func_db.c: Fix spelling/grammar (issue 7493)
  • channels/chan_oss.c: Spell extension correctly in documentation for chan_oss dial (issue #7487 reported by flefoll)
  • channels/chan_sip.c: Tell clients based on old SIP standard that we only support MD5 digest authentication...
  • channels/chan_sip.c: issue #7470 - Need larger buffer for record-route headers...
  • asterisk.c: fix a race condition that caused asterisk to log a *ton* of warnings on mac osx about poll returning an error because the polled file descriptor was bad.
  • channels/chan_mgcp.c, channels/chan_phone.c, channels/chan_local.c, channels/chan_misdn.c, channels/chan_sip.c, channels/chan_skinny.c, channels/chan_agent.c, channels/chan_features.c, channels/chan_h323.c, channels/chan_modem.c, channels/chan_iax2.c: use ast_set_callerid to be more consistent and to make sure that the "callerid" option in the conf files is always handled the same way and sets ANI (issue #7285, gkloepfer)
  • dsp.c: fix the build with BUSYDETECT_TONEONLY defined (issue #7414)
  • apps/app_directory.c: Bug 7349 - Directory did not work correctly when USE_ODBC_STORAGE was defined.
  • Makefile: Bug 7388 - compatibility changes for Solaris
  • configs/queues.conf.sample: clarify documentation for 'persistentmembers' setting
  • configs/sip.conf.sample: add documentation for peer-specific 'outboundproxy' setting
  • channels/chan_sip.c: Don't delete scheduled item twice in sip_destroy (already fixed in svn trunk)
  • channels/chan_sip.c: ensure that two SIP channels that exist at the same moment will not have the same channel names (issue #7245, different fix)
  • channels/chan_sip.c: Issue 6997 maybe, but anyway - don't retransmit responses to NON-invite requests.
  • channels/chan_sip.c: Bug 7425 - Size of buffer is passed in by len
  • apps/app_queue.c: We should lock the queue before we go making changes to member interface statuses.
  • configs/indications.conf.sample: Add Venezuelan indications (issue #7402 reported by palillo)
  • stdtime/private.h: Bug 7398 - Solaris puts its zoneinfo files in a nonstandard place
  • channels/chan_sip.c: Issue #6820 - Possible fix (already implemented in trunk)
  • apps/app_voicemail.c: Call reset_user_pw upon changing the password using externpass (issue #7395 reported by Ryan Cumming)
  • apps/app_voicemail.c: Issue 7357 - txt file left behind when going to operator. Also, fix a possible file descriptor leak.
  • pbx.c: don't set state to BUSY if the channel is already in the UP state (issue #7376, backported from trunk)
  • configs/iax.conf.sample, channels/chan_iax2.c: don't store multiple secrets delimited with semicolons for peers because this is only valid for users. Instead, only keep the last specified secret for a peer entry. Also, document how multiple secrets are handled in the sample config. (Reported by PCadach on #asterisk-bugs)
  • channels/chan_iax2.c: Zero out a declared structure so as to not crash if it contains invalid data (reported by Qwell on #asterisk-dev)
  • channels/chan_sip.c: Issue 7294 - patch by phsultan - Asterisk sends Invite instead of BYE in some cases.
  • apps/app_queue.c: don't use prefixed structure names for internal structures don't use a plural structure name for a singular object
  • apps/app_voicemail.c: VoicemailMain exits on any key, when the language is set to Italian, instead of properly handling the key (issue 7353).
  • apps/app_queue.c: coding style cleanups on queue interface handling code that was committed for the last release
  • channels/chan_iax2.c: use existing dial string parser for strings supplied to iax2_devicestate, because they can be complete dial strings, not just device names
  • include/asterisk/plc.h, jitterbuf.c, plc.c, apps/app_dumpchan.c, apps/app_chanspy.c: clarify file headers that mention disclaimer usage
  • file.c: don't output 'no format found' when we _did_ find the format but couldn't open the desired file for some other reason
  • apps/app_mixmonitor.c: memory allocation optimizations
  • pbx.c: remove duplicate mutex_unlock
  • apps/app_voicemail.c: fix various places where the code returns without unlocking vmlock or destroying loaded configuration
  • apps/app_festival.c: add a missing close of an open fd, destroy of open config, and removal of the calling channel from the localusers list
  • asterisk.c: revert a change that caused more problems than it fixed and fix the real problem in this code. fds was declared as an array of zero size which caused some weird problems, some of which would only be seen when compiling without optimizations. (fixes issues #7071, #7326, and #7305)
  • include/asterisk/chanspy.h, apps/app_mixmonitor.c, channel.c: Greatly simply the mixmonitor thread, and move channel reference directly to spy structure so that the core can modify it.
  • res/res_agi.c: fix a place where a frame would be free'd twice
  • channels/chan_local.c: only allow chan_local to masquerade the outbound channel onto its owner, instead of the other way around (this will ensure that group variables on the outbound channel are preserved)
  • res/res_agi.c: Move set priority up, because at this point in the code, stdout is no longer the console. If we're unable to set priority, the error goes to Asterisk as if it were an AGI command (issue 7335).
  • pbx.c: fix another place where a frame does not get free'd
  • apps/app_meetme.c: fix up five little places where frames would not be free'd and remove an unnecessary mutex_unlock where there is no way for it to be locked at that time
  • apps/app_ices.c: fix a place that would leak a frame (all of these fixes are in applications that call ast_read() on a channel but have code paths in them that would not free the frame)
  • apps/app_festival.c: fix a couple places that would leak a frame
  • apps/app_alarmreceiver.c: fix two places that would cause a frame to be leaked
  • apps/app_url.c: fix a case where an HTML frame would be leaked
  • apps/app_test.c: Free frames read from the channel when measuring noise. This resulted in about 9 or 10 seconds of leaked frames in both the TestClient and TestServer applications
  • apps/app_zapbarge.c, apps/app_zapscan.c: backport a couple of frame leak fixes from the trunk (revisions 33446, 33447)
  • apps/app_meetme.c: Allow the format outputted by meetme list to be used for meetme commands (like kick) (issue #7322 reported by darkskiez)
  • channels/chan_iax2.c: Remove an unneeded double lock (issue #7310 reported by arkadia)
  • apps/app_dial.c: Handle hangup during recording of screened name (issue #7304 reported by kulldominique)
  • apps/app_meetme.c: Add missing newlines (issue #7323 reported by darkskiez)
  • channels/chan_sip.c: Do not require a context on a domain= setting
  • frame.c: handle out-of-memory conditions properly in ast_frisolate() (reported by Slav Kenov on asterisk-dev mailing list)
  • channels/chan_iax2.c: fix some broken code with BRIDGE_OPTIMIZATION defined (issue #7292)
  • apps/app_voicemail.c: Bug 7287 - A too short voicemail with ODBC_STORAGE will cause the first voicemail to be deleted erroneously
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Reacties (3)

[zeur]
Lekker handig om de hele changelog hier te posten :P, normaal is dat toch een linkje als het zo lang is?
Waar is Obelix ???
Obelisk weet ik niet, maar Idefisk is er wel :D

http://www.voip-info.org/wiki/index.php?page=Idefisk+how+to

Verder: Erg interessant project... wil er voor zowel werk als privé mee gaan experimenteren. Nu nog besluiten of ik het bv. bij Debian packages houd en een management tool zoeken, of wellicht een Trixbox installatie uitvoeren.

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