[svn-commits] kpfleming: tag 1.2.13-netsec r45422 -
/tags/1.2.13-netsec/
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Tue Oct 17 16:01:56 MST 2006
Author: kpfleming
Date: Tue Oct 17 18:01:54 2006
New Revision: 45422
URL: http://svn.digium.com/view/asterisk?rev=45422&view=rev
Log:
importing files for 1.2.13-netsec release
Added:
tags/1.2.13-netsec/.lastclean (with props)
tags/1.2.13-netsec/.version (with props)
tags/1.2.13-netsec/ChangeLog (with props)
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--- tags/1.2.13-netsec/ChangeLog (added)
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+2006-10-17 Kevin P. Fleming <kpfleming at digium.com>
+
+ * Asterisk 1.2.13 released
+
+2006-10-17 20:37 +0000 [r45380] Joshua Colp <jcolp at digium.com>
+
+ * 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 <jparker at digium.com>
+
+ * channels/chan_skinny.c: Fix an integer signedness problem.
+
+2006-10-17 17:22 +0000 [r45326] Kevin P. Fleming <kpfleming at digium.com>
+
+ * LICENSE: provide licensing language for IAXy firmware file
+
+2006-10-17 15:50 +0000 [r45306] Olle Johansson <oej at edvina.net>
+
+ * 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 <jcolp at digium.com>
+
+ * 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 <murf at digium.com>
+
+ * 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 <kpfleming at digium.com>
+
+ * 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 <christian.richter at beronet.com>
+
+ * channels/chan_misdn.c: avoiding warning, fixing potential bug
+ (backported from 1.2)
+
+2006-10-13 17:01 +0000 [r45060] Joshua Colp <jcolp at digium.com>
+
+ * 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 <kpfleming at digium.com>
+
+ * 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 <jcolp at digium.com>
+
+ * 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 <christian.richter at beronet.com>
+
+ * 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 <kpfleming at digium.com>
+
+ * 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 <christian.richter at beronet.com>
+
+ * 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 <oej at edvina.net>
+
+ * channels/chan_sip.c: Issue #8101 - wrong parameter for screening
+ in remote-party-id
+
+2006-10-06 16:52 +0000 [r44501-44580] Joshua Colp <jcolp at digium.com>
+
+ * 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 <christian.richter at beronet.com>
+
+ * channels/chan_misdn.c: fixed segfault which happens during
+ hold/transfer action
+
+2006-10-05 01:27 +0000 [r44392-44432] Kevin P. Fleming <kpfleming at digium.com>
+
+ * 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 <murf at digium.com>
+
+ * 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 <christian.richter at beronet.com>
+
+ * 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 <kpfleming at digium.com>
+
+ * 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 <jcolp at digium.com>
+
+ * 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 <christian.richter at beronet.com>
+
+ * 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 <russell at digium.com>
+
+ * 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 <kpfleming at digium.com>
+
+ * cli.c: proper fix for ast_group_t change
+
+2006-09-28 18:00 +0000 [r43924] Joshua Colp <jcolp at digium.com>
+
+ * 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 <kpfleming at digium.com>
+
+ * 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 <bweschke at btwtech.com>
+
+ * 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 <kpfleming at digium.com>
+
+ * cli.c: eliminate compiler warning introduced by recent changes
+
+2006-09-28 16:13 +0000 [r43891] Joshua Colp <jcolp at digium.com>
+
+ * 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 <bweschke at btwtech.com>
+
+ * 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 <tilghman at mail.jeffandtilghman.com>
+
+ * 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 <jparker at digium.com>
+
+ * 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 <russell at digium.com>
+
+ * 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 <christian.richter at beronet.com>
+
+ * 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 <russell at digium.com>
+
+ * 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 <jcolp at digium.com>
+
+ * .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 <russell at digium.com>
+
+ * 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 <tilghman at mail.jeffandtilghman.com>
+
+ * 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 <russell at digium.com>
+
+ * 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 <jcolp at digium.com>
+
+ * 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 <tilghman at mail.jeffandtilghman.com>
+
+ * 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 <kpfleming at digium.com>
+
+ * 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 <mogorman at digium.com>
+
+ * 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 <tilghman at mail.jeffandtilghman.com>
+
+ * 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 <oej at edvina.net>
+
+ * 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 <jcolp at digium.com>
+
+ * 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 <murf at digium.com>
+
+ * configs/zapata.conf.sample: Clarified what "callwaiting" does in
+ zapata.conf.
+
+2006-09-18 15:05 +0000 [r43159] Joshua Colp <jcolp at digium.com>
+
+ * 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 <tilghman at mail.jeffandtilghman.com>
+
+ * 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 <tilghman at mail.jeffandtilghman.com>
+
+ * 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 <tilghman at mail.jeffandtilghman.com>
+
+ * channels/chan_zap.c: Error message references wrong argument
+ (Issue 7951)
+
+2006-09-13 19:51 +0000 [r42892] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Backport bugfix patch from 7918 to 1.2 -
+ msg_cfg destroyed before used
+
+2006-09-11 Kevin P. Fleming <kpfleming at digium.com>
+
+ * Asterisk 1.2.12.1 released
+
+2006-09-11 21:47 +0000 [r42697-42783] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * apps/app_meetme.c, apps/app_page.c: When paging, only wait 5
+ seconds for the marked user to enter the conference. After that,
+ assume the paging already completed by the time the channel
+ entered the conference and drop back out. (Issue 7275)
+
+ * configs/extensions.conf.sample, configs/alsa.conf.sample,
+ configs/zapata.conf.sample, configs/iax.conf.sample,
+ configs/osp.conf.sample, configs/dundi.conf.sample,
+ configs/enum.conf.sample, configs/vpb.conf.sample,
+ configs/cdr.conf.sample, configs/voicemail.conf.sample,
+ configs/phone.conf.sample, configs/misdn.conf.sample,
+ configs/sip.conf.sample, configs/skinny.conf.sample,
+ configs/features.conf.sample: Spelling/grammar fixes (Issue 7929)
+
+ * configs/voicemail.conf.sample: Two grammar issues (bug 7927)
+
+2006-09-09 20:24 +0000 [r42600] Joshua Colp <jcolp at digium.com>
+
+ * channel.c: Only truly consider the channel in the same format if
+ the format matches the raw format OR if a translation path
+ already exists to translate between them. (issue #7887 reported
+ by softins & issue #7803 reported by alvaro_palma_aste). Thanks
+ goes to stubert for giving me access to a box and showing me a
+ scenario where this occured.
+
+2006-09-09 12:14 +0000 [r42535] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: - Reset proper flag - Don't delete SIP
+ dialog prematurely Strangely enough imported from svn trunk...
+ It's confusing here in Greenland. (Committing from 36.000 feet
+ above Greenland, on the way to asterisk at von
+ http://www.pulver.com/asterisk )
+
+2006-09-08 Kevin P. Fleming <kpfleming at digium.com>
+
+ * Asterisk 1.2.12 released
+
+2006-09-08 18:50 +0000 [r42452] Joshua Colp <jcolp at digium.com>
+
+ * channel.c: Swap spies during masquerading
+
+2006-09-08 16:06 +0000 [r42421] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * apps/app_authenticate.c: Jump logic was backwards: goto returns 0
+ if it succeeds, and we should jump if authentication fails. (Bug
+ #7907)
+
+2006-09-08 04:37 +0000 [r42402] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_local.c: Use ast_best_codec to set the read/write
+ format
+
+2006-09-07 23:12 +0000 [r42355] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * apps/app_record.c: Format vulnerability fix - allowing the user
+ to specify a format is not a good idea (Bug 7811)
+
+2006-09-07 16:30 +0000 [r42260] Joshua Colp <jcolp at digium.com>
+
+ * cdr.c: Let's use the same thing we use in other places to
+ calculate our time for ast_cond_timedwait (issue #7697 reported
+ by bn999)
+
+2006-09-07 02:14 +0000 [r42150-42200] Steve Murphy <murf at digium.com>
+
+ * logger.c: This should fix the problem reported in 7564: logger
+ config file errors getting lost because logging isn't configured
+ yet. The problem was that the code that exists to handle this
+ case was not getting reached, because other tests were causing an
+ early return from ast_log().
+
+ * Makefile: added hours,minutes,seconds .gsm files to the install
+ portion of the makefile, as per bug 7545
+
+2006-09-06 20:02 +0000 [r42148] Joshua Colp <jcolp at digium.com>
+
+ * res/res_agi.c: Don't close the second file descriptor if it's the
+ same as the first one, as it will have already been closed
+ elsewhere and could cause massive panic. (issue #7699 reported by
+ bn999)
+
+2006-09-06 18:16 +0000 [r42133] BJ Weschke <bweschke at btwtech.com>
+
+ * channels/chan_agent.c: Look ma! No more deadlocks! <sic> As
+ posted from #7458 and others similar to it in Mantis: p->app_lock
+ was a mutex really designed for use with agents not in callback
+ mode. That being the case, I've tried to code it so that when
+ callback mode is used, the app_lock mutex will not be
+ locked/unlocked at all. Please let me know how you make out - and
+ if you continue to deadlock now, please reproduce the deadlock
+ logging information and post to Mantis.
+
+2006-09-06 17:10 +0000 [r42110] Christian Richter <christian.richter at beronet.com>
+
+ * channels/chan_misdn.c: fixed pipe consuming bug when using
+ chanIsAvail (#7878), also moved a debug log to the very begining
+ of misdn_hangup.
+
+2006-09-06 15:55 +0000 [r42054-42086] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_iax2.c: Make realtime regseconds work as people
+ expected (0 on registration expiration or release, and actual on
+ normal state) (issue #7684 reported by kshumard)
+
+ * include/asterisk/chanspy.h, apps/app_chanspy.c,
+ apps/app_mixmonitor.c, channel.c: Merge in last round of spy
+ fixes. This should hopefully eliminate all the issues people have
+ been seeing by distinctly separating what each component
+ (core/spy) is responsible for. Core is responsible for adding a
+ spy to a channel, feeding frames to the spy, removing the spy
+ from a channel, and telling the spy to stop. Spy is responsible
+ for reading frames in, and cleaning up after itself.
+
+2006-09-05 16:27 +0000 [r42014] Jason Parker <jparker at digium.com>
+
+ * configs/zapata.conf.sample: Small typo in zapata.conf.sample
+ Reported by ppyy in 7881
+
+2006-09-04 15:46 +0000 [r41989] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Don't kill the pvt before we have sent ACK
+ on CANCEL
+
+2006-09-03 17:38 +0000 [r41827-41882] BJ Weschke <bweschke at btwtech.com>
+
+ * apps/app_queue.c: Make sure the forwarded channel inherits
+ variables appropriately when we receive a call forward in the
+ queue. (#7867 - raarts reported and patched)
+
+ * apps/app_queue.c: Don't keep trying the same member in certain
+ strategies when members of the queue are unavailable (#7278 -
+ diLLec reported and patched)
+
+ * apps/app_chanspy.c: Let's NOT spy on Zap/psuedo channels,
+ mmmmmmmmk?
+
+ * apps/app_queue.c: Setting a retry of 0 is generally not a good
+ idea and shouldn't be allowed. (#7574 - reported by regin)
+
+2006-09-01 22:49 +0000 [r41768] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Only wipe the redirected audio & video
+ IP/port if it's specified, and trigger a reinvite.
+
+2006-09-01 17:35 +0000 [r41716] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/chan_sip.c, include/asterisk/rtp.h, rtp.c: put in proper
+ fix for issue #7294 instead of the broken partial fix that was
+ committed, and thereby also fix issue #7438
+
+2006-09-01 16:33 +0000 [r41690-41691] Joshua Colp <jcolp at digium.com>
+
+ * channel.c: Finish up the last commit (was worse then originally
+ reported)
+
+ * channel.c: Don't treat an unexpected control subclass as voice
+ (issue #7858 reported by PCadach)
+
+2006-08-30 19:01 +0000 [r41423] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Issue #7572 - Hangup when receiving a buggy
+ 487 response to an INVITE
+
+2006-08-30 18:59 +0000 [r41411] Russell Bryant <russell at digium.com>
+
+ * 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_features.c, channels/chan_h323.c,
+ channels/chan_iax2.c: Restore original functionality of 1.2 in
+ places where ANI was not set, but was changed to be set. The
+ original change was done to ensure that the behavior of the
+ "callerid" option in each channel driver was consistent, but it
+ caused an unexpected behavior change of CDR records for users, so
+ this change is being reverted in 1.2. (issue #7695)
+
+2006-08-30 17:58 +0000 [r41390] Joshua Colp <jcolp at digium.com>
+
+ * include/asterisk/lock.h: Properly handle an ETIMEDOUT result from
+ pthread_cond_timedwait (issue #7318 reported by arkadia)
+
+2006-08-30 14:31 +0000 [r41334] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Issue 7822 - don't use SRV lookups if it's
+ disabled.
+
+2006-08-29 13:33 +0000 [r41269] Russell Bryant <russell at digium.com>
+
+ * pbx/pbx_config.c: clean up last commit ... most notably, there is
+ no reason to do heap allocations here, and it also included a
+ potential memory leak
+
+2006-08-29 05:49 +0000 [r41239-41262] Steve Murphy <murf at digium.com>
+
+ * pbx/pbx_config.c: Fixes for bug 7813, via patch submitted by
+ stevens.
+
+ * doc/README.variables: Removed from the docs the mention of the !
+ and =~ operators, as these were knocked out of ast_expr2 because
+ they were new features. Let's hope I can keep them from getting
+ knocked out of the trunk, too!
+
+ * apps/app_macro.c: According to a note added to 7731 by
+ mneuhauser, this will repair a break caused by the last fix
+ (7731).
+
+2006-08-25 15:21 +0000 [r41066-41069] Matt Frederickson <creslin at digium.com>
+
+ * channels/chan_zap.c: Don't send proceeding twice (#7800)
+
+2006-08-25 15:07 +0000 [r41065] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Text only - clarify the reason for entry
+ into authentication mode when the skipuser option is ignored
+
+2006-08-24 19:41 +0000 [r40994] Russell Bryant <russell at digium.com>
+
+ * include/asterisk/linkedlists.h, channel.c, pbx.c: Fix a few
+ issues related to the handling of channel variables - in
+ pbx_builtin_serialize_variables(), the variable list traversal
+ would stop on a variables with empty name/values, which is not
+ appropriate - When removing the GROUP variables, use
+ AST_LIST_REMOVE_CURRENT instead of AST_LIST_REMOVE - During
+ masquerading, when copying the variables list from one channel to
+ the other, using AST_LIST_INSERT_TAIL is not valid for appending
+ a whole list. It leaves the tail pointer of the list invalid.
+ Introduce a new macro, AST_LIST_APPEND_LIST that appends a list
+ properly. (issue #7802, softins)
+
+2006-08-24 17:13 +0000 [r40971-40979] Joshua Colp <jcolp at digium.com>
+
+ * configs/zapata.conf.sample: Minor documentation fix to add the
+ 'dynamic' dialplan option from angler
+
+2006-08-23 16:05 +0000 [r40901] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * res/res_agi.c: Revert last change - breaks retrieval of builtin
+ variables
+
+2006-08-22 Kevin P. Fleming <kpfleming at digium.com>
+
+ * Asterisk 1.2.11 released
+
+2006-08-22 02:59 +0000 [r40821] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * apps/app_random.c: Bug 7779 - Using initstate(3) means that we
+ cannot unload this module once loaded.
+
+2006-08-21 22:34 +0000 [r40798] Matt O'Gorman <mogorman at digium.com>
+
+ * 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.
+
+2006-08-20 22:09 +0000 [r40692] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * CREDITS: Reformat to match the contribution style of other
+ contributors
+
+2006-08-20 04:49 +0000 [r40601] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Turn media level c= parsing on by default
+ (issue #7725 reported by psm)
+
+2006-08-19 01:03 +0000 [r40446] Jason Parker <jparker at digium.com>
+
+ * 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)
+
+2006-08-18 19:10 +0000 [r40310-40392] Kevin P. Fleming <kpfleming at digium.com>
+
+ * 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
+
+2006-08-17 21:57 +0000 [r40306] Christian Richter <christian.richter at beronet.com>
+
+ * 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.
+
+2006-08-17 16:36 +0000 [r40227] Russell Bryant <russell at digium.com>
+
+ * 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)
+
+2006-08-17 16:22 +0000 [r40223-40225] Joshua Colp <jcolp at digium.com>
+
+ * 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
+
+2006-08-17 16:07 +0000 [r40220] Steve Murphy <murf at digium.com>
+
+ * 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.
+
+2006-08-16 18:57 +0000 [r40057] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/chan_mgcp.c: don't allow AUEP responses to overflow the
+ stack during a string copy (reported by Mu Security)
+
+2006-08-15 22:49 +0000 [r39935] Russell Bryant <russell at digium.com>
+
+ * res/res_agi.c: use pbx_builtin_getvar_helper() so that GET
+ VARIABLE can retrieve global variables (issue #7609)
+
+2006-08-15 22:13 +0000 [r39931] Steve Murphy <murf at digium.com>
+
+ * 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.
+
+2006-08-08 18:39 +0000 [r39379] Kevin P. Fleming <kpfleming at digium.com>
+
+ * CREDITS: add explicit listing of anthm's contributions (issue
+ #7683)
+
+2006-08-08 17:04 +0000 [r39350] Russell Bryant <russell at digium.com>
+
+ * channels/chan_sip.c: Increase the buffer size for the callid
+ (issue #7675, reported by pssatcs)
+
+2006-08-07 01:28 +0000 [r39081] Russell Bryant <russell at digium.com>
+
+ * 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.
+
+2006-08-07 00:15 +0000 [r39056] Joshua Colp <jcolp at digium.com>
+
+ * 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)
+
+2006-08-05 09:01 +0000 [r38903-38982] Russell Bryant <russell at digium.com>
+
+ * 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
+
+2006-08-03 19:54 +0000 [r38825] Joshua Colp <jcolp at digium.com>
+
+ * res/res_musiconhold.c: Treat the file as invalid if we have no
+ valid formats for it (issue #7643 reported by KNK)
+
+2006-08-03 05:22 +0000 [r38761] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: Bug 7648 - Checking wrong count for
+ plurality on new messages for Dutch language
+
+2006-08-02 19:29 +0000 [r38686-38731] Kevin P. Fleming <kpfleming at digium.com>
+
+ * 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)
+
+2006-08-01 19:20 +0000 [r38654] Joshua Colp <jcolp at digium.com>
+
+ * 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)
+
+2006-07-31 21:14 +0000 [r38611] Kevin P. Fleming <kpfleming at digium.com>
+
+ * 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
+
+2006-07-31 17:09 +0000 [r38585] Joshua Colp <jcolp at digium.com>
+
+ * 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)
+
+2006-07-31 04:06 +0000 [r38546-38547] Russell Bryant <russell at digium.com>
+
+ * 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
+
+2006-07-29 23:18 +0000 [r38501] Joshua Colp <jcolp at digium.com>
+
+ * 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)
+
+2006-07-28 18:49 +0000 [r38420] Joshua Colp <jcolp at digium.com>
+
+ * 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)
+
+2006-07-27 22:23 +0000 [r38347-38370] Kevin P. Fleming <kpfleming at digium.com>
+
+ * 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
+
+2006-07-27 04:25 +0000 [r38328] Russell Bryant <russell at digium.com>
+
+ * 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.
+
[... 2875 lines stripped ...]
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