aboutsummaryrefslogtreecommitdiffstats
diff options
context:
space:
mode:
authorlmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b>2009-01-21 13:13:32 +0000
committerlmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b>2009-01-21 13:13:32 +0000
commitd84f827ff6a3ac7853c45205604010ef13715276 (patch)
treecfb9c7cb392694a4f7f4e8dee9d4dc0de02ebfa5
parent2a4ccfbbfd3b65e2733b9ceb96fc6d5c925e1a5f (diff)
parentce93b3d1647a7eb6bb899bea22c78e7abd887fb8 (diff)
Creating tag for the release of asterisk-1.4.23
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.23@169666 f38db490-d61c-443f-a65b-d21fe96a405b
-rw-r--r--.lastclean1
-rw-r--r--.version1
-rw-r--r--ChangeLog22076
-rw-r--r--apps/app_userevent.c11
-rw-r--r--apps/app_voicemail.c246
-rw-r--r--channels/chan_local.c8
-rw-r--r--channels/chan_sip.c6
-rw-r--r--include/asterisk/say.h4
-rw-r--r--main/manager.c8
-rw-r--r--main/say.c109
-rw-r--r--res/res_features.c26
11 files changed, 248 insertions, 22248 deletions
diff --git a/.lastclean b/.lastclean
deleted file mode 100644
index bb95160cb..000000000
--- a/.lastclean
+++ /dev/null
@@ -1 +0,0 @@
-33
diff --git a/.version b/.version
deleted file mode 100644
index 580ef4b6c..000000000
--- a/.version
+++ /dev/null
@@ -1 +0,0 @@
-1.4.23
diff --git a/ChangeLog b/ChangeLog
deleted file mode 100644
index c4f174bdc..000000000
--- a/ChangeLog
+++ /dev/null
@@ -1,22076 +0,0 @@
-2009-01-20 Leif Madsen <lmadsen@digium.com>
-
- * Asterisk 1.4.23 released.
-
-2009-01-16 00:19 +0000 [r168745] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_ael.c: This patch fixes a problem where a goto (or jump,
- in this case) fails a consistency check because it can't find a
- matching extension. The problem was a missing instruction to end
- the range notation in the code where it converts the pattern into
- a regex and uses the regex code to determine the match. I tested
- using the AEL code the user supplied, and now, the consistency
- check passes. (closes issue #14141) Reported by: dimas
-
-2009-01-15 18:43 +0000 [r168721] Olle Johansson <oej@edvina.net>
-
- * configs/extconfig.conf.sample: Meetme actually has realtime but
- wasn't documented
-
-2009-01-15 18:22 +0000 [r168716] Terry Wilson <twilson@digium.com>
-
- * res/res_features.c: Convert call to park_call_full to
- masq_park_call_announce Since we removed the AST_PBX_KEEPALIVE
- return value, we need to use masqueraded parking, otherwise we
- will try to call ast_hangup() in __pbx_run() and in
- do_parking_thread() and then promptly crash. (closes issue
- #14215) Reported by: waverly360 Tested by: otherwiseguy (closes
- issue #14228) Reported by: kobaz Tested by: otherwiseguy
-
-2009-01-15 01:20 +0000 [r168633] Tilghman Lesher <tlesher@digium.com>
-
- * /: Blocked revision 168632 from /branches/1.2: 1.2 regression on
- security fix AST-2009-001 (Closes issue #14238)
-
-2009-01-15 00:11 +0000 [r168628] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Fix some crashes from bad datastore handling in
- app_queue.c * The queue_transfer_fixup function was searching for
- and removing the datastore from the incorrect channel, so this
- was fixed. * Most datastore operations regarding the
- queue_transfer datastore were being done without the channel
- locked, so proper channel locking was added, too. (closes issue
- #14086) Reported by: ZX81 Patches: 14086v2.patch uploaded by
- putnopvut (license 60) Tested by: ZX81, festr
-
-2009-01-14 21:48 +0000 [r168622] Richard Mudgett <rmudgett@digium.com>
-
- * channels/misdn/isdn_lib.c: * Fixed create_process() allocation of
- process ID values. The allocated process IDs could overflow their
- respective NT and TE fields. Affects outgoing calls.
-
-2009-01-14 20:52 +0000 [r168614] Sean Bright <sean.bright@gmail.com>
-
- * contrib/scripts/autosupport: Update autosupport script to supply
- info for both Zaptel and DAHDI in 1.4 and be sure to run
- dahdi_test in 1.6.x and trunk instead of zttest. (closes issue
- #14132) Reported by: dsedivec Patches:
- asterisk-1.4-autosupport.patch uploaded by dsedivec (license 638)
- asterisk-trunk-autosupport.patch uploaded by dsedivec (license
- 638)
-
-2009-01-14 19:34 +0000 [r168608] Steve Murphy <murf@digium.com>
-
- * apps/app_page.c: app_page was failing to compile in dev-mode on
- my gcc-4.2.4 system. This change gets rid of the warning.
-
-2009-01-14 19:02 +0000 [r168603] Tilghman Lesher <tlesher@digium.com>
-
- * main/udptl.c: Don't read into a buffer without first checking if
- a value is beyond the end. (closes issue #13600) Reported by:
- atis Patches: 20090106__bug13600.diff.txt uploaded by Corydon76
- (license 14) Tested by: atis
-
-2009-01-14 16:19 +0000 [r168598] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_agent.c: Fix a logic error I found while searching
- through chan_agent.c I found that the allow_multiple_logins
- function would never return 0 due to an incorrect comparison
- being used when traversing the list of agents. While I was
- modifying this function, I also did a little bit of coding
- guidelines cleanup, too.
-
-2009-01-14 01:27 +0000 [r168593] Terry Wilson <twilson@digium.com>
-
- * apps/app_page.c: Don't overflow when paging more than 128
- extensions The number of available slots for calls in app_page
- was hardcoded to 128. Proper bounds checking was not in place to
- enforce this limit, so if more than 128 extensions were passed to
- the Page() app, Asterisk would crash. This patch instead
- dynamically allocates memory for the ast_dial structures and
- removes the (non-functional) arbitrary limit. This issue would
- have special importance to anyone who is dynamically creating the
- argument passed to the Page application and allowing more than
- 128 extensions to be added by an outside user via some external
- interface. The patch posted by a_villacis was slightly modified
- for some coding guidelines and other cleanups. Thanks,
- a_villacis! (closes issue #14217) Reported by: a_villacis
- Patches: 20080912-asterisk-app_page-fix-buffer-overflow.patch
- uploaded by a (license 660) Tested by: otherwiseguy
-
-2009-01-13 19:13 +0000 [r168561] Russell Bryant <russell@digium.com>
-
- * main/indications.c, main/channel.c, apps/app_read.c,
- channels/chan_misdn.c, funcs/func_channel.c,
- include/asterisk/indications.h, apps/app_disa.c, main/app.c,
- res/snmp/agent.c, include/asterisk/channel.h,
- res/res_indications.c: Revert unnecessary indications API change
- from rev 122314
-
-2009-01-13 18:34 +0000 [r168551] Terry Wilson <twilson@digium.com>
-
- * channels/chan_sip.c: Don't pass a value with a side effect to a
- macro (closes issue #14176) Reported by: paraeco Patches:
- chan_sip.c.diff uploaded by paraeco (license 658)
-
-2009-01-13 17:48 +0000 [r168546] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_logic.c: If either conditional is NULL, don't try
- copying it. (closes issue #14226) Reported by: caspy Patches:
- 20090113__bug14226.diff.txt uploaded by Corydon76 (license 14)
-
-2009-01-12 21:42 +0000 [r168507-168516] Jeff Peeler <jpeeler@digium.com>
-
- * res/res_agi.c: (closes issue #13881) Reported by: hoowa Update
- the app CDR field for AGI commands that are not executing an
- application via "exec".
-
- * channels/chan_agent.c: (closes issue #12269) Reported by: IgorG
- Tested by: denisgalvao This gits rid of the notion of an
- owning_app allowing the request and hangup to be initiated by
- different threads. Originating from an active agent channel
- requires this. The implementation primarily changes __login_exec
- to wait on a condition variable rather than a lock. Review:
- http://reviewboard.digium.com/r/35/
-
-2009-01-12 14:58 +0000 [r168482] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: I am reverting the fix made in revision
- 168128 (and its upward merges) after being contacted by Olle
- Johansson and being shown how this fix is incorrect. Thanks to
- Olle for clearing this up for me.
-
-2009-01-12 14:57 +0000 [r168480] Russell Bryant <russell@digium.com>
-
- * configs/indications.conf.sample: s/ringdance/ringcadence/ for
- Bulgaria
-
-2009-01-10 20:47 +0000 [r168267-168382] Kevin P. Fleming <kpfleming@digium.com>
-
- * README: small commit to test new server
-
- * README: small commit to test new server
-
- * sounds/Makefile: update to use new sound file packages that
- include license files
-
-2009-01-09 22:14 +0000 [r168198] Russell Bryant <russell@digium.com>
-
- * res/res_musiconhold.c: Make this compile for mvanbaak
-
-2009-01-09 21:28 +0000 [r168191] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_misdn.c: * Fix for JIRA AST-175/ABE-1757 *
- Miscellaneous doxygen comments added.
-
-2009-01-09 20:08 +0000 [r168128] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: Add check_via calls to more request handlers
- INFO, NOTIFY, OPTIONS, REFER, and MESSAGE requests were not
- checking the topmost Via to determine where to send the response.
- Adding check_via calls to those request handlers solves this.
- (closes issue #13071) Reported by: baron Patches: check_via.patch
- uploaded by baron (license 531) Tested by: baron
-
-2009-01-08 22:08 +0000 [r167840] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_agi.c: Don't truncate database results at 255 chars.
- (closes issue #14069) Reported by: evandro Patches:
- 20081214__bug14069.diff.txt uploaded by Corydon76 (license 14)
-
-2009-01-08 17:24 +0000 [r167620-167714] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_sip.c: remove an unnecessary argument to
- queue_request()
-
- * channels/chan_sip.c: When a SIP request or response arrives for a
- dialog with an associated Asterisk channel, and the lock on that
- channel cannot be obtained because it is held by another thread,
- instead of dropping the request/response, queue it for later
- processing when the channel lock becomes available.
- http://reviewboard.digium.com/r/117/
-
-2009-01-07 22:35 +0000 [r167432-167566] Russell Bryant <russell@digium.com>
-
- * main/file.c: Fix the last couple of places where free() was
- improperly used directly.
-
- * main/file.c: Don't fclose() the file early, the filestream
- destructor will handle it.
-
- * main/file.c: Only try to close the file if one was actually
- opened
-
- * main/file.c: Don't use free() directly. This caused a crash since
- ast_filestream is now an ao2 object. Reported by JunK-Y on IRC,
- #asterisk-dev
-
- * main/indications.c: Treat an empty string the same way as a NULL
- country argument. In passing, simplify the handling of returning
- a default tone zone.
-
-2009-01-06 21:35 +0000 [r167299] Mark Michelson <mmichelson@digium.com>
-
- * main/db.c: Use the correct variable when creating the format
- string (closes issue #14177) Reported by: nic_bellamy Patches:
- asterisk-trunk-svn-r167242-ast_db_gettree.patch uploaded by nic
- (license 299)
-
-2009-01-06 20:48 +0000 [r167260] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 167259 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r167259 | tilghman | 2009-01-06 14:44:03 -0600 (Tue, 06
- Jan 2009) | 2 lines Security fix AST-2009-001. ........
-
-2009-01-05 16:51 +0000 [r167179] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: A couple of changes to T.38 SDP attribute
- handling There are some boolean attributes for T.38 such as
- T38FaxFillBitRemoval, T38FaxTranscodingMMR, and
- T38FaxTranscodingJBIG. By simply being present, we should treat
- these as a "true" value. The current code, however, was requiring
- a 1 or 0 as the value of the attribute in order to parse it. This
- is due to the fact that there are some T.38 endpoints and
- gateways that also transmit this information incorrectly. This
- patch follows the "be liberal in what you accept and strict in
- what you send" philosophy by accepting both the correctly- and
- incorrectly-formatted attributes, but only sending information as
- it is supposed to be sent. It was also discovered that a
- particular type of T.38 gateway sends some non-standard T.38 SDP
- attributes. Instead of using T38FaxMaxDatagram and T38MaxBitRate,
- it used T38MaxDatagram and T38FaxMaxRate respectively. We now
- will properly accept these attributes as well. Note that there
- are a lot of patches cited in the below commit message template.
- This is because the person who submitted these patches is an
- awesome person and wrote 1.4, 1.6.0, and 1.6.1 variants. (closes
- issue #13976) Reported by: linulin Patches:
- chan_sip.c.1.4-update1.diff uploaded by arcivanov (license 648)
- chan_sip.c.1.6.0-update1.diff uploaded by arcivanov (license 648)
- chan_sip.c.1.6.1-update1.diff uploaded by arcivanov (license 648)
- chan_sip.c.1.4-relaxedT38_update1.diff uploaded by arcivanov
- (license 648) chan_sip.c.1.6.0-relaxedT38_update1.diff uploaded
- by arcivanov (license 648)
- chan_sip.c.1.6.1-relaxedT38_update1.diff uploaded by arcivanov
- (license 648) Tested by: arcivanov
-
-2009-01-01 00:01 +0000 [r166953-167095] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_alsa.c: Repeat attempts to write when we receive
- -EAGAIN from the driver, as detailed in the ALSA sample code (see
- http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html#a32)
- Reported by: Jerry Geis (via the -users list) Fixed by: me
- (license 14)
-
- * channels/chan_local.c: Also inherit the musiconhold class.
- (Closes #14153) Reported by: Jerry Geis, via the users list.
- Patch by: me (license 14)
-
-2008-12-28 15:13 +0000 [r166772] Russell Bryant <russell@digium.com>
-
- * channels/misdn_config.c: Use strncat() instead of an sprintf() in
- which source and target buffers overlap
- http://lists.digium.com/pipermail/asterisk-dev/2008-December/035919.html
-
-2008-12-23 15:35 +0000 [r166592] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c, channels/chan_iax2.c: Compile, even if both
- DAHDI and Zaptel are not installed. (Closes issue #14120)
-
-2008-12-23 15:16 +0000 [r166568] Mark Michelson <mmichelson@digium.com>
-
- * main/channel.c: Fix a crash resulting from a datastore with
- inheritance but no duplicate callback The fix for this is to
- simply set the newly created datastore's data pointer to NULL if
- it is inherited but has no duplicate callback. (closes issue
- #14113) Reported by: francesco_r Patches: 14113.patch uploaded by
- putnopvut (license 60) Tested by: francesco_r
-
-2008-12-23 04:05 +0000 [r166509] Tilghman Lesher <tlesher@digium.com>
-
- * main/channel.c: Use the integer form of condition for integer
- comparisons. (closes issue #14127) Reported by: andrew
-
-2008-12-22 20:56 +0000 [r166380] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_dahdi.c: Fix a deadlock relating to channel locks
- and autoservice It has been discovered that if a channel is
- locked prior to a call to ast_autoservice_stop, then it is likely
- that a deadlock will occur. The reason is that the call to
- ast_autoservice_stop has a check built into it to be sure that
- the thread running autoservice is not currently trying to
- manipulate the channel we are about to pull out of autoservice.
- The autoservice thread, however, cannot advance beyond where it
- currently is, though, because it is trying to acquire the lock of
- the channel for which autoservice is attempting to be stopped.
- The gist of all this is that a channel MUST NOT be locked when
- attempting to stop autoservice on the channel. In this particular
- case, the channel was locked by a call to ast_read. A call to
- ast_exists_extension led to autoservice being started and stopped
- due to the existence of dialplan switches. It may be that there
- are future commits which handle the same symptoms but in a
- different location, but based on my looks through the code, it is
- very rare to see a construct such as this one. (closes issue
- #14057) Reported by: rtrauntvein Patches: 14057v3.patch uploaded
- by putnopvut (license 60) Tested by: rtrauntvein Review:
- http://reviewboard.digium.com/r/107/
-
-2008-12-22 17:22 +0000 [r166262-166297] Russell Bryant <russell@digium.com>
-
- * main/utils.c: Fix up timeout handling in ast_carefulwrite().
-
- * include/asterisk/strings.h, res/res_musiconhold.c: Re-work ref
- count handling of MoH classes using astobj2 to resolve crashes.
- (closes issue #13566) Reported by: igorcarneiro Tested by:
- russell Review: http://reviewboard.digium.com/r/106/
-
-2008-12-19 23:34 +0000 [r166157] Mark Michelson <mmichelson@digium.com>
-
- * main/channel.c, funcs/func_audiohookinherit.c (added),
- channels/chan_sip.c, include/asterisk/audiohook.h,
- main/audiohook.c, CHANGES: Backport of AUDIOHOOK_INHERIT for
- Asterisk 1.4 (closes issue #13538) Reported by: mbit Patches:
- 13538.patch uploaded by putnopvut (license 60) Tested by:
- putnopvut
-
-2008-12-19 22:30 +0000 [r166093] Steve Murphy <murf@digium.com>
-
- * apps/app_dial.c, res/res_features.c, include/asterisk/pbx.h,
- apps/app_queue.c: This merges the masqpark branch into 1.4 These
- changes eliminate the need for (and use of) the KEEPALIVE return
- code in res_features.c; There are other places that use this
- result code for similar purposes at a higher level, these appear
- to be left alone in 1.4, but attacked in trunk. The reason these
- changes are being made in 1.4, is that parking ends a channel's
- life, in some situations, and the code in the bridge (and some
- other places), was not checking the result code properly, and
- dereferencing the channel pointer, which could lead to memory
- corruption and crashes. Calling the masq_park function eliminates
- this danger in higher levels. A series of previous commits have
- replaced some parking calls with masq_park, but this patch puts
- them ALL to rest, (except one, purposely left alone because a
- masquerade is done anyway), and gets rid of the code that tests
- the KEEPALIVE result, and the NOHANGUP_PEER result codes. While
- bug 13820 inspired this work, this patch does not solve all the
- problems mentioned there. I have tested this patch (again) to
- make sure I have not introduced regressions. Crashes that
- occurred when a parked party hung up while the parking party was
- listening to the numbers of the parking stall being assigned, is
- eliminated. These are the cases where parking code may be
- activated: 1. Feature one touch (eg. *3) 2. Feature blind xfer to
- parking lot (eg ##700) 3. Run Park() app from dialplan (eg sip
- xfer to 700) (eg. dahdi hookflash xfer to 700) 4. Run Park via
- manager. The interesting testing cases for parking are: I. A
- calls B, A parks B a. B hangs up while A is getting the numbers
- announced. b. B hangs up after A gets the announcement, but
- before the parking time expires c. B waits, time expires, A is
- redialed, A answers, B and A are connected, after which, B hangs
- up. d. C picks up B while still in parking lot. II. A calls B, B
- parks A a. A hangs up while B is getting the numbers announced.
- b. A hangs up after B gets the announcement, but before the
- parking time expires c. A waits, time expires, B is redialed, B
- answers, A and B are connected, after which, A hangs up. d. C
- picks up A while still in parking lot. Testing this throroughly
- involves acting all the permutations of I and II, in situations
- 1,2,3, and 4. Since I added a few more changes (ALL references to
- KEEPALIVE in the bridge code eliimated (I missed one earlier), I
- retested most of the above cases, and no crashes. H-extension
- weirdness. Current h-extension execution is not completely
- correct for several of the cases. For the case where A calls B,
- and A parks B, the 'h' exten is run on A's channel as soon as the
- park is accomplished. This is expected behavior. But when A calls
- B, and B parks A, this will be current behavior: After B parks A,
- B is hung up by the system, and the 'h' (hangup) exten gets run,
- but the channel mentioned will be a derivative of A's... Thus, if
- A is DAHDI/1, and B is DAHDI/2, the h-extension will be run on
- channel Parked/DAHDI/1-1<ZOMBIE>, and the start/answer/end info
- will be those relating to Channel A. And, in the case where A is
- reconnected to B after the park time expires, when both parties
- hang up after the joyful reunion, no h-exten will be run at all.
- In the case where C picks up A from the parking lot, when either
- A or C hang up, the h-exten will be run for the C channel. CDR's
- are a separate issue, and not addressed here. As to WHY this
- strange behavior occurs, the answer lies in the procedure
- followed to accomplish handing over the channel to the parking
- manager thread. This procedure is called masquerading. In the
- process, a duplicate copy of the channel is created, and most of
- the active data is given to the new copy. The original channel
- gets its name changed to XXX<ZOMBIE> and keeps the PBX
- information for the sake of the original thread (preserving its
- role as a call originator, if it had this role to begin with),
- while the new channel is without this info and becomes a call
- target (a "peer"). In this case, the parking lot manager thread
- is handed the new (masqueraded) channel. It will not run an
- h-exten on the channel if it hangs up while in the parking lot.
- The h exten will be run on the original channel instead, in the
- original thread, after the bridge completes. See bug 13820 for
- our intentions as to how to clean up the h exten behavior.
- Review: http://reviewboard.digium.com/r/29/
-
-2008-12-19 19:48 +0000 [r165991] Jeff Peeler <jpeeler@digium.com>
-
- * include/asterisk/dahdi_compat.h, main/asterisk.c, main/channel.c,
- apps/app_dahdibarge.c, channels/chan_dahdi.c, apps/app_meetme.c,
- apps/app_dahdiscan.c, codecs/codec_dahdi.c,
- res/res_musiconhold.c, channels/chan_iax2.c: (closes issue
- #13480) Reported by: tzafrir Replace a bunch of if defined checks
- for Zaptel/DAHDI through several new defines in dahdi_compat.h.
- This removes a lot of code duplication. Example from bug: #ifdef
- HAVE_ZAPTEL fd = open("/dev/zap/pseudo", O_RDWR); #else fd =
- open("/dev/dahdi/pseudo", O_RDWR); #endif is replaced with: fd =
- open(DAHDI_FILE_PSEUDO, O_RDRW);
-
-2008-12-19 15:03 +0000 [r165796-165889] Russell Bryant <russell@digium.com>
-
- * apps/app_chanspy.c: Ensure that the chanspy datastore is fully
- initialized. This patch resolved some random crash issues
- observed by a user on a BSD system (closes issue #14111) Reported
- by: ys Patches: app_chanspy.c.diff uploaded by ys (license 281)
-
- * main/utils.c: Make ast_carefulwrite() be more careful. This patch
- handles some additional cases that could result in partial writes
- to the file description. This was done to address complaints
- about partial writes on AMI. (issue #13546) (more changes needed
- to address potential problems in 1.6) Reported by: srt Tested by:
- russell Review: http://reviewboard.digium.com/r/99/
-
-2008-12-18 21:14 +0000 [r165767] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: Add mutexes around accesses to the IMAP
- library interface. This prevents certain crashes, especially when
- shared mailboxes are used. (closes issue #13653) Reported by:
- howardwilkinson Patches:
- asterisk-1.4.21.2-appvoicemail-sharedimap-lock.patch uploaded by
- howardwilkinson (license 590) Tested by: jpeeler
-
-2008-12-18 18:52 +0000 [r165661] Russell Bryant <russell@digium.com>
-
- * res/res_musiconhold.c: Set the process group ID on the MOH
- process so that all children will get killed (closes issue
- #14099) Reported by: caspy Patches:
- res_musiconhold.c.patch.killpg.try2 uploaded by caspy (license
- 645)
-
-2008-12-18 17:11 +0000 [r165537-165591] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Only care about a compatible codec for early bridging
- if we are actually bridging to another channel. If we are not we
- actually want to bring the audio back to us. (closes issue
- #13545) Reported by: davidw
-
- * apps/app_followme.c: Do not crash if we are not passed in a
- followme id. (closes issue #14106) Reported by: ys Patches:
- app_followme.c.2.diff uploaded by ys (license 281)
-
-2008-12-17 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.23-rc3 released.
-
-2008-12-17 21:14 +0000 [r165317] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_macro.c: Reverse the fix from issue #6176 and add proper
- handling for that issue. (Closes issue #13962, closes issue
- #13363) Fixed by myself (license 14)
-
-2008-12-17 20:51 +0000 [r164977-165255] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_meetme.c, apps/app_realtime.c, apps/app_directory.c,
- apps/app_queue.c: Fix some memory leaks found while looking at
- how realtime configs are handled. Also cleaned up some coding
- guidelines violations in app_realtime.c, mostly related to
- spacing
-
- * channels/chan_sip.c: After looking through SIP registration code
- most of the day, this is one of the few things I could find that
- was just plain wrong. Even though it probably isn't possible for
- it to happen, it seems weird to have code that checks if a
- pointer is NULL and then immediately dereferences that pointer if
- it was NULL.
-
-2008-12-16 21:38 +0000 [r164672-164881] Russell Bryant <russell@digium.com>
-
- * main/utils.c: Fix an issue where DEBUG_THREADS may erroneously
- report that a thread is exiting while holding a lock. If the last
- lock attempt was a trylock, and it failed, it will still be in
- the list of locks so that it can be reported. (closes issue
- #13219) Reported by: pj
-
- * apps/app_macro.c: Do not dereference the channel if
- AST_PBX_KEEPALIVE has been returned. This is a bug I noticed
- while looking at the code for app_macro. This return code means
- that another thread has assumed ownership of the channel and it
- can no longer be touched. (I hate this return code with a
- passion, by the way.)
-
- * main/manager.c: Add "restart gracefully" to the AMI blacklist of
- CLI commands. "module unload" was already identified as a command
- that can not be used from the AMI. "restart gracefully"
- effectively unloads all modules, and will run in to the same
- problems. (closes issue #13894) Reported by: kernelsensei
-
- * include/asterisk/threadstorage.h, main/threadstorage.c: Fix
- memory leak and invalid reporting issues with DEBUG_THREADLOCALS.
- One issue was that the ast_mutex_* API was being used within the
- context of the thread local data destructors. We would go off and
- allocate more thread local data while the pthread lib was in the
- middle of destroying it all. This led to a memory leak. Another
- issue was an invalid argument being provided to the the
- object_add API call. (closes issue #13678) Reported by: ys Tested
- by: Russell
-
- * channels/chan_sip.c: Fix a memory leak related to the use of the
- "setvar" configuration option. The problem was that these
- variables were being appended to the list of vars on the sip_pvt
- every time a re-registration or re-subscription came in. Since
- it's just a waste of memory to put them there unless the request
- was an INVITE, then the fix is to check the request type before
- copying the vars. (closes issue #14037) Reported by: marvinek
- Tested by: russell
-
-2008-12-16 15:15 +0000 [r164634] Steve Murphy <murf@digium.com>
-
- * main/pbx.c: I added a sentence to clarify why - and ' ' are
- ignored in patterns as per bug 14076. Leif says he'll put some
- stuff about it in the extensions.conf sample, etc.
-
-2008-12-16 14:28 +0000 [r164605] Russell Bryant <russell@digium.com>
-
- * res/res_musiconhold.c: Don't try to change working directory if a
- directory was not configured. (closes issue #14089) Reported by:
- caspy
-
-2008-12-15 19:53 +0000 [r164416-164422] Mark Michelson <mmichelson@digium.com>
-
- * include/asterisk/pbx.h: Add the deadlock note to
- ast_spawn_extension as well
-
- * include/asterisk/channel.h, include/asterisk/pbx.h: Add notes to
- autoservice and pbx doxygen regarding a potential deadlock
- scenario so that it is avoided in the future
-
-2008-12-15 18:11 +0000 [r164204-164350] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Do not try to unlock a non-existant channel
- if the transfer fails. (closes issue #13800) Reported by: dwagner
- Patches: asterisk-1.4.22-chan-sip-nullp.patch uploaded by tweety
- (license 608)
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac,
- include/asterisk/channel.h: Use autoconf logic to determine
- whether the system has timersub or not. Do not blindly assume
- Solaris does not. (closes issue #13838) Reported by: ano
-
- * apps/app_dial.c: Can we try not to assign an unsigned int to -1?
- (closes issue #14074) Reported by: wetwired
-
-2008-12-15 14:31 +0000 [r164201] Russell Bryant <russell@digium.com>
-
- * main/channel.c, res/res_features.c: Handle a case where a call
- can be bridged to a channel that is still ringing. The issue that
- was reported was about a case where a RINGING channel got
- redirected to an extension to pick up a call from parking. Once
- the parked call got taken out of parking, it heard silence until
- the other side answered. Ideally, the caller that was parked
- would get a ringing indication. This patch fixes this case so
- that the caller receives ringback once it comes out of parking
- until the other side answers. The fixes are: - Make sure we
- remember that a channel was an outgoing channel when doing a
- masquerade. This prevents an erroneous ast_answer() call on the
- channel, which causes a bogus 200 OK to be sent in the case of
- SIP. - Add some additional comments to explain related parts of
- code. - Update the handling of the ast_channel visible_indication
- field. Storing values that are not stateful is pointless. Control
- frames that are events or commands should be ignored. - When a
- bridge first starts, check to see if the peer channel needs to be
- given ringing indication because the calling side is still
- ringing. - Rework ast_indicate_data() a bit for the sake of
- readability. (closes issue #13747) Reported by: davidw Tested by:
- russell Review: http://reviewboard.digium.com/r/90/
-
-2008-12-13 23:22 +0000 [r164082] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_dial.c: Change the default calldurationlimit from the
- special value 0 to -1, so we can better detect an exceptional
- case. This follows on to the changes made in revision 156386.
- Related to issue #13851. (closes issue #13974) Reported by:
- paradise Patches: 20081208__bug13974.diff.txt uploaded by
- Corydon76 (license 14) Tested by: file, blitzrage, ZX81
-
-2008-12-12 22:20 +0000 [r163785] Russell Bryant <russell@digium.com>
-
- * /: Set the reviewboard:url property on 1.4, as well
-
-2008-12-12 22:03 +0000 [r163761] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c, main/editline/read.c: Simple fix for Ctrl-C not
- immediately exiting Asterisk, but also add a pointer inside
- editline to look back to asterisk.c, so others don't spend as
- much time as I did looking (in the wrong place) for the
- appropriate function. Reported by: ZX81, via the #asterisk-users
- channel Fixed by: me (license 14)
-
-2008-12-12 14:40 +0000 [r163448-163511] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_dundi.c: Specify uint32_t for variables storing a CRC32
- so that it is actually 32 bits on 64-bit machines, as well.
- (inspired by issue #13879)
-
- * main/channel.c, main/autoservice.c, include/asterisk/channel.h:
- Resolve issues that could cause DTMF to be processed out of
- order. These changes come from team/russell/issue_12658 1) Change
- autoservice to put digits on the head of the channel's frame
- readq instead of the tail. If there were frames on the readq that
- autoservice had not yet read, the previous code would have
- resulted in out of order processing. This required a new API call
- to queue a frame to the head of the queue instead of the tail. 2)
- Change up the processing of DTMF in ast_read(). Some of the
- problems were the result of having two sources of pending DTMF
- frames. There was the dtmfq and the more generic readq. Both were
- used for pending DTMF in various scenarios. Simplifying things to
- only use the frame readq avoids some of the problems. 3) Fix a
- bug where a DTMF END frame could get passed through when it
- shouldn't have. If code set END_DTMF_ONLY in the middle of digit
- emulation, and a digit arrived before emulation was complete,
- digits would get processed out of order. (closes issue #12658)
- Reported by: dimas Tested by: russell, file Review:
- http://reviewboard.digium.com/r/85/
-
-2008-12-11 23:35 +0000 [r163383] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c: When a Ctrl-C or Ctrl-D ends a remote console,
- on certain shells, the terminal is messed up. By intercepting
- those events with a signal handler in the remote console, we can
- avoid those issues. (closes issue #13464) Reported by: tzafrir
- Patches: 20081110__bug13464.diff.txt uploaded by Corydon76
- (license 14) Tested by: blitzrage
-
-2008-12-11 22:44 +0000 [r163316] Matt Nicholson <mnicholson@digium.com>
-
- * pbx/pbx_dundi.c: Clean up the dundi cache every 5 minutes.
- (closes issue #13819) Reported by: adomjan Patches:
- pbx_dundi.c-clearcache.patch uploaded by adomjan (license 487)
- dundi_clearecache3.diff uploaded by mnicholson (license 96)
- Tested by: adomjan
-
-2008-12-11 21:46 +0000 [r163092-163253] Russell Bryant <russell@digium.com>
-
- * funcs/func_strings.c, funcs/func_cut.c: Fix some observed
- slowdowns in dialplan processing. The change is to remove
- autoservice usage from dialplan functions that do not need it
- because they do not perform operations that potentially block.
- (closes issue #13940) Reported by: tbelder
-
- * res/res_features.c: Fix an issue that made it so you could only
- have a single caller executing a custom feature at a time. This
- was especially problematic when custom features ran for any
- appreciable amount of time. The fix turned out to be quite
- simple. The dynamic features are now stored in a read/write list
- instead of a list using a mutex. (closes issue #13478) Reported
- by: neutrino88 Fix suggested by file
-
-2008-12-11 16:51 +0000 [r163088] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_agi.c: Don't wait forever, if there's a specified
- recording timeout. (closes issue #13885) Reported by: bamby
- Patches: res_agi.c.patch uploaded by bamby (license 430)
-
-2008-12-11 16:46 +0000 [r163080-163084] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Revert this cast to long. Using time_t here
- causes build failures on a FreeBSD 32-bit build.
-
- * apps/app_queue.c: Fix a potential crash due to unsafe datastore
- handling. This patch also contains a conversion from using long
- to time_t for representing times for a queue, as well as some
- whitespace fixes. (closes issue #14060) Reported by: nivek
- Patches: datastore_fixup.patch.corrected uploaded by nivek
- (license 636) with slight modification from me Tested by: nivek
-
-2008-12-10 22:52 +0000 [r162874-162926] Jeff Peeler <jpeeler@digium.com>
-
- * res/res_musiconhold.c: Oops, inverted logic for a strcasecmp
- check. Pointed out by mmichelson, thanks!
-
- * res/res_musiconhold.c: (closes issue #13229) Reported by:
- clegall_proformatique Ensure that moh_generate does not return
- prematurely before local_ast_moh_stop is called. Also, the sleep
- in mp3_spawn now only occurs for http locations since it seems to
- have been added originally only for failing media streams.
-
-2008-12-10 19:01 +0000 [r162738-162804] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Fix subscription based MWI up a bit. We only
- want to put sip: at the beginning of the URI if it is not already
- there and revert code to ignore destination check if subscribing
- for MWI. (closes issue #12560) Reported by: vsauer Patches:
- patch001.diff uploaded by ramonpeek (license 266)
-
- * channels/chan_sip.c: When a SIP peer unregisters set the expiry
- time back to 0 so that the 200 OK contains an expires of 0.
- (closes issue #13599) Reported by: hjourdain Patches:
- chan_sip.c.diff uploaded by hjourdain (license 583)
-
-2008-12-10 16:45 +0000 [r162671] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael_lex.c, pbx/ael/ael.flex: (closes issue #14022)
- Reported by: wetwired Tested by: murf I checked, and I added a
- mod to the trunk version of Asterisk that would make it 8-bit
- transparent on 27 Nov 2007, but I made no such updates to 1.4. My
- best guess is that 1.4 was released, and it was not appropriate
- to commit an enhancement. But I'm going to add the same fixes to
- 1.4 now, for the following reasons: 1. wetwired is correct; 1.4
- is **mostly** 8-bit transparent now. This is because the lexical
- token forming rules use . in most 'word' state continuances. It's
- just the beginning of a 'word' that is picky. 2. Accepting 8-bit
- chars in some places and not others leads to bug reports like
- this.
-
-2008-12-10 16:44 +0000 [r162659-162670] Mark Michelson <mmichelson@digium.com>
-
- * include/asterisk/stringfields.h: Update to stringfield handling
- so that side-effects on parameters are not evaluated multiple
- times. An example where this caused a problem was in chan_sip.c,
- with the line ast_string_field_set(p, fromdomain, ++fromdomain);
- This patch was originally uploaded to issue #13783 by jamessan.
- While the issue was closed for other reasons, this patch is valid
- and fixes a separate problem, and is thus being committed.
-
- * channels/chan_sip.c: Revert fix for issue 13570. It has caused
- more problems than it helped to fix. (closes issue #13783)
- Reported by: navkumar (closes issue #14025) Reported by: ffs
-
- * doc/misdn.txt: Add missing documentation to misdn.txt (closes
- issue #14052) Reported by: festr Patches: misdn.txt.patch
- uploaded by festr (license 443)
-
-2008-12-10 16:05 +0000 [r162653] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Increment the sequence number on the end packets for
- RFC2833. After reading the RFC some more and doing some testing I
- agree with this change. (closes issue #12983) Reported by: vt
- Patches: dtmf_inc_seqnum_on_end_pkts.diff uploaded by vt (license
- 520)
-
-2008-12-09 23:08 +0000 [r162463] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: Oops, should be "tz", not "zonetag".
-
-2008-12-09 22:17 +0000 [r162413] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c, include/asterisk/utils.h, main/utils.c: Remove
- the test_for_thread_safety() function completely. The test is not
- valid. Besides, if we actually suspected that recursive mutexes
- were not working, we would get a ton of LOG_ERROR messages when
- DEBUG_THREADS is turned on. (inspired by a discussion on the
- asterisk-dev list)
-
-2008-12-09 21:53 +0000 [r162348] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: We appear to have documented tz= in the
- [general] section of voicemail.conf, without actually having
- implemented it. Oops. (Reported by Olivier on the -users list)
-
-2008-12-09 21:14 +0000 [r162341] Joshua Colp <jcolp@digium.com>
-
- * apps/app_directed_pickup.c: Add 'down' as a valid state for
- directed call pickup. This creeps up when we receive session
- progress when dialing a device and not ringing. (closes issue
- #14005) Reported by: ddl
-
-2008-12-09 20:57 +0000 [r162286] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c: Fix an issue where callers on an incoming call
- on an SLA trunk would not hear ringback. We need to make sure
- that we don't start writing audio to the trunk channel until
- we're actually ready to answer it. Otherwise, the channel driver
- will treat it as inband progress, even though all they are
- getting is silence. (closes issue #12471) Reported by: mthomasslo
-
-2008-12-09 20:44 +0000 [r162273] Joshua Colp <jcolp@digium.com>
-
- * apps/app_festival.c: Fix double declaration of 'x' on the PPC
- platform. (closes issue #14038) Reported by: ffloimair
-
-2008-12-09 20:28 +0000 [r162265] Mark Michelson <mmichelson@digium.com>
-
- * main/pbx.c: If we fail to start a thread for the pbx to run in,
- we need to be sure to decrease the number of active calls on the
- system. This fix may relate to ABE-1713, but it is not certain
- yet.
-
-2008-12-09 20:20 +0000 [r162264] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael_lex.c, pbx/ael/ael.flex: In discussion with
- seanbright on #asterisk-dev, I have added a default rule, and an
- option to suppress the default rule from being generated in the
- flex output, for the sake of those OS's where they didn't tweak
- flex's ECHO macro, and the compiler doesn't like it. The
- regressions are OK with this.
-
-2008-12-09 19:47 +0000 [r162188-162204] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Make sure that the timestamp for DTMF is not the same
- as the previous voice frame and do not send audio when
- transmitting DTMF as this confuses some equipment. (closes issue
- #13209) Reported by: ip-rob Patches: 13209.diff uploaded by file
- (license 11) Tested by: ip-rob, bujones
-
- * main/rtp.c: Take video into account when early bridging RTP.
- (closes issue #13535) Reported by: davidw
-
-2008-12-09 18:13 +0000 [r162136] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael_lex.c, pbx/ael/ael.flex: Previous fix used ast_malloc
- and ast_copy_string and messed up the standalone stuff. Fixed.
-
-2008-12-09 17:07 +0000 [r162071] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_phone.c: For some reason, after a distclean, gcc
- started returning 'value computed is not used'. Fixing (for
- --enable-dev-mode).
-
-2008-12-09 16:46 +0000 [r162014] Russell Bryant <russell@digium.com>
-
- * apps/app_disa.c: Allow DISA to handle extensions that start with
- #. (closes issue #13330) Reported by: jcovert
-
-2008-12-09 16:31 +0000 [r162013] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael_lex.c, pbx/pbx_ael.c, include/asterisk/ael_structs.h,
- pbx/ael/ael.flex: (closes issue #14019) Reported by: ckjohnsonme
- Patches: 14019.diff uploaded by murf (license 17) Tested by:
- ckjohnsonme, murf This crash was the result of a few small errors
- that would combine in 64-bit land to result in a crash. 32-bit
- land might have seen these combine to mysteriously drop the args
- to an application call, in certain circumstances. Also, in trying
- to find this bug, I spotted a situation in the flex input, where,
- in passing back a 'word' to the parser, it would allocate a
- buffer larger than necessary. I changed the usage in such
- situations, so that strdup was not used, but rather, an
- ast_malloc, followed by ast_copy_string. I removed a field from
- the pval struct, in u2, that was never getting used, and set in
- one spot in the code. I believe it was an artifact of a previous
- fix to make switch cases work invisibly with extens. And, for
- goto's I removed a '!' from before a strcmp, that has been there
- since the initial merging of AEL2, that might prevent the proper
- target of a goto from being found. This was pretty harmless on
- its own, as it would just louse up a consistency check for users.
- Many thanks to ckjohnsonme for providing a simplified and
- complete set of information about the bug, that helped
- considerably in finding and fixing the problem. Now, to get
- aelparse up and running again in trunk, and out of its "horribly
- broken" state, so I can run the regression suite!
-
-2008-12-09 14:52 +0000 [r161948] Russell Bryant <russell@digium.com>
-
- * main/app.c: Fix a problem with GROUP() settings on a masquerade.
- The previous code carried over group settings from the old
- channel to the new one. However, it did nothing with the group
- settings that were already on the new channel. This patch removes
- all group settings that already existed on the new channel. I
- have a more complicated version of this patch which addresses
- only the most blatant problem with this, which is that a channel
- can end up with multiple group settings in the same category.
- However, I could not think of a use case for keeping any of the
- group settings from the old channel, so I went this route for
- now. (closes AST-152)
-
-2008-12-08 17:52 +0000 [r161725] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Make the usereqphone option work again.
- (closes issue #13474) Reported by: mmaguire Patches:
- 20080912_bug13474.diff uploaded by mmaguire (license 571)
-
-2008-12-05 21:02 +0000 [r161426] Sean Bright <sean.bright@gmail.com>
-
- * main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions
- 161421 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r161421 | seanbright | 2008-12-05 15:50:23 -0500 (Fri, 05 Dec
- 2008) | 8 lines Fix build errors on FreeBSD (uint -> unsigned
- int). (closes issue #14006) Reported by: alphaque Patches:
- astobj2.h-patch uploaded by alphaque (license 259) (Slightly
- modified by seanbright) ........
-
-2008-12-05 16:51 +0000 [r161354] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * utils/smsq.c: kill a warning
-
-2008-12-05 14:12 +0000 [r161287] Russell Bryant <russell@digium.com>
-
- * main/pbx.c: Fix a NULL format string warning found by buildbot.
-
-2008-12-04 18:30 +0000 [r161013] Jeff Peeler <jpeeler@digium.com>
-
- * main/rtp.c: (closes issue #13835) Reported by: matt_b Tested by:
- jpeeler This mirrors a check that was present in ast_rtp_read to
- also be in ast_rtp_raw_write to not schedule sending the receiver
- report if the remote RTCP endpoint address isn't present in the
- RTCP structure. Closes AST-142.
-
-2008-12-04 16:44 +0000 [r160943] Mark Michelson <mmichelson@digium.com>
-
- * main/callerid.c: Fix a callerid parsing issue. If someone
- formatted callerid like the following: "name <number>" (including
- the quotation marks), then the parts would be parsed as name:
- "name number: number This is because the closing quotation mark
- was not discovered since the number and everything after was
- parsed out of the string earlier. Now, there is a check to see if
- the closing quote occurs after the number, so that we can know if
- we should strip off the opening quote on the name. Closes AST-158
-
-2008-12-03 21:54 +0000 [r160770] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: Some compilers warn on null format strings;
- some don't (caught by buildbot)
-
-2008-12-03 21:38 +0000 [r160764] Jason Parker <jparker@digium.com>
-
- * channels/chan_agent.c: Only show this warning when we want to
- show it. (closes issue #13982) Reported by: coolmig Patches:
- chan_agent.c.patch uploaded by coolmig (license 621)
-
-2008-12-03 20:41 +0000 [r160703] Steve Murphy <murf@digium.com>
-
- * funcs/func_callerid.c: (closes issue #13597) Reported by:
- john8675309 Patches: patch.13597 uploaded by murf (license 17)
- Tested by: murf, john8675309 This patch causes the setcid func to
- update the CDR clid after setting the channel field.
-
-2008-12-03 17:55 +0000 [r160480-160570] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_iax2.c: During bridge code, the channel bridge may
- return a retry code, if a transfer was initiated but not yet
- completed. If the bridge is immediately retried, then we may send
- a storm of TXREQ packets, even though the first set is sent
- reliably (retransmitted). Fixes AST-137.
-
- * pbx/pbx_spool.c: If an entry is added to the directory during a
- scan when another entry expires, then that new entry will not be
- processed promptly, but must wait for either a future entry to
- start or a current entry's retry to occur. If no other entries
- exist in the directory (other than the new entries) when a bunch
- expire, then the new entries must wait until another new entry is
- added to be processed. This was a rather weird race condition,
- really. Fixes AST-147.
-
- * pbx/pbx_spool.c: Don't start scanning the directory until all
- modules are loaded, because some required modules (channels,
- apps, functions) may not yet be in memory yet. Fixes AST-149.
-
- * channels/chan_sip.c: Jon Bonilla (Manwe) pointed out on the -dev
- list: "I guess that having only ip-phones in mind is not a good
- approach. Since it is possible to have a sip proxy connected to
- asterisk we could receive a 407 (unauthorized) or 483 (too many
- hops) as response and dialog ending would not be a good
- behavior." So modified.
-
-2008-12-02 23:58 +0000 [r160390-160411] Terry Wilson <twilson@digium.com>
-
- * res/res_features.c: Channel is masqueraded, don't keep alive
-
- * res/res_features.c: A situation like A calls B, A builtin_atxfers
- B to C, C parks B would lead to a crash. Thanks to file for
- telling me how to fix it! (closes issue #13854) Reported by: Adam
- Lee Tested by: otherwiseguy
-
-2008-12-02 17:42 +0000 [r160297] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_sip.c: When the text does not match exactly (e.g.
- RTP/SAVP), then the %n conversion fails, and the resulting
- integer is garbage. Thus, we must initialize the integer and
- check it afterwards for success. (closes issue #14000) Reported
- by: folke Patches: asterisk-sipbg-sscanf-1.4.22.diff uploaded by
- folke (license 626) asterisk-sipbg-sscanf-1.6.0.1.diff uploaded
- by folke (license 626) asterisk-sipbg-sscanf-trunk-r159896.diff
- uploaded by folke (license 626)
-
-2008-12-02 01:16 +0000 [r160266] Terry Wilson <twilson@digium.com>
-
- * include/asterisk/astmm.h: make compile with dev mode and malloc
- debug
-
-2008-12-02 00:25 +0000 [r160207] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/stringfields.h, apps/app_voicemail.c,
- main/pbx.c, main/frame.c: Ensure that Asterisk builds with
- --enable-dev-mode, even on the latest gcc and glibc.
-
-2008-12-01 Tilghman Lesher <tilghman@digium.com>
-
- * Released 1.4.23-rc2
-
-2008-12-01 17:27 +0000 [r160003] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Apply some logic used in iax2_indicate() to
- iax2_setoption(), as well, since they both have the potential to
- send control frames in the middle of call setup. We have to wait
- until we have received a message back from the remote end before
- we try to send any more frames. Otherwise, the remote end will
- consider it invalid, and we'll get stuck in an INVAL/VNAK storm.
-
-2008-12-01 16:08 +0000 [r159976] Michiel van Baak <michiel@vanbaak.info>
-
- * main/manager.c: Get rid of the useless format string and argument
- in the Bogus/ manager channelname. Noted by kpfleming and name
- Bogus/manager suggested by eliel
-
-2008-12-01 14:52 +0000 [r159900] Russell Bryant <russell@digium.com>
-
- * .cleancount: Force a "make clean" to avoid a bizarre build issue
- ...
-
-2008-12-01 14:05 +0000 [r159897] Michiel van Baak <michiel@vanbaak.info>
-
- * main/manager.c: make manager compile on OpenBSD. The last (10th)
- argument to ast_channel_alloc here should be a pointer and NULL
- is not really a pointer.
-
-2008-11-29 16:58 +0000 [r159808] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/enum.c, utils/frame.c, configure, res/res_agi.c,
- include/asterisk/module.h, main/logger.c, main/dns.c,
- include/asterisk/threadstorage.h, include/asterisk/utils.h,
- include/asterisk/devicestate.h, channels/chan_sip.c,
- include/asterisk/dundi.h, main/jitterbuf.c,
- channels/chan_agent.c, configure.ac, utils/astman.c,
- include/asterisk/cli.h, include/asterisk/channel.h,
- include/jitterbuf.h, include/asterisk/manager.h,
- main/ast_expr2.c, Makefile, include/asterisk/logger.h,
- include/asterisk/res_odbc.h, main/srv.c, channels/chan_misdn.c,
- include/asterisk/linkedlists.h, include/asterisk/lock.h,
- include/asterisk/strings.h, makeopts.in,
- include/asterisk/stringfields.h, utils/check_expr.c,
- channels/chan_vpb.cc, res/res_features.c, channels/chan_iax2.c:
- update dev-mode compiler flags to match the ones used by default
- on Ubuntu Intrepid, so all developers will see the same warnings
- and errors since this branch already had some printf format
- attributes, enable checking for them and tag functions that
- didn't have them format attributes in a consistent way
-
-2008-11-26 20:21 +0000 [r159476-159571] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_oss.c, channels/busy.h (removed),
- channels/ring_tone.h (added), channels/chan_alsa.c,
- channels/ringtone.h (removed), channels/busy_tone.h (added),
- channels/Makefile: rename these files so as to avoid conflicts
- when users update their working copies and have unversioned files
- already in place
-
- * channels, agi/Makefile, utils/Makefile, channels/busy.h (added),
- Makefile.moddir_rules, Makefile.rules, channels/ringtone.h
- (added), channels/Makefile: simplify (and slightly bug-fix) the
- recent developer-oriented COMPILE_DOUBLE mode add channels/busy.h
- and channels/ringtone.h to the repository instead of generating
- them repeatedtly; most users do not change the settings to build
- them, but the Makefile rules are still there if they wish to do
- so ensure that 'make clean' removes dependency files for .i files
- that are created in COMPILE_DOUBLE mode
-
-2008-11-25 22:41 +0000 [r159316] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, channels/chan_iax2.c: (closes issue #12694) Reported
- by: yraber Patches: 12694.2nd.diff uploaded by murf (license 17)
- Tested by: murf, laurav Thanks to file (Joshua Colp) for his IAX
- fix. the change to cdr.c allows no-answer to percolate up into
- CDR's, and feels like the right place to locate this fix; if BUSY
- is done here, no-answer should be, too.
-
-2008-11-25 21:56 +0000 [r159246-159269] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_iax2.c: Don't try to send a response on a NULL pvt.
- (closes issue #13919) Reported by: barthpbx Patches:
- chan_iax2.c.patch uploaded by eliel (license 64) Tested by:
- barthpbx
-
- * /, channels/chan_iax2.c: Merged revisions 159245 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r159245 | tilghman | 2008-11-25 15:37:06 -0600 (Tue, 25
- Nov 2008) | 7 lines Regression fix for last security fix. Set the
- iseqno correctly. (closes issue #13918) Reported by: ffloimair
- Patches: 20081119__bug13918.diff.txt uploaded by Corydon76
- (license 14) Tested by: ffloimair ........
-
-2008-11-25 17:34 +0000 [r159158] Russell Bryant <russell@digium.com>
-
- * main/astobj2.c, include/asterisk/astobj2.h: Add ao2_trylock() to
- go along with ao2_lock() and ao2_unlock()
-
-2008-11-25 16:23 +0000 [r159096] Terry Wilson <twilson@digium.com>
-
- * apps/app_festival.c: Add missing variable declaration in the PPC
- code
-
-2008-11-25 04:50 +0000 [r159025] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_rpt.c, configure, include/asterisk/autoconfig.h.in,
- configure.ac: System call ioperm is non-portable, so check for
- its existence in autoconf. (Closes issue #13863)
-
-2008-11-22 00:04 +0000 [r158629] Jeff Peeler <jpeeler@digium.com>
-
- * include/asterisk/dahdi_compat.h, channels/chan_dahdi.c: (closes
- issue #13786) Reported by: tzafrir When compiling against Zaptel
- dahdi_compat will now only define all the DAHDI defines if the
- Zaptel define is present. Also, there is no such thing as
- DAHDI_PRI.
-
-2008-11-21 23:14 +0000 [r158603] Steve Murphy <murf@digium.com>
-
- * res/res_features.c: In reference to the fix made for 13871, I was
- merging the fix into 1.6.0 and realized I missed the code in the
- h-exten block, and didn't catch it because my test case had the
- h-exten commented out. So, this corrects the code I missed, as a
- preventative against another crash report. Tested with the
- h-exten defined, all is well.
-
-2008-11-21 23:07 +0000 [r158600] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c: The passed extension may not be the same in the list
- as the current entry, because we strip spaces when copying the
- extension into the structure. Therefore, use the copied item to
- place the item into the list. (found by lmadsen on -dev, fixed by
- me)
-
-2008-11-21 22:05 +0000 [r158539] Russell Bryant <russell@digium.com>
-
- * main/astobj2.c, include/asterisk/astobj2.h: When compiling with
- DEBUG_THREADS, report the real file/func/line for
- ao2_lock/ao2_unlock
-
-2008-11-21 21:19 +0000 [r158483] Steve Murphy <murf@digium.com>
-
- * res/res_features.c: (closes issue #13871) Reported by: mdu113
- This one is totally my fault. The code doesn't even create a
- bridge if the channel CDR has POST_DISABLED. I didn't check for
- that at the end of the bridge. Fixed with a few small insertions.
- Tested. Looks good. No cdr generated, no crash, no unnecc. data
- objects created either.
-
-2008-11-21 15:24 +0000 [r158053-158306] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: This change had somehow gotten reverted due to
- a completely unrelated commit. Thanks to Theo Belder on the
- Asterisk-dev list for pointing this out.
-
- * include/asterisk/file.h, main/frame.c, main/file.c,
- include/asterisk/frame.h: There was an issue when attempting to
- reference an embedded frame in a freed ast_filestream. This patch
- makes use of the ao2 functions to make sure that we do not free
- an ast_filestream structure until the embedded ast_frame has been
- "freed" as well. (closes issue #13496) Reported by: fst-onge
- Patches: filestream_frame_1_4.diff uploaded by putnopvut (license
- 60) Tested by: putnopvut Closes AST-89
-
- * channels/chan_sip.c: We don't handle 4XX responses to BYE well.
- According to section 15 of RFC 3261, we should terminate a dialog
- if we receive a 481 or 408 in response to our BYE. Since I am
- aware of at least one phone manufacturer who may sometimes send a
- 404 as well, I am being liberal and saying that any 4XX response
- to a BYE should result in a terminated dialog. (closes issue
- #12994) Reported by: pabelanger Patches: 12994.patch uploaded by
- putnopvut (license 60) Closes AST-129
-
- * apps/app_dial.c, channels/chan_sip.c: Make sure to set the hangup
- cause on the calling channel in the case that ast_call() fails.
- For incoming SIP channels, this was causing us to send a 603
- instead of a 486 when the call-limit was reached on the
- destination channel. (closes issue #13867) Reported by: still_nsk
- Patches: 13867.diff uploaded by putnopvut (license 60) Tested by:
- blitzrage
-
-2008-11-20 01:46 +0000 [r158010] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_misdn.c: Merged revision 157977 from
- https://origsvn.digium.com/svn/asterisk/team/group/issue8824
- ........ Fixes JIRA ABE-1726 The dial extension could be empty if
- you are using MISDN_KEYPAD to control ISDN provider features.
-
-2008-11-19 21:34 +0000 [r157859] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/stdtime/Makefile, codecs/gsm/src, main/db1-ast/btree,
- channels/misdn/Makefile, main/db1-ast/recno, pbx/ael, channels,
- main/db1-ast/Makefile, main/stdtime, main/db1-ast/hash,
- codecs/gsm/Makefile, main/db1-ast/db, Makefile.moddir_rules,
- channels/misdn, main/db1-ast/mpool, pbx/Makefile, Makefile.rules,
- res/snmp, res/Makefile: the gcc optimizer frequently finds broken
- code (use of uninitalized variables, unreachable code, etc.),
- which is good. however, developers usually compile with the
- optimizer turned off, because if they need to debug the resulting
- code, optimized code makes that process very difficult. this
- means that we get code changes committed that weren't adequately
- checked over for these sorts of problems. with this build system
- change, if (and only if) --enable-dev-mode was used and
- DONT_OPTIMIZE is turned on, when a source file is compiled it
- will actually be preprocessed (into a .i or .ii file), then
- compiled once with optimization (with the result sent to
- /dev/null) and again without optimization (but only if the first
- compile succeeded, of course). while making these changes, i did
- some cleanup work in Makefile.rules to move commonly-used
- combinations of flag variables into their own variables, to make
- the file easier to read and maintain
-
-2008-11-18 22:47 +0000 [r157503] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: Add some missing invite state changes
- necessary in the sip_write function. Not setting the invite state
- correctly on the call was resulting in the Record application
- leaving empty files. I also have updated the doxygen comment next
- to the declaration of the INV_EARLY_MEDIA constant to reflect
- that we also use this state when we *send* a 18X response to an
- INVITE. (closes issue #13878) Reported by: nahuelgreco Patches:
- sip-early-media-recording-1.4.22.patch uploaded by nahuelgreco
- (license 162) Tested by: putnopvut
-
-2008-11-18 19:13 +0000 [r157365] Jeff Peeler <jpeeler@digium.com>
-
- * apps/app_meetme.c: (closes issue #13899) Reported by: akkornel
- This fix is the result of a bug fix in ast_app_separate_args
- r124395. If an argument does not exist it should always be set to
- a null string rather than a null pointer.
-
-2008-11-18 18:25 +0000 [r157305] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_dial.c, channels/chan_local.c, res/res_features.c,
- include/asterisk/channel.h, apps/app_followme.c: Fix a crash in
- the end_bridge_callback of app_dial and app_followme which would
- occur at the end of an attended transfer. The error occurred
- because we initially stored a pointer to an ast_channel which
- then was hung up due to a masquerade. This commit adds a "fixup"
- callback to the bridge_config structure to allow for
- end_bridge_callback_data to be changed in the case that a new
- channel pointer is needed for the end_bridge_callback.
-
-2008-11-15 19:31 +0000 [r157104-157163] Kevin P. Fleming <kpfleming@digium.com>
-
- * Makefile, Makefile.rules: when an individual directory dist-clean
- is run, run clean in that directory first, and when running
- top-level dist-clean, do not run subdirectory clean operations
- twice
-
- * Makefile.moddir_rules: dist-clean should remove dependency
- information files as well
-
- * contrib/asterisk-ng-doxygen: major update to doxygen
- configuration file: 1) update to doxygen 1.5.x style file, as
- used in trunk 2) tell doxygen where are header files are, so
- include-file processing can be done 3) make all macros that are
- used to define variables/functions be expanded, so that doxygen
- will properly document the resulting variable/function 4) make
- all macros that are used to provide the contents of a variable
- (structure) be expanded, so that doxygen will be able to document
- the resulting fields 5) suppress compiler attributes
- (__attribute__(xxx)) from being seen by doxygen, so it will
- properly match up function definition and usage (for an example
- of th effect of this, look at the doxygen docs for ast_log() from
- before and afte this commit)
-
-2008-11-14 15:18 +0000 [r156816] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: If the prompt to reenter a voicemail
- password timed out, it resulted in the password not being saved,
- even if the input matched what you gave when first prompted to
- enter a new password. This is because the return value of
- ast_readstring was checked, but not checked properly. This bug
- was discovered by Jared Smith during an Asterisk training course.
- Thanks for reporting it!
-
-2008-11-14 00:41 +0000 [r156688-156755] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_while.c: ast_waitfordigit() requires that the channel be
- up, for no good logical reason. This prevents While/EndWhile from
- working within the "h" extension. Reported by: jgalarneau (for
- ABE C.2) Fixed by: me
-
- * main/manager.c: Provide more space for all the data which can
- appear in an originating channel name. (closes issue #13398)
- Reported by: bamby Patches: manager.c.diff uploaded by bamby
- (license 430)
-
-2008-11-13 11:58 +0000 [r156485-156510] Kevin P. Fleming <kpfleming@digium.com>
-
- * configure, autoconf/ast_gcc_attribute.m4: revert this change...
- non-functional changes don't belong here
-
- * configure, autoconf/ast_gcc_attribute.m4: correct minor syntax
- error... no functional change
-
-2008-11-12 21:18 +0000 [r156386] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_dial.c: When using call limits under 1 second, infinite
- call lengths are allowed, instead. (closes issue #13851) Reported
- by: ruddy
-
-2008-11-12 19:36 +0000 [r156297] Steve Murphy <murf@digium.com>
-
- * main/pbx.c: It turns out that the 0x0XX00 codes being returned
- for N, X, and Z are off by one, as per conversation with jsmith
- on #asterisk-dev; he was teaching a class and disconcerted that
- this published rule was not being followed, with patterns _NXX,
- _[1-8]22 and _[2-9]22... and NXX was winning, but [1-8] should
- have been. This change, tested on these 3 patterns now picks the
- proper one. However, this change may surprise users who set up
- dialplans based on previous behavior, which has been there for
- what, 2 and half years or so now.
-
-2008-11-12 19:26 +0000 [r156294] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_meetme.c: If the SLA thread is not started, then reload
- causes a memory leak. (closes issue #13889) Reported by: eliel
- Patches: app_meetme.c.patch uploaded by eliel (license 64)
-
-2008-11-12 19:10 +0000 [r156289] Jeff Peeler <jpeeler@digium.com>
-
- * apps/app_meetme.c: For whatever reason, gcc only warned me about
- the possible use of an uninitialized variable when compiling
- 1.6.1.
-
-2008-11-12 18:39 +0000 [r156229] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_iax2.c: Revert revision 132506, since it
- occasionally caused IAX2 HANGUP packets not to be sent, and
- instead, schedule a task to destroy the iax2 pvt structure 10
- seconds later. This allows the IAX2 HANGUP packet to be queued,
- transmitted, and ACKed before the pvt is destroyed. (closes issue
- #13645) Reported by: dzajro Patches:
- 20081111__bug13645__3.diff.txt uploaded by Corydon76 (license 14)
- Tested by: vazir Reviewed: http://reviewboard.digium.com/r/51/
-
-2008-11-12 17:53 +0000 [r156178] Jeff Peeler <jpeeler@digium.com>
-
- * apps/app_meetme.c: (closes issue #13173) Reported by: pep This
- change adds an announce_thread responsible for playing
- announcements to an existing conference. This allows all
- announcing to be immediately stopped if necessary but more
- importantly allows other threads that need to play something to
- not block. There are multiple benefits to this, but the actual
- bug is for solving the scenario for a channel to be unusable
- after hang up for the entire duration of the parting
- announcement. The parting announcement can be extremely long
- depending on what the user recorded upon joining the conference.
- Reviewed by Russell on Review Board:
- http://reviewboard.digium.com/r/25/
-
-2008-11-12 17:38 +0000 [r156167] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_dial.c: When doing some tests, I was having a crash at
- the end of every call if an attended transfer occurred during the
- call. I traced the cause to the CDR on one of the channels being
- NULL. murf suggested a check in the end bridge callback to be
- sure the CDR is non-NULL before proceeding, so that's what I'm
- adding.
-
-2008-11-12 17:29 +0000 [r156164] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c: Move the sanity check that makes sure "always
- fork" is not set along with the console option to be after the
- code that reads options from asterisk.conf. This resolves a
- situation where Asterisk can start taking up 100% when
- misconfigured. (Thanks to Bryce Porter (x86 on IRC) for letting
- me log in to his system to figure out what was causing the 100%
- CPU problem.)
-
-2008-11-10 21:07 +0000 [r155861] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_agent.c: Channel drivers assume that when their
- indicate callback is invoked, that the channel on which the
- callback was called is locked. This patch corrects an instance in
- chan_agent where a channel's indicate callback is called directly
- without first locking the channel. This was leading to some
- observed locking issues in chan_local, but considering that all
- channel drivers operate under the same expectations, the generic
- fix in chan_agent is the right way to go. AST-126
-
-2008-11-10 20:49 +0000 [r155803] Tilghman Lesher <tlesher@digium.com>
-
- * doc/valgrind.txt: I got tired of saying this in every single
- bugnote referring to this file.
-
-2008-11-09 01:08 +0000 [r155553] Sean Bright <sean.bright@gmail.com>
-
- * apps/app_dial.c, res/res_features.c, include/asterisk/channel.h,
- apps/app_followme.c: Use static functions here instead of nested
- ones. This requires a small change to the ast_bridge_config
- struct as well. To understand the reason for this change, see the
- following post:
- http://gcc.gnu.org/ml/gcc-help/2008-11/msg00049.html
-
-2008-11-07 22:27 +0000 [r155398] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_sip.c: Clarify error message. (closes issue #13809)
- Reported by: denke Patches: 20081104__bug13809.diff.txt uploaded
- by Corydon76 (license 14) Tested by: denke
-
-2008-11-06 19:45 +0000 [r155011] Mark Michelson <mmichelson@digium.com>
-
- * configs/voicemail.conf.sample: The documentation listed the
- ability to set 'maxmsg' per context. The truth is that you can
- only set this in the general section or per mailbox. Thus I am
- updating the sample config file to be more accurate. Thanks to
- sasargen on IRC for bringing up this issue.
-
-2008-11-05 16:44 +0000 [r154724] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_agent.c: The logic of a strcasecmp call was
- reversed (closes issue #13841) Reported by: clegall_proformatique
-
-2008-11-05 16:06 +0000 [r154685] Steve Murphy <murf@digium.com>
-
- * main/channel.c: This fix was prompted by communication from user,
- who was seeing thousands of error logs... looks like EAGAIN. Made
- such uninteresting.
-
-2008-11-04 20:49 +0000 [r154365] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_iax2.c: On busy systems, it's possible for the
- values checked within a single line of code to change, unless the
- structure is locked to ensure a consistent state. (closes issue
- #13717) Reported by: kowalma Patches: 20081102__bug13717.diff.txt
- uploaded by Corydon76 (license 14) Tested by: kowalma
-
-2008-11-04 19:01 +0000 [r154266] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_misdn.c: JIRA ABE-1703 mISDN sets the channel to
- the wrong state when it receives the indication
- AST_CONTROL_RINGING.
-
-2008-11-04 18:58 +0000 [r154060-154263] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_h323.c: Make the monitor thread non-detached, so it
- can be joined (suggested by Russell on -dev list).
-
- * apps/app_voicemail.c: Attempting to expunge a mailbox when the
- mailstream is NULL will crash Asterisk. (Closes issue #13829)
- Reported by: jaroth Patch by: me (modified jaroth's patch)
-
- * main/rtp.c: Remove the potential for a division by zero error.
- (Closes issue #13810)
-
-2008-11-03 13:01 +0000 [r153823] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_oss.c, channels/chan_dahdi.c, funcs/func_odbc.c,
- main/file.c, main/http.c, main/utils.c, pbx/pbx_config.c,
- res/res_jabber.c: somehow missed a bunch of gcc 4.3.x warnings in
- this branch on the first pass
-
-2008-11-02 19:51 +0000 [r153651] Russell Bryant <russell@digium.com>
-
- * include/asterisk/features.h: features.h depends on linkedlists.h,
- so include it
-
-2008-11-01 18:22 +0000 [r153337] Kevin P. Fleming <kpfleming@digium.com>
-
- * utils/frame.c, main/cli.c, utils/stereorize.c, main/channel.c,
- funcs/func_enum.c, channels/chan_dahdi.c, main/manager.c,
- channels/chan_skinny.c, main/ast_expr2f.c, res/res_agi.c,
- pbx/ael/ael_lex.c, main/http.c, channels/chan_alsa.c,
- pbx/ael/ael.flex, formats/format_gsm.c, apps/app_adsiprog.c,
- formats/format_wav.c, apps/app_festival.c,
- main/db1-ast/hash/hash_page.c, main/translate.c,
- res/res_crypto.c, agi/eagi-test.c, formats/format_ogg_vorbis.c,
- utils/astman.c, channels/chan_oss.c, agi/eagi-sphinx-test.c,
- pbx/ael/ael.tab.c, main/file.c, pbx/ael/ael.tab.h,
- apps/app_sms.c, pbx/pbx_dundi.c, res/res_indications.c,
- utils/streamplayer.c, apps/app_chanspy.c, main/asterisk.c,
- apps/app_voicemail.c, utils/muted.c, pbx/ael/ael.y,
- apps/app_authenticate.c, formats/format_wav_gsm.c,
- res/res_musiconhold.c, channels/chan_iax2.c: fix a bunch of
- potential problems found by gcc 4.3.x, primarily bare strings
- being passed to printf()-like functions and ignored results from
- read()/write() and friends
-
-2008-10-31 22:36 +0000 [r153270] Terry Wilson <twilson@digium.com>
-
- * res/res_features.c, apps/app_followme.c: Add end_bridge_callback
- for app_follome and add AUTOLOOP flag to res_features
-
-2008-10-31 Tilghman Lesher <tlesher@digium.com>
-
- * Asterisk 1.4.23-rc1 released.
-
-2008-10-31 16:30 +0000 [r153114] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_sip.c: Turn off qualify on uncached realtime peers.
- (Closes issue #13383)
-
-2008-10-31 15:45 +0000 [r153095] Terry Wilson <twilson@digium.com>
-
- * apps/app_dial.c, res/res_features.c, include/asterisk/channel.h:
- Recent CDR fixes moved execution of the 'h' exten into the
- bridging code, so variables that were set after ast_bridge_call
- was called would not show up in the 'h' exten. Added a callback
- function to handle setting variables, etc. from w/in the bridging
- code. Calls back into a nested function within the function
- calling ast_bridge_call (closes issue #13793) Reported by:
- greenfieldtech
-
-2008-10-30 20:58 +0000 [r152992] Sean Bright <sean.bright@gmail.com>
-
- * bootstrap.sh: The -I argument to aclocal needs a space before the
- include directory name.
-
-2008-10-30 20:33 +0000 [r152922-152958] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_h323.c: Cannot join detached threads. See
- http://www.opengroup.org/onlinepubs/000095399/functions/pthread_join.html
- (Closes issue #13400)
-
- * channels/chan_local.c: Unlock before returning, when extension
- doesn't exist. (closes issue #13807) Reported by: eliel Patches:
- chan_local.c.patch uploaded by eliel (license 64)
-
-2008-10-30 16:53 +0000 [r152811] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/cdr.c: instead of comparing the string pointer to 0, let's
- compare the value that was actually parsed out of the string
- (found by sparse)
-
-2008-10-29 05:23 +0000 [r152539] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Fix an incorrect usage of sizeof() (closes
- issue #13795) Reported by: andrew53 Patches:
- chan_sip_sizeof.patch uploaded by andrew53 (license 519)
-
-2008-10-29 05:19 +0000 [r152535-152538] Steve Murphy <murf@digium.com>
-
- * configs/features.conf.sample, apps/app_dial.c, apps/app_queue.c:
- A little documentation cross-ref between features and dial and
- queue... I wasted some time (stupidly) trying to get the
- one-touch parking stuff working, because it didn't occur to me
- that I had to also have the corresponding options in the dial
- command! Duh! (In all this time, I never set this up before!) So,
- to keep some poor fool from suffering the same fate, I made the
- features.conf.sample file mention the corresponding opts in
- dial/queue; and the docs for dial/app specifically mention the
- corresponding decls in the feature.conf file. I hope this doesn't
- spoil some vast, eternal plan...
-
- * apps/app_dial.c, res/res_features.c, funcs/func_channel.c,
- include/asterisk/pbx.h, apps/app_queue.c: The magic trick to
- avoid this crash is not to try to find the channel by name in the
- list, which is slow and resource consuming, but rather to pay
- attention to the result codes from the ast_bridge_call, to which
- I added the AST_PBX_NO_HANGUP_PEER_PARKED value, which now are
- returned when a channel is parked. If you get AST_PBX_KEEPALIVE,
- then don't touch the channel pointer. If you get
- AST_PBX_NO_HANGUP_PEER, or AST_PBX_NO_HANGUP_PEER_PARKED, then
- don't touch the peer pointer. Updated the several places where
- the results from a bridge were not being properly obeyed, and
- fixed some code I had introduced so that the results of the
- bridge were not overridden (in trunk). All the places that
- previously tested for AST_PBX_NO_HANGUP_PEER now have to check
- for both AST_PBX_NO_HANGUP_PEER and
- AST_PBX_NO_HANGUP_PEER_PARKED. I tested this against the 4 common
- parking scenarios: 1. A calls B; B answers; A parks B; B hangs up
- while A is getting the parking slot announcement, immediately
- after being put on hold. 2. A calls B; B answers; A parks B; B
- hangs up after A has been hung up, but before the park times out.
- 3. A calls B; B answers; B parks A; A hangs up while B is getting
- the parking slot announcement, immediately after being put on
- hold. 4. A calls B; B answers; B parks A; A hangs up after B has
- been hung up, but before the park times out. No crash. I also ran
- the scenarios above against valgrind, and accesses looked good.
-
-2008-10-28 22:32 +0000 [r152368-152463] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: Quoting in the wrong direction (Fixes
- AST-107)
-
- * apps/app_dial.c: Reset all DIAL variables back to blank, in case
- Dial is called multiple times per call (which could otherwise
- lead to inconsistent status reports). (closes issue #13216)
- Reported by: ruddy Patches: 20081014__bug13216.diff.txt uploaded
- by Corydon76 (license 14) Tested by: ruddy
-
-2008-10-27 23:28 +0000 [r152286] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c: Buffer policy setting for half is not
- needed.
-
-2008-10-27 21:32 +0000 [r152215] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_local.c: Inherit ALL elements of CallerID across a
- local channel. (closes issue #13368) Reported by: Peter Schlaile
- Patches: 20080826__bug13368.diff.txt uploaded by Corydon76
- (license 14)
-
-2008-10-26 20:23 +0000 [r152059] Sean Bright <sean.bright@gmail.com>
-
- * funcs/func_strings.c: Since passing \0 as the second argument to
- strchr is valid (and will match the trailing \0 of a string) we
- need to check that first, otherwise we end up with incorrect
- results. Fix suggested by reporter. (closes issue #13787)
- Reported by: meitinger
-
-2008-10-25 10:59 +0000 [r151905] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c: Move AMI initialization to occur after loading
- modules. This prevents a deadlock when someone tries to initiate
- a module reload from the AMI just as Asterisk is starting.
- (closes issue #13778) Reported by: hotsblanc Fix suggested by
- hotsblanc
-
-2008-10-23 16:04 +0000 [r151763] Terry Wilson <twilson@digium.com>
-
- * configs/features.conf.sample, res/res_features.c, CHANGES:
- Backport fix from 1.6.0 that allows you to set
- parkedcalltransfers=no|caller|callee|both, but default to both
- which would be the equivalent of the existing behaviour. The
- problem was that if someone parked a call, the callee and caller
- would both get assigned the builtin transfer feature, which would
- not only be potentially giving someone the ability to transfer
- themselves when they shouldn't have it, but would also dissallow
- reinviting the media off of the call. (closes issue #12854)
- Reported by: davidw Patches: parkingfix4.diff.txt uploaded by
- otherwiseguy Tested by: davidw, otherwiseguy
-
-2008-10-20 04:57 +0000 [r151240-151241] Kevin P. Fleming <kpfleming@digium.com>
-
- * autoconf/ast_check_pwlib.m4, autoconf/ast_check_openh323.m4,
- configure.ac: rename this macro to properly reflect what it does
-
- * autoconf/ast_check_pwlib.m4 (added), autoconf (added),
- autoconf/acx_pthread.m4 (added), autoconf/ast_func_fork.m4
- (added), configure, autoconf/ast_gcc_attribute.m4 (added),
- bootstrap.sh, autoconf/ast_check_gnu_make.m4 (added),
- autoconf/ast_ext_lib.m4 (added), autoconf/ast_prog_ld.m4 (added),
- autoconf/ast_c_compile_check.m4 (added),
- autoconf/ast_c_define_check.m4 (added),
- autoconf/ast_prog_egrep.m4 (added),
- autoconf/ast_check_openh323.m4 (added),
- autoconf/ast_prog_ld_gnu.m4 (added), autoconf/ast_prog_sed.m4
- (added), acinclude.m4 (removed): break up acinclude.m4 into
- individual files, which will make it easier to maintain, easier
- to add new macros (less patching) and will ease maintenance of
- these macros across Asterisk branches
-
-2008-10-19 19:51 +0000 [r151100-151167] BJ Weschke <bweschke@btwtech.com>
-
- * main/asterisk.c: As per kpfleming's comments to the prior commit,
- I'm reverting some of the changes here. A comment was made in bug
- #13726 "3. The same mistake as in (2) is done in a few other
- places in the code that check for: #if defined(HAVE_ZAPTEL) ||
- defined(HAVE_DAHDI) Harmless, but still incorrect." In the case
- of main/asterisk.c, this is not incorrect because without
- HAVE_ZAPTEL defined, we're missing the include for ioctl and the
- namespace that defines DAHDI_TIMERCONFIG which is still required
- when using Zaptel with the 1.4 branch.
-
- * main/asterisk.c: Fix the 1.4 branch compile again broken with
- r150557 when using with Zaptel and not DAHDI (closes issue
- #13740) reported by: jmls patch by: bweschke
-
-2008-10-18 01:42 +0000 [r150816] BJ Weschke <bweschke@btwtech.com>
-
- * main/manager.c: Using the GetVar handler in AMI is potentially
- dangerous (insta-crash [tm]) when you use a dialplan function
- that requires a channel and then you don't provide one or provide
- an invalid one in the Channel: parameter. We'll handle this
- situation exactly the same way it was handled in pbx.c back on
- r61766. We'll create a bogus channel for the function call and
- destroy it when we're done. If we have trouble allocating the
- bogus channel then we're not going to try executing the function
- call at all and run the risk of crashing. (closes issue #13715)
- reported by: makoto patch by: bweschke
-
-2008-10-17 17:18 +0000 [r150637] Steve Murphy <murf@digium.com>
-
- * res/res_features.c: Interesting crash. In this case, you exit the
- bridge with peer completely GONE. I moved the channel find call
- up to cover the whole peer CDR reset code segment. This appears
- to solve the crash without changing the logic at all.
-
-2008-10-17 15:31 +0000 [r150557] Jason Parker <jparker@digium.com>
-
- * main/asterisk.c, main/channel.c, channels/chan_dahdi.c,
- configure, configure.ac: Correctly allow chan_dahdi to compile
- against older versions of Zaptel. Don't always define
- HAVE_ZAPTEL_CHANALARMS (since we check if it's defined..) Minor
- cleanup to make things clear. (closes issue #13726) Reported by:
- tzafrir Patches: dahdi_def.diff uploaded by tzafrir (license 46)
-
-2008-10-16 23:40 +0000 [r150298-150304] Mark Michelson <mmichelson@digium.com>
-
- * main/manager.c: Reverting changes from commits 150298 and 150301
- since I was mistakenly under the assumption that dialplan
- functions *always* required that a channel be present. I need to
- go home earlier, I think :)
-
- * main/manager.c: And don't forget to return on the error condition
-
- * main/manager.c: Don't try to call a dialplan function's read
- callback from the manager's GetVar handler if an invalid channel
- has been specified. Several dialplan functions, including CHANNEL
- and SIP_HEADER, do not check for NULL-ness of the channel being
- passed in. (closes issue #13715) Reported by: makoto
-
-2008-10-16 15:56 +0000 [r150124] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_misdn.c: Fix memory leak found by customer
-
-2008-10-16 15:26 +0000 [r150056] Steve Murphy <murf@digium.com>
-
- * cdr/cdr_odbc.c: This patch is relevant to: ABE-1628 and
- RYM-150398 and AST-103 in internal Digium bug trackers. These
- fixes address a really subtle memory corruption problem that
- would happen in machines heavily loaded in production
- environments. The corruption would always take the form of the
- STMT object getting nulled out and one of the unixODBC calls
- would crash trying to access statement->connection. It isn't
- fully proven yet, but the server has now been running 2.5 days
- without appreciable memory growth, or any gain of %cpu, and no
- crashes. Whether this is the problem or not on that server, these
- fixes are still warranted. As it turns out, **I** introduced
- these errors unwittingly, when I corrected another crash earlier.
- I had formed the build_query routine, and failed to remove
- mutex_unlock calls in 3 places in the transplanted code. These
- unlocks would only happen in error situations, but unlocking the
- mutex early set the code up for a catastrophic failure, it
- appears. It would happen only once every 100K-200K or more calls,
- under heavy load... but that is enough. If another crash occurs,
- with the same MO, I'll come back and remove my confession from
- the log, and we'll keep searching, but the fact that we have
- Asterisk dying from an asynchronous wiping of the STMT object,
- only on some connection error, and that the server has lived for
- 2.5 days on this code without a crash, sure make it look like
- this was the problem! Also, in several points, Statement handles
- are set to NULL after SQLFreeHandle. This was mainly for
- insurance, to guarantee a crash. As it turns out, the code does
- not appear to be attempting to use these freed pointers. Asterisk
- owes a debt of gratitude to Federico Alves and Frediano Ziglio
- for their untiring efforts in finding this bug, among others.
-
-2008-10-15 21:34 +0000 [r149683-149840] BJ Weschke <bweschke@btwtech.com>
-
- * CHANGES: Another documentation fix. (closes issue #13708)
-
- * configs/agents.conf.sample: An update to the
- documentation/example of agents.conf.sample with the correct
- parameter for this feature as defined in chan_agent.c (closes
- issue #13709)
-
-2008-10-15 10:30 +0000 [r149452] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_sip.c: fix some problems when parsing SIP messages
- that have the maximum number of headers or body lines that we
- support
-
-2008-10-14 23:43 +0000 [r149130-149266] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: Change this warning to an error message.
- Suggestion comes from Sean Bright. Thanks Sean!
-
- * channels/chan_sip.c: Call register_peer_exten even in the case
- that the peer's IP/port does not change. (closes issue #13309)
- Reported by: dimas Patches: v2-13309.patch uploaded by dimas
- (license 88)
-
- * include/asterisk/audiohook.h, main/audiohook.c: Add a tolerance
- period for sync-triggered audiohooks so that if packetization of
- audio is close (but not equal) we don't end up flushing the
- audiohooks over small inconsistencies in synchronization. Related
- to issue #13005, and solves the issue for most people who were
- experiencing the problem. However, a small number of people are
- still experiencing the problem on long calls, so I am not closing
- the issue yet
-
- * apps/app_queue.c: Update the queue with the correct number of
- calls and whether the call was completed within the service level
- when a transfer takes place. This way, we do not "break" the
- leastrecent and fewestcalls strategies by not logging a call
- until after the transferred call has ended. (closes issue #13395)
- Reported by: Marquis Patches: app_queue.c.transfer.patch uploaded
- by Marquis (license 32)
-
- * channels/chan_sip.c: Don't allow reserved characters to be used
- in register lines in sip.conf. (closes issue #13570) Reported by:
- putnopvut
-
-2008-10-14 20:09 +0000 [r149061] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_waitforsilence.c: Check correct values in the return of
- ast_waitfor(); also, get rid of a possible memory leak. (closes
- issue #13658) Reported by: explidous Patch by: me
-
-2008-10-14 19:05 +0000 [r148990] Leif Madsen <lmadsen@digium.com>
-
- * CHANGES: Add in some missing updates to the CHANGES file for
- sip.conf (closes issue #13100) Reported and patch by:
- gknispel_proformatique
-
-2008-10-14 19:03 +0000 [r148916-148987] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: Some compilers warn, some don't. Fixing.
-
- * apps/app_voicemail.c: Ensure that mail headers are 7-bit clean,
- even when UTF-8 characters are used in headers like 'Subject' and
- 'To'. Closes AST-107.
-
-2008-10-14 17:33 +0000 [r148912] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_local.c: Deadlock prevention in chan_local. (closes
- issue #13676) Reported by: tacvbo Patches: 13676.patch uploaded
- by putnopvut (license 60) Tested by: tacvbo
-
-2008-10-14 10:30 +0000 [r148611-148736] Kevin P. Fleming <kpfleming@digium.com>
-
- * Makefile: on Ubuntu (at least), recent versions of ld in binutils
- delete all debugging symbols when -x is supplied; since the
- reasons why -x is being passed are lost in the mists of time,
- remove it so debugging will work properly
-
- * main/translate.c: it would be nice if this message printing code
- had actually been tested before it was committed...
-
-2008-10-10 16:25 +0000 [r147997-148257] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: User not notified of temporary greeting, if
- ODBC storage is in use. (closes issue #13659) Reported by:
- moliveras Patches: 20081009__bug13659.diff.txt uploaded by
- Corydon76 (license 14) Tested by: moliveras
-
- * apps/app_voicemail.c: When blank, callerid name and number should
- display "unknown caller" in voicemail emails. (Closes issue
- #13643)
-
-2008-10-09 18:56 +0000 [r147941] Jeff Peeler <jpeeler@digium.com>
-
- * res/res_features.c: (closes issue #13139) Reported by: krisk84
- Tested by: krisk84 This change prevents a call that is placed in
- the parkinglot to be picked up before the PBX is finished. If
- another extension dials the parking extension before the PBX
- thread has completed at minimum warnings will occur about the PBX
- not properly being terminated. At worst, a crash could occur.
-
-2008-10-08 22:22 +0000 [r147681] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_dahdi.c: when parsing a text configuration option,
- ensure that the buffer on the stack is actually large enough to
- hold the legal values of that option, and also ensure that
- sscanf() knows to stop parsing if it would overrun the buffer
- (without these changes, specifying "buffers=...,immediate" would
- overflow the buffer on the stack, and could not have worked as
- expected)
-
-2008-10-08 14:51 +0000 [r147517] Joshua Colp <jcolp@digium.com>
-
- * apps/app_speech_utils.c: If we receive DTMF make sure that the
- state of the speech structure goes back to being not ready.
- (issue #LUMENVOX-8)
-
-2008-10-07 23:14 +0000 [r147429-147430] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_dahdi.c: revert this change until i can understand
- why it results in locking order changes
-
- * channels/chan_dahdi.c: don't start a PBX on incoming PRI call
- channels until after we're done setting channel variables and
- other things on the channel, otherwise the channel might go away
- (if the dialplan hangs up quickly) before we are done, which
- results in a spectacular crash
-
-2008-10-07 16:48 +0000 [r147193] Sean Bright <sean.bright@gmail.com>
-
- * apps/app_voicemail.c: Make 'imapsecret' an alias to
- 'imappassword' in voicemail.conf.
-
-2008-10-06 20:52 +0000 [r146711-146799] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_callerid.c, apps/app_speech_utils.c,
- funcs/func_curl.c, funcs/func_groupcount.c, res/res_smdi.c,
- channels/chan_sip.c, funcs/func_timeout.c, funcs/func_odbc.c,
- funcs/func_cdr.c, funcs/func_math.c, channels/chan_iax2.c:
- Dialplan functions should not actually return 0, unless they have
- modified the workspace. To signal an error (and no change to the
- workspace), -1 should be returned instead. (closes issue #13340)
- Reported by: kryptolus Patches: 20080827__bug13340__2.diff.txt
- uploaded by Corydon76 (license 14)
-
- * channels/chan_local.c: Check whether an extension exists in the
- _call method, rather than the _alloc method, because we need to
- evaluate the callerid (since that data affects whether an
- extension exists). (closes issue #13343) Reported by: efutch
- Patches: 20080915__bug13343.diff.txt uploaded by Corydon76
- (license 14) Tested by: efutch
-
-2008-10-06 15:57 +0000 [r146643] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_dahdi.c: ensure that the private structure for
- pseudo channels is created without 'leaking' configuration data
- from other configured channels (closes issue #13555) Reported by:
- jeffg Patches: issue_13555.patch uploaded by kpfleming (license
- 421) Tested by: jeffg
-
-2008-10-05 21:17 +0000 [r146448] Jason Parker <jparker@digium.com>
-
- * channels/chan_sip.c: Fix silly formatting.
-
-2008-10-03 22:51 +0000 [r146244] Sean Bright <sean.bright@gmail.com>
-
- * apps/app_rpt.c: Change some preprocessor macros to struct
- definitions so that we get app_rpt to build with DAHDI. (closes
- issue #13576) Reported by: blitzrage
-
-2008-10-03 20:44 +0000 [r146129] Jeff Peeler <jpeeler@digium.com>
-
- * include/asterisk/features.h, res/res_features.c, res/res_agi.c:
- (closes issue #13425) Reported by: mdu113 Tested by: mdu113
- Similar to r143204, masquerade the channel in the case of Park
- being called from AGI.
-
-2008-10-03 17:12 +0000 [r146026] Steve Murphy <murf@digium.com>
-
- * res/res_features.c: (closes issue #13579) Reported by: dwagner
- (closes issue #13584) Reported by: dwagner Tested by: murf,
- putnopvut The thought occurred to me that the res= from the
- extension spawn was ending up being returned from the bridge.
- "Thou shalt not poison the return value". Made the change and it
- appears to allow blind xfers to work as normal. If I'm wrong,
- reopen the bugs. But it looks good to me! Many thanks to
- putnopvut for helping me reproduce this!
-
-2008-10-02 16:39 +0000 [r145751-145839] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_odbc.c: Backport support for some of the keyword
- modifications used in 1.6 (while warning that some options aren't
- really supported) and add some warning messages. Some credit to
- oej, who was complaining in #asterisk-dev.
-
- * res/res_odbc.c: Some sanity checks that may have led to prior
- crashes, found by codefreeze-lap (murf) on IRC. Also some cleanup
- of incorrectly-used constants.
-
-2008-10-01 17:18 +0000 [r145479] Leif Madsen <lmadsen@digium.com>
-
- * contrib/scripts/realtime_pgsql.sql: Update the realtime_pgsql.sql
- script to create the setinterfacevar column. (closes issue
- #13549) Reported by: fiddur
-
-2008-10-01 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.22 released.
-
-2008-09-09 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.22-rc5 released.
-
-2008-09-09 15:40 +0000 [r142063] Russell Bryant <russell@digium.com>
-
- * res/res_features.c: Ensure that the stored CDR reference is still
- valid after the bridge before poking at it. Also, keep the
- channel locked while messing with this CDR. (fixes crashes
- reported in issue #13409)
-
-2008-09-08 21:10 +0000 [r141809] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: Fix pedantic mode of chan_sip to only check
- the remote tag of an endpoint once a dialog has been confirmed.
- Up until that point, it is possible and legal for the far-end to
- send provisional responses with a different To: tag each time.
- With this patch applied, these provisional messages will not
- cause a matching problem. (closes issue #11536) Reported by: ibc
- Patches: 11536v2.patch uploaded by putnopvut (license 60)
-
-2008-09-08 21:02 +0000 [r141806] Russell Bryant <russell@digium.com>
-
- * main/pbx.c: When doing an async goto, detect if the channel is
- already in the middle of a masquerade. This can happen when
- chan_local is trying to optimize itself out. If this happens,
- fail the async goto instead of bursting into flames. (closes
- issue #13435) Reported by: geoff2010
-
-2008-09-08 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.22-rc4 released.
-
-2008-09-08 20:15 +0000 [r141741] Jason Parker <jparker@digium.com>
-
- * Makefile, redhat (removed): Remove RPM package targets from
- Makefile (and all associated parts). This has never worked in
- 1.4, and we decided that it makes no sense to be done here. There
- are many distros out there that already have "proper" spec files
- that can be (re)used. Closes issue #13113 Closes issue #10950
- Closes issue #10952
-
-2008-09-08 16:26 +0000 [r141678] Russell Bryant <russell@digium.com>
-
- * configure, configure.ac: Actually use Zaptel CFLAGS if using
- Zaptel instead of DAHDI This fixes building against Zaptel when
- using a custom path
-
-2008-09-06 20:13 +0000 [r141565] Steve Murphy <murf@digium.com>
-
- * channels/chan_sip.c: This fix comes from Joshua Colp The
- Brilliant, who, given the trace, came up with a solution. This
- will most likely will close 13235 and 13409. I'll wait till
- Monday to verify, and then close these bugs.
-
-2008-09-06 15:23 +0000 [r141503] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_agi.c: Reverting behavior change (AGI should not exit
- non-zero on SUCCESS) (closes issue #13434) Reported by:
- francesco_r
-
-2008-09-05 21:10 +0000 [r141217-141366] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_agent.c: Agent's should not try to call a channel's
- indicate callback if the channel has been hung up. It will likely
- crash otherwise ABE-1159
-
- * apps/app_voicemail.c: Since greetings are not stored in IMAP, we
- should not be DISPOSE'ing of them the same way we do with other
- messages. (closes issue #13414) Reported by: mthomasslo Patches:
- 13414v2.patch uploaded by putnopvut (license 60) Tested by:
- mthomasslo
-
- * channels/chan_sip.c: Commit 140417 had a logic flaw in it which
- caused port 5060 to always be used when dialing a peer if no
- explicit port was specified. This broke the behavior of
- implicitly using the port from which the peer registered if no
- port is specified. This commit fixes the logic flaw. (closes
- issue #13424) Reported by: mdu113 Patches: 13424.patch uploaded
- by putnopvut (license 60) Tested by: mdu113
-
-2008-09-05 14:15 +0000 [r141094-141156] Steve Murphy <murf@digium.com>
-
- * main/channel.c: A small change to prevent double-posting of
- CDR's; thanks to Daniel Ferrer for bringing it to our attention
-
- * pbx/ael/ael-test/ref.ael-vtest25 (added),
- pbx/ael/ael-test/ael-vtest25/extensions.ael (added),
- pbx/ael/ael-test/ael-vtest25 (added), pbx/ael/ael_lex.c,
- pbx/ael/ael-test/ref.ael-test6, pbx/ael/ael.flex: (closes issue
- #13357) Reported by: pj Tested by: murf (closes issue #13416)
- Reported by: yarns Tested by: murf If you find this message
- overly verbose, relax, it's probably not meant for you. This
- message is meant for probably only two people in the whole world:
- me, or the poor schnook that has to maintain this code because
- I'm either dead or unavailable at the moment. This fix solves two
- reports, both having to do with embedding a function call in a
- ${} construct. It was tricky because the funccall syntax has
- parenthesis () in it. And up till now, the 'word' token in the
- flex stuff didn't allow that, because it would tend to steal the
- LP and RP tokens. To be truthful, the "word" token was the
- trickiest, most unstable thing in the whole lexer. I was lucky it
- made this long without complaints. I had to choose every
- character in the pattern with extreme care, and I knew that
- someday I'd have to revisit it. Well, the day has come. So, my
- brilliant idea (and I'm being modest), was to use the surrounding
- ${} construct to make a state machine and capture everything in
- it, no matter what it contains. But, I have to now treat the word
- token like I did with comments, in that I turn the whole thing
- into a state-machine sort of spec, with new contexts
- "curlystate", "wordstate", and "brackstate". Wait a minute,
- "brackstate"? Yes, well, it didn't take very many regression
- tests to point out if I do this for ${} constructs, I also have
- to do it with the $[] constructs, too. I had to create a separate
- pcbstack2 and pcbstack3 because these constructs can occur inside
- macro argument lists, and when we have two state machines
- operating on the same structures we'd get problems otherwise. I
- guess I could have stopped at pcbstack2 and had the brackstate
- stuff share it, but it doesn't hurt to be safe. So, the pcbpush
- and pcbpop routines also now have versions for "2" and "3". I had
- to add the {KEYWORD} construct to the initial pattern for "word",
- because previously word would match stuff like "default7",
- because it was a longer match than the keyword "default". But,
- not any more, because the word pattern only matches only one or
- two characters now, and it will always lose. So, I made it the
- winner again by making an optional match on any of the keywords
- before it's normal pattern. I added another regression test to
- make sure we don't lose this in future edits, and had to fix just
- one regression, where it no longer reports a 'cascaded' error,
- which I guess is a plus. I've given some thought as to whether to
- apply these fixes to 1.4 and the 1.6.x releases, vs trunk; I
- decided to put it in 1.4 because one of the bug reports was
- against 1.4; and it is unexpected that AEL cannot handle this
- situation. It actually reduced the amount of useless "cascade"
- error messages that appeared in the regressions (by one line,
- ehhem). There is a possible side-effect in that it does now do
- more careful checking of what's in those ${} constructs, as far
- as matching parens, and brackets are concerned. Some users may
- find a an insidious problem and correct it this way. This should
- be exceedingly rare, I hope.
-
-2008-09-04 17:00 +0000 [r141028] Jeff Peeler <jpeeler@digium.com>
-
- * res/res_features.c, res/res_agi.c: (closes issue #11979) Fixes
- multiple parking problems: Crash when executing a park on an
- extension dialed by AGI due to not returning the proper return
- code. Crash when using a builtin feature that was a subset of a
- enabled dynamic feature. Crash due to always hanging up the peer
- despite the fact that the peer was supposed to be parked.
-
-2008-09-03 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.22-rc3 released.
-
-2008-09-03 14:29 +0000 [r140850] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Fix voicemail forwarding when using ODBC
- storage. (closes issue #13387) Reported by: moliveras Patches:
- 13387.patch uploaded by putnopvut (license 60) Tested by:
- putnopvut, moliveras
-
-2008-09-03 13:24 +0000 [r140816] Russell Bryant <russell@digium.com>
-
- * main/poll.c: Don't freak out if the poll emulation receives NULL
- for the pollfds array (closes issue #13307) Reported by: jcovert
-
-2008-09-02 23:47 +0000 [r140751] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: After adding the context checking to
- app_voicemail for IMAP storage, I left out a crucial place to
- copy the context to the vm_state structure. This is the
- correction.
-
-2008-09-02 23:36 +0000 [r140670-140747] Steve Murphy <murf@digium.com>
-
- * main/cdr.c: I am turning the warnings generated in ast_cdr_free
- and post_cdr into verbose level 2 messages. Really, they matter
- little to end users. You either get the CDR's you wanted, or you
- don't, and it is a bug.
-
- * main/channel.c: After reconsidering, with respect to 13409,
- ast_cdr_detach should be OK, better in fact, than ast_cdr_free,
- which generates lots of useless warnings that will undoubtably
- generate complaints.
-
- * main/channel.c, main/pbx.c: (closes issue #13409) Reported by:
- tomaso Patches: asterisk-1.6.0-rc2-cdrmemleak.patch uploaded by
- tomaso (license 564) I basically spent the day, verifying that
- this patch solves the problem, and doesn't hurt in non-problem
- cases. Why valgrind did not plainly reveal this leak absolutely
- mystifies and stuns me. Many, many thanks to tomaso for finding
- and providing the fix.
-
-2008-09-02 18:14 +0000 [r140605] Sean Bright <sean.bright@gmail.com>
-
- * channels/chan_iax2.c: Make sure to use the correct length of the
- mohinterpret and mohsuggest buffers when copying configuration
- values. (closes issue #13336) Reported by:
- decryptus_proformatique Patches:
- chan_iax2_mohinterpret_mohsuggest_general_settings.patch uploaded
- by decryptus (license 555)
-
-2008-08-29 17:34 +0000 [r140417-140488] Mark Michelson <mmichelson@digium.com>
-
- * main/manager.c, apps/app_queue.c, channels/chan_iax2.c: After
- working on the ao2_containers branch, I noticed something a bit
- strange. In all cases where we provide a callback function to
- ao2_container_alloc, the callback function would only return 0 or
- CMP_MATCH. After inspecting the ao2_callback() code carefully, I
- found that if you're only looking for one specific item, then you
- should return CMP_MATCH | CMP_STOP. Otherwise, astobj2 will
- continue traversing the current bucket until the end searching
- for more matches. In cases like chan_iax2 where in 1.4, all the
- peers are shoved into a single bucket, this makes for potentially
- terrible performance since the entire bucket will be traversed
- even if the peer is one of the first ones come across in the
- bucket. All the changes I have made were for cases where the
- callback function defined was passed to ao2_container_alloc so
- that calls to ao2_find could find a unique instance of whatever
- object was being stored in the container.
-
- * apps/app_voicemail.c: Add context checking when retrieving a
- vm_state. This was causing a problem for people who had
- identically named mailboxes in separate voicemail contexts. This
- commit affects IMAP storage only. (closes issue #13194) Reported
- by: moliveras Patches: 13194.patch uploaded by putnopvut (license
- 60) Tested by: putnopvut, moliveras
-
- * channels/chan_sip.c: Fix SIP's parsing so that if a port is
- specified in a string to Dial(), it is not ignored. (closes issue
- #13355) Reported by: acunningham Patches: 13355v2.patch uploaded
- by putnopvut (license 60) Tested by: acunningham
-
-2008-08-27 19:49 +0000 [r140299] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: Fix tag checking in get_sip_pvt_byid_locked
- when in pedantic mode. The problem was that the wrong tags would
- be compared depending on the direction of the call. (closes issue
- #13353) Reported by: flefoll Patches:
- chan_sip.c.br14.139015.patch-refer-pedantic uploaded by flefoll
- (license 244)
-
-2008-08-26 16:49 +0000 [r140115] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c: add HAVE_PRI if define around
- dahdi_close_pri_fd
-
-2008-08-26 16:07 +0000 [r140060] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Fix some bogus scheduler usage in chan_sip.
- This code used the return value of a completely unrelated
- function to determine whether the scheduler should be run or not.
- This would have caused the scheduler to not run in cases where it
- should have. Also, leave a note about another scheduler issue
- that needs to be addressed at some point.
-
-2008-08-26 15:57 +0000 [r140056] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c: (closes issue #12071) Reported by: tzafrir
- Patches: dahdi_close.diff uploaded by tzafrir (license 46) Tested
- by: tzafrir, jpeeler This patch fixes closing open file
- descriptors in the case of an error.
-
-2008-08-26 15:27 +0000 [r140051] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix a race condition with the IAX scheduler
- thread. A lock and condition are used here to allow newly
- scheduled tasks to wake up the scheduler just in case the new
- task needs to run sooner than the current wakeup time when the
- thread is sleeping. However, there was a race condition such that
- a newly scheduled task would not properly wake up the scheduler
- or affect the wake up period. The order of execution would have
- been: 1) Scheduler thread determines wake up time of N ms. 2)
- Another thread schedules a task and signals the condition, with
- an execution time of < N ms. 3) Scheduler thread locks and goes
- to sleep for N ms. By moving the sleep time determination to
- inside the critical section, this possibility is avoided.
-
-2008-08-26 15:22 +0000 [r140050] Terry Wilson <twilson@digium.com>
-
- * Makefile: sounds/Makefile installs sounds using the "new"
- language directory structure, but languageprefix needs to be set
- = yes for sounds in subdirectories (digits/1, etc.) to play as
- the correct language. Fix the generation of asterisk.conf to
- include languageprefix=yes
-
-2008-08-26 14:09 +0000 [r140029] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_dahdi.c: correct a file location in an error
- message
-
-2008-08-25 21:47 +0000 [r139927] Jeff Peeler <jpeeler@digium.com>
-
- * main/manager.c: Fix a typo I made. Lesson learned, apply the
- patch if one exists.
-
-2008-08-25 21:31 +0000 [r139909] Sean Bright <sean.bright@gmail.com>
-
- * build_tools/get_moduleinfo, build_tools/get_makeopts: Some
- versions of awk (nawk, for example) don't like empty regular
- expressions so be slightly more verbose. (closes issue #13374)
- Reported by: dougm Patches: 13374.diff uploaded by seanbright
- (license 71) Tested by: dougm
-
-2008-08-25 20:46 +0000 [r139869] Terry Wilson <twilson@digium.com>
-
- * channels/chan_sip.c: Make SIPADDHEADER() propagate indefinitely
-
-2008-08-25 15:52 +0000 [r139769] Mark Michelson <mmichelson@digium.com>
-
- * main/config.c: Fix the logic in config_text_file_save so that if
- an UpdateConfig manager action is issued and the file specified
- in DstFileName does not yet exist, an error is not returned.
- (closes issue #13341) Reported by: vadim Patches: 13341.patch
- uploaded by putnopvut (license 60) (with small modification from
- seanbright)
-
-2008-08-25 15:33 +0000 [r139764] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, res/res_features.c: This patch reverts the changes
- made via 139347, and 139635, as users are seeing adverse
- difference. I will un-close 13251. Back to the drawing board/
- concept/ beginning/ whatever!
-
-2008-08-22 22:24 +0000 [r139635] Steve Murphy <murf@digium.com>
-
- * res/res_features.c: I found some problems with the code I
- committed earlier, when I merged them into trunk, so I'm coming
- back to clean up. And, in the process, I found an error in the
- code I added to trunk and 1.6.x, that I'll fix using this patch
- also.
-
-2008-08-22 21:36 +0000 [r139621] Jeff Peeler <jpeeler@digium.com>
-
- * main/manager.c: (closes issue #13359) Reported by: Laureano
- Patches: originate_channel_check.patch uploaded by Laureano
- (license 265)
-
-2008-08-22 19:45 +0000 [r139456-139553] Mark Michelson <mmichelson@digium.com>
-
- * include/asterisk/threadstorage.h: Fix compilation when
- DEBUG_THREAD_LOCALS is selected (closes issue #13298) Reported
- by: snuffy Patches: bug13298_20080822.diff uploaded by snuffy
- (license 35)
-
- * main/frame.c: Remove show_frame_stats_deprecated since it is not
- used anywhere and causes build errors if building under dev-mode
- with TRACE_FRAMES selected in menuselect. (closes issue #13362)
- Reported by: snuffy
-
- * channels/chan_iax2.c: Fix the build. Thanks, mvanbaak!
-
- * channels/chan_iax2.c: Prevent a deadlock in chan_iax2 resulting
- from incorrect locking order between iax2_pvt and ast_channel
- structures. AST-13
-
-2008-08-21 23:39 +0000 [r139387] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c: Fixes loop that could possibly never exit
- in the event of a channel never being able to be opened or
- specify after a restart. (closes issue #11017)
-
-2008-08-21 23:03 +0000 [r139347] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, res/res_features.c: (closes issue #13251) Reported
- by: sergee Tested by: murf THis is a bold move for a static
- release fix, but I wouldn't have made it if I didn't feel
- confident (at least a *bit* confident) that it wouldn't mess
- everyone up. The reasoning goes something like this: 1. We simply
- cannot do anything with CDR's at the current point (in pbx.c,
- after the __ast_pbx_run loop). It's way too late to have any
- affect on the CDRs. The CDR is already posted and gone, and the
- remnants have been cleared. 2. I was very much afraid that moving
- the running of the 'h' extension down into the bridge code (where
- it would be now practical to do it), would result in a lot more
- calls to the 'h' exten, so I implemented it as another exten
- under another name, but found, to my pleasant surprise, that
- there was a 1:1 correspondence to the running of the 'h' exten in
- the pbx_run loop, and the new spot at the end of the bridge. So,
- I ifdef'd out the current 'h' loop, and moved it into the bridge
- code. The only difference I can see is the stuff about the
- AST_PBX_KEEPALIVE, and hopefully, if this is still an important
- decision point, I can replicate it if there are complaints. To be
- perfectly honest, the KEEPALIVE situation is not totally clear to
- me, and how it relates to a post-bridge situation is less clear.
- I suspect the users will point out everything in total clarity if
- this steps on anyone's toes! 3. I temporarily swap the bridge_cdr
- into the channel before running the 'h' exten, which makes it
- possible for users to edit the cdr before it goes out the door.
- And, of course, with the endbeforehexten config var set, the
- users can also get at the billsec/duration vals. After the h
- exten finishes, the cdr is swapped back and processing continues
- as normal. Please, all who deal with CDR's, please test this
- version of Asterisk, and file bug reports as appropriate!
-
-2008-08-21 10:11 +0000 [r139283] Philippe Sultan <philippe.sultan@gmail.com>
-
- * channels/chan_gtalk.c: Apply fix for issue #13310 to branch 1.4,
- too.
-
-2008-08-20 22:14 +0000 [r139213] Russell Bryant <russell@digium.com>
-
- * apps/app_chanspy.c: Fix a crash in the ChanSpy application. The
- issue here is that if you call ChanSpy and specify a spy group,
- and sit in the application long enough looping through the
- channel list, you will eventually run out of stack space and the
- application with exit with a seg fault. The backtrace was always
- inside of a harmless snprintf() call, so it was tricky to track
- down. However, it turned out that the call to snprintf() was just
- the biggest stack consumer in this code path, so it would always
- be the first one to hit the boundary. (closes issue #13338)
- Reported by: ruddy
-
-2008-08-20 19:52 +0000 [r139151] Shaun Ruffell <sruffell@digium.com>
-
- * codecs/codec_dahdi.c: Fix bug where the samples were not accurate
- when in G723 mode, which would cause the timestamp field of the
- RTP header to be invalid.
-
-2008-08-20 19:35 +0000 [r139145] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_dahdi.c, configure,
- include/asterisk/autoconfig.h.in, configure.ac: Backport support
- for Zaptel/DAHDI channel-level alarms from trunk/1.6, because not
- doing so just makes it difficult for people with channels that
- are in alarm when Asterisk starts up to get them going once the
- alarm is cleared (closes issue #12160) Reported by: tzafrir
- Patches: asterisk-chanalarms_14.patch uploaded by tzafrir
- (license 46) Tested by: tzafrir
-
-2008-08-20 17:14 +0000 [r139074] Steve Murphy <murf@digium.com>
-
- * main/cdr.c: (closes issue #13263) Reported by: brainy Tested by:
- murf The specialized reset routine is tromping on the flags field
- of the CDR. I made a change to not reset the DISABLED bit. This
- should get rid of this problem.
-
-2008-08-20 15:37 +0000 [r139015] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: sip_read should properly handle a NULL
- return from sip_rtp_read. (closes issue #13257) Reported by:
- travishein
-
-2008-08-19 23:22 +0000 [r138949] Jeff Peeler <jpeeler@digium.com>
-
- * include/asterisk/dahdi_compat.h: add DAHDI_POLICY_WHEN_FULL
- compatability define for Zaptel
-
-2008-08-19 23:17 +0000 [r138942] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_agent.c: Reset agent_pvt variables back to the
- values in agents.conf (from what the corresponding channel
- variables were set to) when the agent logs out. (closes issue
- #13098) Reported by: davidw Patches:
- 20080731__issue13098_agent_ackcall_not_reset.diff uploaded by
- bbryant (license 36) Tested by: davidw
-
-2008-08-19 22:56 +0000 [r138938] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c: Add configuration option to
- chan_dahdi.conf to allow buffering policy and number of buffers
- to be configured per channel. Syntax: buffers=<num of
- buffers>,<policy> Where the number of buffers is some
- non-negative integer and the policy is either "full", "half", or
- "immediate".
-
-2008-08-19 18:50 +0000 [r138685-138886] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_chanspy.c: Add a lock and unlock prior to the
- destruction of the chanspy_ds lock to ensure that no other
- threads still have it locked. While this should not happen under
- normal circumstances, it appears that if the spyer and spyee hang
- up at nearly the same time, the following may occur. 1.
- ast_channel_free is called on the spyee's channel. 2. The chanspy
- datastore is removed from the spyee's channel in
- ast_channel_free. 3. In the spyer's thread, the spyer attempts to
- remove and destroy the datastore from the spyee channel, but the
- datastore has already been removed in step 2, so the spyer
- continues in the code. 4. The spyee's thread continues and calls
- the datastore's destroy callback, chanspy_ds_destroy. This
- involves locking the chanspy_ds. 5. Now the spyer attempts to
- destroy the chanspy_ds lock. The problem is that in step 4, the
- spyee has locked this lock, meaning that the spyer is attempting
- to destroy a lock which is currently locked by another thread.
- The backtrace provided in issue #12969 supports the idea that
- this is possible (and has even occurred). This commit does not
- close the issue, but should help in preventing one type of crash
- associated with the use of app_chanspy.
-
- * apps/app_queue.c: Change the inequalities used in app_queue with
- regards to timeouts from being strict to non-strict for more
- accuracy. (closes issue #13239) Reported by: atis Patches:
- app_queue_timeouts_v2.patch uploaded by atis (license 242)
-
-2008-08-18 16:57 +0000 [r138663] Kevin P. Fleming <kpfleming@digium.com>
-
- * codecs/codec_dahdi.c: look for transcoder in proper place based
- on build against Zaptel or DAHDI
-
-2008-08-18 11:57 +0000 [r138569] Sean Bright <sean.bright@gmail.com>
-
- * channels/chan_dahdi.c: You know what's awesome? Code that
- compiles... ;)
-
-2008-08-18 02:05 +0000 [r138516] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c: fix compilation warnings
-
-2008-08-16 01:12 +0000 [r138309-138360] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c: fixes use count to properly decrement if
- an active dahdi channel is destroyed allowing module to be
- unloaded
-
- * channels/chan_dahdi.c: add forgotten locks around ss_thread_count
- in ss_thread for dahdi restart
-
-2008-08-15 22:33 +0000 [r138258] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_sip.c, configs/sip.conf.sample: More fixes for
- realtime peers. (closes issue #12921) Reported by: Nuitari
- Patches: 20080804__bug12921.diff.txt uploaded by Corydon76
- (license 14) 20080815__bug12921.diff.txt uploaded by Corydon76
- (license 14) Tested by: Corydon76
-
-2008-08-15 21:28 +0000 [r138119-138238] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c: initialize condition variable
- ss_thread_complete using ast_cond_init
-
- * channels/chan_dahdi.c: declared static mutexes using
- AST_MUTEX_DEFINE_STATIC macro
-
- * channels/chan_dahdi.c: Fixes the dahdi restart functionality.
- Dahdi restart allows one to restart all DAHDI channels, even if
- they are currently in use. This is different from unloading and
- then loading the module since unloading requires the use count to
- be zero. Reloading the module is different in that the signalling
- is not changed from what it was originally configured. Also, this
- fixes not closing all the file descriptors for D-channels upon
- module unload (which would prevent loading the module
- afterwards). (closes issue #11017)
-
-2008-08-15 15:07 +0000 [r138027] Russell Bryant <russell@digium.com>
-
- * main/autoservice.c: Ensure that when a hangup occurs in
- autoservice, that a hangup frame gets properly deferred to be
- read from the channel owner when it gets taken out of
- autoservice. (closes issue #12874) Reported by: dimas Patches:
- v1-12874.patch uploaded by dimas (license 88)
-
-2008-08-15 14:51 +0000 [r137847-138023] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_strings.c: Additional check for more string specifiers
- than arguments. (closes issue #13299) Reported by: adomjan
- Patches: 20080813__bug13299.diff.txt uploaded by Corydon76
- (license 14) func_strings.c-sprintf.patch uploaded by adomjan
- (license 487) Tested by: adomjan
-
- * channels/chan_dahdi.c: Oops, wrong direction
-
- * channels/chan_dahdi.c: When creating the secondary subchannel
- name, it is necessary to compare to the existing channel name
- without the "Zap/" or "DAHDI/" prefix, since our test string is
- also without that prefix. (closes issue #13027) Reported by:
- dferrer Patches: chan_zap-1.4.21.1_fix2.patch uploaded by dferrer
- (license 525) (Slightly modified by me, to compensate for both
- names)
-
-2008-08-14 14:05 +0000 [r137731] Russell Bryant <russell@digium.com>
-
- * configs/sip.conf.sample: Comments in this config file were
- aligned only if your tab size was set to 8. So, convert tabs to
- spaces so that things should be aligned regardless of what tab
- size you use in your editor.
-
-2008-08-14 02:03 +0000 [r137677-137679] Kevin P. Fleming <kpfleming@digium.com>
-
- * Zaptel-to-DAHDI.txt: forgot one module name that changed
-
- * include/asterisk/dahdi_compat.h, channels/chan_dahdi.c,
- build_tools/menuselect-deps.in, configure, configure.ac,
- codecs/codec_dahdi.c: add support for Zaptel versions that
- contain the new transcoder interface
-
-2008-08-13 21:35 +0000 [r137580] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c: Register DAHDISendKeypadFacility
- application if dahdi_chan_mode is set to DAHDI + Zap. Mark
- ZapSendKeypadFacility application as deprecated on usage.
-
-2008-08-13 20:46 +0000 [r137527-137530] Kevin P. Fleming <kpfleming@digium.com>
-
- * Zaptel-to-DAHDI.txt (added): add document describing what users
- will need to be aware of when upgrading to this version and using
- DAHDI
-
- * apps/app_meetme.c: remove some more chan_zap references
-
- * doc/asterisk-conf.txt, channels/chan_dahdi.c: document
- dahdichanname option in doc/asterisk-conf.txt make chan_dahdi
- read its configuration from zapata.conf if dahdichanname has been
- set to 'no'
-
-2008-08-13 14:33 +0000 [r137348-137405] Sean Bright <sean.bright@gmail.com>
-
- * doc/cdrdriver.txt: Update docs to reflect the change to cdr_tds
-
- * cdr/cdr_tds.c: Bring cdr_tds in line with the other CDR backends
- and have it try to store CDR(userfield) if it is set. The new
- behavior is to check for the userfield column on module load, and
- if it exists, we will store CDR(userfield) when CDRs are written.
- A similar patch already went into trunk and 1.6.0. (closes issue
- #13290) Reported by: falves11
-
-2008-08-11 13:33 +0000 [r137188] Kevin P. Fleming <kpfleming@digium.com>
-
- * apps/app_meetme.c: convert this module to be able to handle DAHDI
- or Zaptel (reported on asterisk-users, don't know how this got
- missed before)
-
-2008-08-11 00:20 +0000 [r137138] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_odbc.c: Deallocate database connection handle on
- disconnect, as we allocate another one on connect. (closes issue
- #13271) Reported by: dveiga
-
-2008-08-09 17:11 +0000 [r136999] Russell Bryant <russell@digium.com>
-
- * configure, configure.ac: Ensure PBX_DAHDI_TRANSCODE will evaluate
- to 0 if not found instead of empty. pointed out by tzafrir on
- #asterisk-dev
-
-2008-08-09 15:25 +0000 [r136946] Tilghman Lesher <tlesher@digium.com>
-
- * /, include/asterisk/compat.h, include/asterisk/astobj2.h: Merged
- revisions 136945 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r136945 | tilghman | 2008-08-09 10:24:36 -0500 (Sat, 09 Aug 2008)
- | 2 lines Regression fixes for Solaris ........
-
-2008-08-08 00:15 +0000 [r136726] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18,
- pbx/ael/ael-test/ref.ael-vtest13,
- pbx/ael/ael-test/ref.ael-ntest10, pbx/pbx_ael.c,
- include/asterisk/ael_structs.h: (closes issue #13236) Reported
- by: korihor Wow, this one was a challenge! I regrouped and ran a
- new strategy for setting the ~~MACRO~~ value; I set it once per
- extension, up near the top. It is only set if there is a switch
- in the extension. So, I had to put in a chunk of code to detect a
- switch in the pval tree. I moved the code to insert the set of
- ~~exten~~ up to the beginning of the gen_prios routine, instead
- of down in the switch code. I learned that I have to push the
- detection of the switches down into the code, so everywhere I
- create a new exten in gen_prios, I make sure to pass onto it the
- values of the mother_exten first, and the exten next. I had to
- add a couple fields to the exten struct to accomplish this, in
- the ael_structs.h file. The checked field makes it so we don't
- repeat the switch search if it's been done. I also updated the
- regressions.
-
-2008-08-07 18:25 +0000 [r136560] Kevin P. Fleming <kpfleming@digium.com>
-
- * build_tools/menuselect-deps.in, configure, configure.ac: change
- the required dependency for codec_dahdi to only be satisfied by
- DAHDI and not Zaptel, as the new transcoder interface is only in
- DAHDI
-
-2008-08-07 18:14 +0000 [r136544] Shaun Ruffell <sruffell@digium.com>
-
- * codecs/codec_dahdi.c: Updated codec_dahdi to use the new
- transcoder interface in the first DAHDI release. Codec dahdi no
- longer functions with the transcoder interface in zaptel at this
- time (which the last zaptel release was 1.4.11). NOTE: Still
- needs an update to the configure script to make sure that
- codec_dahdi is only built if the new transcoder interface is
- present in the drivers. (Issue: DAHDI-42)
-
-2008-08-07 16:50 +0000 [r136488] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_queue.c: Update persistent state on all exit conditions.
- (closes issue #12916) Reported by: sgenyuk Patches:
- app_queue.patch.txt uploaded by neutrino88 (license 297) Tested
- by: sgenyuk, aragon
-
-2008-08-07 16:30 +0000 [r136404-136484] Kevin P. Fleming <kpfleming@digium.com>
-
- * include/asterisk/doxyref.h: add a raw list of all libraries that
- any part of Asterisk links directly to
-
- * apps/app_voicemail.c: work around a bug in gcc-4.2.3 that
- incorrectly ignores the casting away of 'const' for pointers when
- the developer knows it is safe to do so
-
- * Makefile: remove config.cache during distclean, in case the user
- is using autoconf caching
-
-2008-08-07 01:31 +0000 [r136304-136348] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_dahdi.c: Also, parse
- useincomingcalleridonzaptransfer (and add appropriate deprecation
- warnings).
-
- * channels/chan_dahdi.c: For backwards compatibility with previous
- 1.4 versions which used "zapchan" in users.conf, ensure that we
- still support it.
-
-2008-08-06 21:18 +0000 [r136241] Richard Mudgett <rmudgett@digium.com>
-
- * channels/misdn_config.c, channels/chan_misdn.c,
- configs/misdn.conf.sample: * The allowed_bearers setting in
- misdn.conf misspelled one of its options: digital_restricted. *
- Fixed some other spelling errors and typos.
-
-2008-08-06 20:42 +0000 [r136238] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: We only need to unregister the QueueStatus
- manager command once on an unload
-
-2008-08-06 20:14 +0000 [r136190] Tilghman Lesher <tlesher@digium.com>
-
- * contrib/init.d/rc.redhat.asterisk: -C option takes a filename,
- not a directory path. (closes issue #13007) Reported by:
- klaus3000
-
-2008-08-06 18:58 +0000 [r136168] Russell Bryant <russell@digium.com>
-
- * Makefile: Remove the use of --no-print-directory when compiling
- subdirectories. This allows vim :make functionality to work
- properly when errors have occurred in the build. Without printing
- the directories, vim did not know how to find the file that the
- error occurred in. If the extra bit of build noise annoys anyone,
- just let me know, and I'll make this optional.
-
-2008-08-06 15:58 +0000 [r136062] Mark Michelson <mmichelson@digium.com>
-
- * main/rtp.c, channels/chan_skinny.c: Since adding the
- AST_CONTROL_SRCUPDATE frame type, there are places where
- ast_rtp_new_source may be called where the tech_pvt of a channel
- may not yet have an rtp structure allocated. This caused a crash
- in chan_skinny, which was fixed earlier, but now the same crash
- has been reported against chan_h323 as well. It seems that the
- best solution is to modify ast_rtp_new_source to not attempt to
- set the marker bit if the rtp structure passed in is NULL. This
- change to ast_rtp_new_source also allows the removal of what is
- now a redundant pointer check from chan_skinny. (closes issue
- #13247) Reported by: pj
-
-2008-08-06 03:53 +0000 [r135899-135949] Tilghman Lesher <tlesher@digium.com>
-
- * main/channel.c: Fix a longstanding bug in channel walking logic,
- and fix the explanation to make sense. (Closes issue #13124)
-
- * main/translate.c: Since powerof() can return an error condition,
- it's foolhardy not to detect and deal with that condition.
- (Related to issue #13240)
-
- * include/asterisk/threadstorage.h, include/asterisk/utils.h: 1)
- Bugfix for debugging code 2) Reduce compiler warnings for another
- section of debugging code (Closes issue #13237)
-
-2008-08-06 00:29 +0000 [r135841-135850] Mark Michelson <mmichelson@digium.com>
-
- * /: Remove properties that should not be here
-
- * apps/app_skel.c: Revert inadvertent changes to app_skel that
- occurred when I was testing for a memory leak
-
- * include/asterisk/abstract_jb.h, main/channel.c, /,
- apps/app_skel.c, main/abstract_jb.c, main/fixedjitterbuf.h:
- Merging the issue11259 branch. The purpose of this branch was to
- take into account "burps" which could cause jitterbuffers to
- misbehave. One such example is if the L option to Dial() were
- used to inject audio into a bridged conversation at regular
- intervals. Since the audio here was not passed through the
- jitterbuffer, it would cause a gap in the jitterbuffer's
- timestamps which would cause a frames to be dropped for a brief
- period. Now ast_generic_bridge will empty and reset the
- jitterbuffer each time it is called. This causes injected audio
- to be handled properly. ast_generic_bridge also will empty and
- reset the jitterbuffer if it receives an AST_CONTROL_SRCUPDATE
- frame since the change in audio source could negatively affect
- the jitterbuffer. All of this was made possible by adding a new
- public API call to the abstract_jb called ast_jb_empty_and_reset.
- (closes issue #11259) Reported by: plack Tested by: putnopvut
-
-2008-08-05 23:13 +0000 [r135799] Steve Murphy <murf@digium.com>
-
- * apps/app_dial.c, main/cdr.c, main/channel.c, res/res_features.c,
- include/asterisk/cdr.h: (closes issue #12982) Reported by: bcnit
- Tested by: murf I discovered that also, in the previous bug fixes
- and changes, the cdr.conf 'unanswered' option is not being
- obeyed, so I fixed this. And, yes, there are two 'answer' times
- involved in this scenario, and I would agree with you, that the
- first answer time is the time that should appear in the CDR. (the
- second 'answer' time is the time that the bridge was begun). I
- made the necessary adjustments, recording the first answer time
- into the peer cdr, and then using that to override the bridge
- cdr's value. To get the 'unanswered' CDRs to appear, I purposely
- output them, using the dial cmd to mark them as DIALED (with a
- new flag), and outputting them if they bear that flag, and you
- are in the right mode. I also corrected one small mention of the
- Zap device to equally consider the dahdi device. I heavily tested
- 10-sec-wait macros in dial, and without the macro call; I tested
- hangups while the macro was running vs. letting the macro
- complete and the bridge form. Looks OK. Removed all the
- instrumentation and debug.
-
-2008-08-05 21:34 +0000 [r135747] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_iax2.c: In a conversion to use ast_strlen_zero, the
- meaning of the flag IAX_HASCALLERID was perverted. This change
- reverts IAX2 to the original meaning, which was, that the
- callerid set on the client should be overridden on the server,
- even if that means the resulting callerid is blank. In other
- words, if you set "callerid=" in the IAX config, then the
- callerid should be overridden to blank, even if set on the
- client. Note that there's a distinction, even on realtime,
- between the field not existing (NULL in databases) and the field
- existing, but set to blank (override callerid to blank).
-
-2008-08-05 13:25 +0000 [r135597] Sean Bright <sean.bright@gmail.com>
-
- * main/cli.c: Use PATH_MAX for filenames
-
-2008-08-04 20:15 +0000 [r135536] Russell Bryant <russell@digium.com>
-
- * configs/chan_dahdi.conf.sample: fix a config sample typo
-
-2008-08-04 17:07 +0000 [r135479-135482] Tilghman Lesher <tlesher@digium.com>
-
- * contrib/init.d/rc.mandrake.asterisk: Define ASTSBINDIR for script
-
- * apps/app_voicemail.c: Memory leak on unload (closes issue #13231)
- Reported by: eliel Patches: app_voicemail.leak.patch uploaded by
- eliel (license 64)
-
-2008-08-04 16:26 +0000 [r135473] Russell Bryant <russell@digium.com>
-
- * configs/chan_dahdi.conf.sample: Add a minor clarification to the
- documentation of mohinterpret and mohsuggest
-
-2008-08-01 11:43 +0000 [r135055-135058] Michiel van Baak <michiel@vanbaak.info>
-
- * apps/app_ices.c: make app_ices compile on OpenBSD.
-
- * channels/chan_skinny.c: fix some potential deadlocks in
- chan_skinny (closes issue #13215) Reported by: qwell Patches:
- 2008080100_bug13215.diff.txt uploaded by mvanbaak (license 7)
- Tested by: mvanbaak
-
-2008-07-31 22:18 +0000 [r134983] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/http.c: accomodate users who seem to lack a sense of humor
- :-)
-
-2008-07-31 21:53 +0000 [r134976] Tilghman Lesher <tlesher@digium.com>
-
- * sample.call, main/manager.c, pbx/pbx_spool.c: Specify codecs in
- callfiles and manager, to allow video calls to be set up from
- callfiles and AMI. (closes issue #9531) Reported by: Geisj
- Patches: 20080715__bug9531__1.4.diff.txt uploaded by Corydon76
- (license 14) 20080715__bug9531__1.6.0.diff.txt uploaded by
- Corydon76 (license 14) Tested by: Corydon76
-
-2008-07-31 19:37 +0000 [r134915] Russell Bryant <russell@digium.com>
-
- * apps/app_ices.c: Get app_ices working again (closes issue #12981)
- Reported by: dlogan Patches:
- 20080709__app_ices_v2_update_trunk.diff uploaded by bbryant
- (license 36) 20080709__app_ices_v2_update_14.diff uploaded by
- bbryant (license 36) Tested by: bbryant
-
-2008-07-31 19:23 +0000 [r134883] Steve Murphy <murf@digium.com>
-
- * res/res_features.c: (closes issue #11849) Reported by: greyvoip
- Tested by: murf OK, a few days of debugging, a bunch of
- instrumentation in chan_sip, main/channel.c, main/pbx.c, etc. and
- 5 solid notebook pages of notes later, I have made the small
- tweek necc. to get the start time right on the second CDR when: A
- Calls B B answ. A hits Xfer button on sip phone, A dials C and
- hits the OK button, A hangs up C answers ringing phone B and C
- converse B and/or C hangs up But does not harm the scenario
- where: A Calls B B answ. B hits xfer button on sip phone, B dials
- C and hits the OK button, B hangs up C answers ringing phone A
- and C converse A and/or C hangs up The difference in start times
- on the second CDR is because of a Masquerade on the B channel
- when the xfer number is sent. It ends up replacing the CDR on the
- B channel with a duplicate, which ends up getting tossed out. We
- keep a pointer to the first CDR, and update *that* after the
- bridge closes. But, only if the CDR has changed. I hope this
- change is specific enough not to muck up any current CDR-based
- apps. In my defence, I assert that the previous information was
- wrong, and this change fixes it, and possibly other similar
- scenarios. I wonder if I should be doing the same thing for the
- channel, as I did for the peer, but I can't think of a scenario
- this might affect. I leave it, then, as an exersize for the
- users, to find the scenario where the chan's CDR changes and
- loses the proper start time.
-
-2008-07-31 16:45 +0000 [r134814] Russell Bryant <russell@digium.com>
-
- * channels/iax2-parser.c: In case we have some processing threads
- that free more frames than they allocate, do not let the frame
- cache grow forever. (closes issue #13160) Reported by: tavius
- Tested by: tavius, russell
-
-2008-07-31 15:56 +0000 [r134758] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Add more timeout checks into app_queue,
- specifically targeting areas where an unknown and potentially
- long time has just elapsed. Also added a check to try_calling()
- to return early if the timeout has elapsed instead of potentially
- setting a negative timeout for the call (thus making it have *no*
- timeout at all). (closes issue #13186) Reported by:
- miquel_cabrespina Patches: 13186.diff uploaded by putnopvut
- (license 60) Tested by: miquel_cabrespina
-
-2008-07-30 22:39 +0000 [r134704] Tilghman Lesher <tlesher@digium.com>
-
- * main/sched.c, include/asterisk/sched.h: Oops, wrong define
-
-2008-07-30 22:02 +0000 [r134652] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_ael.c: (closes issue #13197) Reported by: pj (closes
- issue #13051) Reported by: pj This patch substitutes commas in
- the expr supplied to the if () statement, as in if ( expr ) ...
- This solves both the bugs above, and makes the source symmetric
- with switch statements, which were earlier reported to need this
- sort of treatment. I tested this using the examples, both for the
- compiler and at run time. Looks good.
-
-2008-07-30 21:38 +0000 [r134649] Tilghman Lesher <tlesher@digium.com>
-
- * configure, configure.ac: Qwell pointed out, via IRC, that the
- previous fix only worked when explicitly set. When nothing is
- set, and the option is implied, it breaks, because configure sets
- the prefix to 'NONE'. Fixing.
-
-2008-07-30 20:37 +0000 [r134540-134595] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_dundi.c: Reduce stack consumption by 12.5% of the max
- stack size to fix a crash when compiled with LOW_MEMORY. (closes
- issue #13154) Reported by: edantie
-
- * funcs/func_curl.c: Fix a memory leak in func_curl. Every thread
- that used this function leaked an allocation the size of a
- pointer. (reported by jmls in #asterisk-dev)
-
-2008-07-30 19:47 +0000 [r134480-134536] Tilghman Lesher <tlesher@digium.com>
-
- * configure, configure.ac: Only override sysconfdir and mandir when
- prefix=/usr (closes issue #13093) Reported by: pabelanger
-
- * res/res_agi.c: launch_netscript sometimes returns -1, which fails
- to set AGISTATUS. Map failure to -1, so that AGISTATUS is always
- set. (closes issue #13199) Reported by: smw1218
-
-2008-07-30 18:31 +0000 [r134475] Mark Michelson <mmichelson@digium.com>
-
- * main/app.c: Fix a spot where a function could return without
- bringing a channel out of autoservice.
-
-2008-07-30 15:29 +0000 [r134254-134352] Kevin P. Fleming <kpfleming@digium.com>
-
- * Makefile: use the proper method for building version.h
-
- * include/asterisk/dahdi_compat.h, apps/app_dahdibarge.c,
- channels/chan_dahdi.c, apps/app_meetme.c, apps/app_flash.c,
- apps/app_dahdiscan.c, apps/app_dahdiras.c, codecs/codec_dahdi.c:
- build against the now-typedef-free dahdi/user.h
-
-2008-07-29 15:54 +0000 [r134223] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Merging the imap_consistency branch. The
- main aim of this branch was to make the IMAP code function in the
- same manner as the ODBC code does, eliminating the need for so
- many IMAP-specific code chunks. The focal point of all of this
- work was to make the various macros (e.g. RETRIEVE, DISPOSE)
- functionally equivalent. While doing the above work, I also fixed
- a few bugs that I came across in my testing. Among these were 1.
- Fixed message forwarding. This was completely broken when using
- IMAP. 2. Fixed the inability to save new messages as old and vice
- versa. 3. Fixed the "delete" options in voicemail.conf when using
- IMAP storage. Even though a few bugs were fixed and the code is a
- lot more consistent, the one thing that was *not* improved in
- this branch was performance. The merge of this to trunk may not
- come immediately due to the amount of work it will probably
- involve. (closes issue #12764) Reported by: balsamcn
-
-2008-07-28 21:50 +0000 [r134161] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: Detect when sox fails to raise the volume,
- because sox can't read the file. (closes issue #12939) Reported
- by: rickbradley Patches: 20080728__bug12939.diff.txt uploaded by
- Corydon76 (license 14) Tested by: rickbradley
-
-2008-07-26 15:31 +0000 [r133980] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c, include/asterisk/doxyref.h: Add the licensing
- section to the docs in 1.4, as well, so that we can work on
- having an accurate list for each version of Asterisk that is
- supported
-
-2008-07-25 18:00 +0000 [r133649-133709] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: Remove unnecessary mmap flag (Closes issue
- #13161)
-
- * main/channel.c, channels/chan_agent.c, main/devicestate.c: Fix
- some errant device states by making the devicestate API more
- strict in terms of the device argument (only without the unique
- identifier appended). (closes issue #12771) Reported by: davidw
- Patches: 20080717__bug12771.diff.txt uploaded by Corydon76
- (license 14) Tested by: davidw, jvandal, murf
-
-2008-07-25 15:00 +0000 [r133578] Russell Bryant <russell@digium.com>
-
- * /, LICENSE: Merged revisions 133577 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r133577 | russell | 2008-07-25 10:00:13 -0500 (Fri, 25 Jul 2008)
- | 2 lines Fix the IAX2 URI for calling Digium ........
-
-2008-07-25 14:40 +0000 [r133572] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: We need to make sure to null-terminate the
- "name" portion of SIP URI parameters so that there are no bogus
- comparisons. Thanks to bbryant for pointing this out.
-
-2008-07-24 21:17 +0000 [r133361-133488] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_sip.c: Fix rtautoclear and rtcachefriends (Closes
- issue #12707)
-
- * /: Blocked revisions 133360 via svnmerge ........ r133360 |
- tilghman | 2008-07-23 22:46:01 -0500 (Wed, 23 Jul 2008) | 2 lines
- This part was not correctly patched for AST-2008-010. ........
-
-2008-07-23 21:49 +0000 [r133295] Jason Parker <jparker@digium.com>
-
- * channels/chan_dahdi.c: inbandrelease is gone - it's now
- inbanddisconnect
-
-2008-07-23 21:05 +0000 [r133226-133237] Kevin P. Fleming <kpfleming@digium.com>
-
- * include/asterisk/stringfields.h, main/utils.c: revert an
- optimization that broke ABI... thanks russell!
-
- * apps/app_chanspy.c, include/asterisk/options.h, main/asterisk.c,
- apps/app_dahdibarge.c, channels/chan_dahdi.c,
- apps/app_dahdiras.c: make some more changes to the dahdi/zap
- channel name support stuff to ensure allthe globals are 'const',
- and clean up mmichelson's changes to app_chanspy to simplify the
- code
-
-2008-07-23 19:39 +0000 [r132974-133169] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_chanspy.c, include/asterisk/options.h, main/asterisk.c,
- channels/chan_dahdi.c: As suggested by seanbright, the
- PSEUDO_CHAN_LEN in app_chanspy should be set at load time, not at
- compile time, since dahdi_chan_name is determined at load time.
- Also changed the next_unique_id_to_use to have the static
- qualifier. Also added the dahdi_chan_name_len variable so that
- strlen(dahdi_chan_name) isn't necessary. Thanks to seanbright for
- the suggestion.
-
- * apps/app_chanspy.c: Zap/pseudo is ten characters, but
- DAHDI/pseudo is twelve. The strncmp call in next_channel should
- account for this.
-
- * apps/app_chanspy.c: Update the "last" channel in next_channel in
- app_chanspy so that the same pseudo channel isn't constantly
- returned. related to issue #13124
-
- * channels/chan_dahdi.c: Small cleanup. Move the declaration of the
- DAHDI_SPANINFO variable to the block where it is used. This
- allows one less #ifdef HAVE_PRI to clutter things up. Thanks to
- Tzafrir for pointing this out on #asterisk-dev
-
- * channels/chan_dahdi.c: Fix building of chan_dahdi when HAVE_PRI
- is not defined.
-
-2008-07-23 15:52 +0000 [r132872-132942] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_dahdi.c: ensure that after a channel is created, if
- it happened to be in 'channel alarm' state, when that alarm
- clears we won't generate a spurious 'alarm cleared' message
- (closes issue #12160) Reported by: tzafrir
-
- * include/asterisk/stringfields.h, main/utils.c: minor optimization
- for stringfields: when a field is being set to a larger value
- than it currently contains and it happens to be the most recent
- field allocated from the currentl pool, it is possible to 'grow'
- it without having to waste the space it is currently using (or
- potentially even allocate a new pool)
-
-2008-07-23 11:37 +0000 [r132826] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c: another Fix because of r119585, this
- commit has broken high frequented BRI Ports, there was a
- possibility that a channel, that was marked as in_use would be
- reused later, the corresponding port could got stuck then. So it
- is recommended to upgrade for chan_misdn users.
-
-2008-07-22 22:14 +0000 [r132790] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: Allow Spiraled INVITEs to work correctly
- within Asterisk. Prior to this change, a spiraled INVITE would
- cause a 482 Loop Detected to be sent to the caller. With this
- change, if a potential loop is detected, the Request-URI is
- inspected to see if it has changed from what was originally
- received. If pedantic mode is on, then this inspection is fully
- RFC 3261 compliant. If pedantic mode is not on, then a string
- comparison is used to test the equality of the two R-URIs. This
- has been tested by using OpenSER to rewrite the R-URI and send
- the INVITE back to Asterisk. (closes issue #7403) Reported by:
- stephen_dredge
-
-2008-07-22 22:11 +0000 [r132784-132787] Kevin P. Fleming <kpfleming@digium.com>
-
- * include/asterisk/options.h, main/asterisk.c,
- apps/app_dahdibarge.c, channels/chan_dahdi.c, apps/app_flash.c,
- apps/app_dahdiras.c: fix up namespace pollution for
- dahdi_chan_mode enum correct registration of AMI actions in
- chan_dahdi; in zap-only mode, only register the Zap flavors of
- the actions (and use Zap prefixes for headers and acks), but in
- dahdi+zap mode, register both Zap and DAHDI flavors of actions
-
- * Makefile.rules: add rules to create preprocessor output... useful
- for debugging macros
-
-2008-07-22 21:19 +0000 [r132713] Tilghman Lesher <tlesher@digium.com>
-
- * configs/iax.conf.sample, /, channels/chan_iax2.c: Merged
- revisions 132711 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r132711 | tilghman | 2008-07-22 16:14:10 -0500 (Tue, 22 Jul 2008)
- | 2 lines Fixes for AST-2008-010 and AST-2008-011 ........
-
-2008-07-22 21:17 +0000 [r132704-132712] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_dahdi.c: ensure that if any alarms exist at channel
- creation time, they are handled identically to if they occurred
- later, so that later alarm clearing will work properly and 'make
- sense' (closes issue #12160) Reported by: tzafrir
-
- * configure, configure.ac, acinclude.m4: make AST_C_COMPILE_CHECK
- able to print a 'pretty' description of what it is doing
-
-2008-07-22 20:10 +0000 [r132645] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c, doc/sip-retransmit.txt (added): The most
- common question on the #asterisk iRC channel and on mailing lists
- seems to be in regards to an error message when retransmit fails.
- This is frequently misunderstood as a failure of Asterisk, not a
- failure of the network to reach the other party. This document
- tries to assist the Asterisk user in sorting out these issues by
- explaining the logic and pointing at some possible causes.
- Hopefully, we will get other questions now :-)
-
-2008-07-22 19:57 +0000 [r132571-132642] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_dahdi.c: correct wording in comment
-
- * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, configure,
- include/asterisk/autoconfig.h.in, configure.ac: use renamed
- libpri API call for controlling this feature (was improperly
- named before)
-
- * channels/chan_dahdi.c: teach chan_dahdi how to find the D-channel
- on BRI spans, and don't attempt to use channel 24 as a D-channel
- on spans of unexpected sizes
-
-2008-07-21 20:51 +0000 [r132506-132507] Brett Bryant <bbryant@digium.com>
-
- * apps/app_sendtext.c: Fix a bug where SENDTEXTSTATUS isn't set
- properly when it isn't supported on a channel (yet _another_
- useful patch by eliel). (issue #13081) Reported by: eliel
- Patches: app_sendtext1.4.c uploaded by eliel (license 64) Tested
- by: eliel
-
- * channels/chan_iax2.c: Fix a bug in 1.4 branch with iax2 channels
- not being removed when a call was rejected (from the calling box,
- not the box that denied the registration). Related to revisions
- 132466 in trunk, and 132467 in 1.6.0. Earlier I had accidently
- tested 1.4 with a backport from those revisions, so I didn't see
- this problem (oops).
-
-2008-07-19 16:45 +0000 [r132311] Kevin P. Fleming <kpfleming@digium.com>
-
- * LICENSE: grant a license exception to allow distribution of
- Asterisk binaries that use the UW IMAP Toolkit (which is licensed
- under a non-GPL-compatible license)
-
-2008-07-18 19:06 +0000 [r131970-132112] Tilghman Lesher <tlesher@digium.com>
-
- * main/say.c: Fix for Taiwanese number syntax (closes issue #12319)
- Reported by: CharlesWang Patches: saynumber-tw-1.4.18.1.patch
- uploaded by CharlesWang (license 444)
-
- * main/config.c: Textual clarification (closes issue #13106)
- Reported by: flefoll Patches:
- config.c.br14.120173.patch-unknown-directive uploaded by flefoll
- (license 244)
-
- * include/asterisk/sched.h, channels/chan_iax2.c: Spinlock within
- the destroy, to allow a scheduled job to continue, if it's
- waiting on the mutex which the destroy thread has.
-
- * main/sched.c: Oops
-
- * main/sched.c, include/asterisk/sched.h: Preserve ABI
- compatibility with last change
-
- * main/sched.c, include/asterisk/sched.h, channels/chan_iax2.c:
- Make the ast_assert call within ast_sched_del report something
- useful.
-
-2008-07-18 16:15 +0000 [r131921] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/dlfcn.c (removed), main/loader.c, main/Makefile,
- include/asterisk/dlfcn-compat.h (removed): remove the dlfcn
- compatibility stuff, because no platforms that Asterisk currently
- runs on it use it, and it doesn't build anyway
-
-2008-07-18 15:34 +0000 [r131915] Brett Bryant <bbryant@digium.com>
-
- * res/res_features.c: Fix a bug in blind transfers where the
- BLINDTRANSFER variable isn't always set to the other end of the
- blind transfer. (closes issue #12586)
-
-2008-07-17 20:35 +0000 [r131790] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_dahdi.c: Revert part of issue #5620 (revision 6965)
- as it appears that it was in error. This should fix talk call
- progress on analog lines. (closes issue #12178) Reported by:
- michael-fig Patches: 20080717__bug12178.diff.txt uploaded by
- Corydon76 (license 14)
-
-2008-07-16 22:17 +0000 [r131491] Brett Bryant <bbryant@digium.com>
-
- * channels/chan_iax2.c: Fix a bug in iax2 registration that allowed
- peers to register with case-insensitive names (user_cmp_cb and
- peer_cmp_cb are now both case-sensitive). (closes issue #13091)
-
-2008-07-16 21:46 +0000 [r131480] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_iax2.c: Apparently, in certain cases, a callno is
- already destroyed when iax2_destroy is called.
-
-2008-07-16 20:47 +0000 [r131421] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Always ensure that the channel's tech_pvt
- reference is NULL after calling the destroy callback. (closes
- issue #13060) Reported by: jpgrayson Patches:
- chan_iax2_tech_pvt_crash.patch uploaded by jpgrayson (license
- 492)
-
-2008-07-16 20:23 +0000 [r131299-131369] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Move the init_queue call back to where it used
- to be (changed Sept 12 last year). It was moved then to prevent a
- memory leak. Since then, the same memory leak recurred and was
- fixed in a better way. Now it has been found that the placement
- of this init_queue call can cause problems if a realtime queue
- has values changed to an empty string. The problem is that the
- default value for that queue parameter would not be set. (closes
- issue #13084) Reported by: elbriga
-
- * apps/app_queue.c: Apparently, "thread safety" is important,
- whatever that means. :P (Thanks Russell!)
-
- * apps/app_queue.c: Make absolutely certain that the transfer
- datastore is removed from the calling channel once the caller is
- finished in the queue. This could have weird con- sequences when
- dialing local queue members when multiple transfers occur on a
- single call. Also fixed a memory leak that would occur when an
- attended transfer occurred from a queue member. (closes issue
- #13047) Reported by: festr
-
-2008-07-16 17:53 +0000 [r131242] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_ael.c: (closes issue #13090) Reported by: murf The
- problem was that, esoteric as it is, because the hangerupper
- context immediately preceded the std-priv-extent macro, that the
- checking code accidentally would fall from traversing hangerupper
- into the std-priv-exten macro, where it would hit the hangerupper
- in the 'includes', and proceed into an infinite recursion. A
- small fix to traverse into the statements of the context instead
- of the context solves this issue. I also added some commented out
- printfs for debug, which were pretty handy in the face of a dorky
- gdb. This was a problem around since the package was first
- written; but evidently pretty rare in turning up in the field.
-
-2008-07-15 17:47 +0000 [r131012] Michiel van Baak <michiel@vanbaak.info>
-
- * main/cdr.c: remove 4 lines of redundant code. (closes issue
- #13080) Reported by: gknispel_proformatique Patches:
- trunk_ast_cdr_setapp.patch uploaded by gknispel (license 261)
-
-2008-07-15 17:19 +0000 [r130889-130959] Tilghman Lesher <tlesher@digium.com>
-
- * main/manager.c, channels/chan_sip.c: astman_send_error does not
- need a newline appended -- the API takes care of that for us.
- (closes issue #13068) Reported by: gknispel_proformatique
- Patches: asterisk_1_4_astman_send.patch uploaded by gknispel
- (license 261) asterisk_trunk_astman_send.patch uploaded by
- gknispel (license 261)
-
- * channels/chan_iax2.c: Override the callerid in all cases when the
- callerid is set in the user, not just when a remote callerid is
- set. Also, if not set in the user, allow the remote CallerID to
- pass through. (closes issue #12875) Reported by: dimas Patches:
- 20080714__bug12875.diff.txt uploaded by Corydon76 (license 14)
-
-2008-07-14 17:50 +0000 [r130792] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_dial.c: Add a check to the CAN_EARLY_BRIDGE macro in
- app_dial to be sure there are no audiohooks present on the
- channels involved. This fixed a one-way audio situation I had in
- my test setup. I couldn't find any open issues that suggested
- one-way audio with regards to mixmonitor (or other audiohook)
- usage, though.
-
-2008-07-14 17:10 +0000 [r130735] Michiel van Baak <michiel@vanbaak.info>
-
- * main/dnsmgr.c: notify the user that dnsmgr refresh wont work when
- dnsmgr is not enabled. Previously this command would
- automagically appear and disappear. This was confusing. (closes
- issue #12796) Reported by: chappell Patches:
- dnsmgr_refresh_3.diff uploaded by chappell (license 8) Tested by:
- russell, chappell, mvanbaak
-
-2008-07-14 10:38 +0000 [r130634] Russell Bryant <russell@digium.com>
-
- * main/audiohook.c: Bump up the debug level for a message.
-
-2008-07-13 22:48 +0000 [r130573] Michiel van Baak <michiel@vanbaak.info>
-
- * main/manager.c: fix memory leak when originate from manager
- cannot create a thread (closes issue #13069) Reported by:
- gknispel_proformatique Patches:
- asterisk_trunk_action_originate.patch uploaded by gknispel
- (license 261) Tested by: gknispel_proformatique, mvanbaak
-
-2008-07-13 17:56 +0000 [r130514] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_iax2.c: Reverting 2 changesets, as it breaks
- incoming IAX2 calls (Related to issue #12963) Reported by:
- mvanbaak
-
-2008-07-12 10:25 +0000 [r130373] Michiel van Baak <michiel@vanbaak.info>
-
- * pbx/pbx_ael.c: in 1.4 the functions still have | as argument
- seperator. This commit fixes the use of RAND in the ael random
- function. (closes issue #13061) Reported by: danpwi
-
-2008-07-11 22:23 +0000 [r130298-130317] Kevin P. Fleming <kpfleming@digium.com>
-
- * Makefile: forcibly remove the modules that are changing names
-
- * include/asterisk/options.h, main/asterisk.c, cdr/cdr_csv.c,
- Makefile, main/channel.c, apps/app_dahdibarge.c,
- channels/chan_dahdi.c, doc/hardware.txt, apps/app_flash.c,
- apps/app_dahdiras.c, main/file.c,
- contrib/utils/zones2indications.c, include/asterisk/channel.h,
- channels/chan_iax2.c: a whole pile of Zaptel/DAHDI compatibility
- work, with lots more to come... this tree is not yet ready for
- users to be easily upgrading or switching, but it needs to be :-)
-
-2008-07-11 20:03 +0000 [r130173-130236] Mark Michelson <mmichelson@digium.com>
-
- * main/audiohook.c: Remove redundant logic
-
- * main/audiohook.c: Fix a typo in audiohook_read_frame_both. While
- this change has not been proven to fix any specific issue, it is
- incorrect and could cause unforeseen problems.
-
-2008-07-11 18:51 +0000 [r130102-130169] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_iax2.c: Ensure that a destination callno of 0 will
- not match for frames that do not start a dialog (new, lagrq, and
- ping). (closes issue #12963) Reported by: russellb Patches:
- chan_iax2_dup_new_fix4.patch uploaded by jpgrayson (license 492)
-
- * channels/chan_agent.c: Pass the devicestate from an underlying
- channel up through the Agent channel. This should make the Agent
- always report the correct device state, even when the underlying
- channel is used for other purposes. (closes issue #12773)
- Reported by: davidw Patches: 20080710__bug12773.diff.txt uploaded
- by Corydon76 (license 14) Tested by: davidw
-
-2008-07-11 16:08 +0000 [r130039-130042] Kevin P. Fleming <kpfleming@digium.com>
-
- * doc/configuration.txt, configs/extensions.conf.sample,
- configs/sla.conf.sample, configs/zapata.conf.sample (removed),
- contrib/scripts/autosupport, README,
- configs/chan_dahdi.conf.sample (added), channels/chan_dahdi.c,
- include/asterisk/doxyref.h, doc/sla.tex, doc/ael.txt,
- configs/extensions.ael.sample, configs/smdi.conf.sample: new
- installations should be using DAHDI instead of Zaptel, so the
- sample config file is now chan_dahdi.conf instead of zapata.conf
- also, convert remaining references to zapata.conf in various
- places
-
- * configs/zapata.conf.sample, channels/chan_dahdi.c, configure,
- include/asterisk/autoconfig.h.in, configure.ac: add support for a
- configuration parameter for 'inband audio during RELEASE', which
- is currently mandatory in libpri-1.4.4 but will become
- configurable in libpri-1.4.5 later today (related to issue
- #13042)
-
-2008-07-11 14:18 +0000 [r129970] Russell Bryant <russell@digium.com>
-
- * include/asterisk/astobj.h: add a simple ASTOBJ_TRYWRLOCK macro
- ...
-
-2008-07-11 14:14 +0000 [r129907-129967] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/astmm.c: simplify calculation
-
- * main/astmm.c: fix a flaw found while experimenting with structure
- alignment and padding; low-fence checking would not work properly
- on 64-bit platforms, because the compiler was putting 4 bytes of
- padding between the fence field and the allocation memory block
- added a very obvious runtime warning if this condition reoccurs,
- so the developer who broke it can be chastised into fixing it :-)
-
- * sounds/Makefile: don't attempt to set user/group ownership of
- extracted sound files (reported on asterisk-users) (closes issue
- #13059)
-
-2008-07-10 21:57 +0000 [r129741-129803] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_iax2.c: Correctly deal with duplicate NEW frames
- (due to retransmission). Also, fixup the destination call number
- matching to be more strict and reliable. (closes issue #12963)
- Reported by: jpgrayson Patches: chan_iax2_dup_new_fix3.patch
- uploaded by jpgrayson (license 492) Tested by: jpgrayson,
- Corydon76
-
- * res/res_config_odbc.c: Oops
-
-2008-07-10 16:03 +0000 [r129567] Russell Bryant <russell@digium.com>
-
- * sample.call: Note that pbx_spool.so is the module used for call
- files (inspired by a question in #asterisk)
-
-2008-07-10 13:57 +0000 [r129505] Sean Bright <sean.bright@gmail.com>
-
- * main/editline: Update svn:ignore
-
-2008-07-09 19:32 +0000 [r129436] Mark Michelson <mmichelson@digium.com>
-
- * main/rtp.c: Fix a problem where inbound rfc2833 audio would be
- sent to the core instead of being P2P bridged. When the core
- regenerated the rfc2833 packet for the outbound leg, the SSRC
- would be different than the RTP audio on the call leg causing
- DTMF detection issues on the far end. (closes issue #12955)
- Reported by: tonyredstone Patches: dynamic_rtp.patch uploaded by
- tsearle (license 373) Tested by: tonyredstone
-
-2008-07-09 13:41 +0000 [r129343] Sean Bright <sean.bright@gmail.com>
-
- * main/editline/makelist (removed), main/editline/makelist.in
- (added), main/editline/configure, main/editline/Makefile.in,
- main/editline/configure.in: Look for the system installed awk
- instead of assuming it's at /usr/bin/awk. Pointed out by jmls via
- #asterisk-dev.
-
-2008-07-08 21:31 +0000 [r129158-129208] Mark Michelson <mmichelson@digium.com>
-
- * doc/imapstorage.txt: Update documentation to have the correct
- option name
-
- * apps/app_voicemail.c, doc/imapstorage.txt: Backport TCP-related
- timeouts to IMAP voicemail in 1.4 since it should solve bugs
- people are experiencing. Specifically, there are times where
- communication with the IMAP server causes system calls to block
- forever. If this should happen when querying the mailbox so that
- chan_sip's do_monitor thread can send MWI to a phone, it means
- that SIP calls cannot be processed any more. The timeout options
- are outlined in doc/imapstorage.txt. Defaults for the timeouts
- are sixty seconds. (closes issue #12987) Reported by: mthomasslo
-
-2008-07-08 20:27 +0000 [r129047-129149] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_dial.c, channels/chan_sip.c, include/asterisk/causes.h:
- Cause SIP to return a 480 instead of a 404 when a sip peer
- exists, but is not registered. (closes issue #12885) Reported by:
- ibc Patches: 20080701__bug12885__2.diff.txt uploaded by Corydon76
- (license 14) Tested by: ibc
-
- * channels/chan_iax2.c: Timestamp decoding for video mini-frames is
- bogus, because the timestamp only includes 15 bits, unlike voice
- frames, which contain a 16-bit timestamp. (closes issue #13013)
- Reported by: jpgrayson Patches: chan_iax2_unwrap_ts.patch
- uploaded by jpgrayson (license 492)
-
-2008-07-08 09:52 +0000 [r128912-128950] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Don't hangup the call if we can't resolve
- the Contact if there's a proxy route set for the call. ---- This
- comment was added a while ago and today it hit me badly. /* OEJ:
- Possible issue that may need a check: If we have a proxy route
- between us and the device, should we care about resolving the
- contact or should we just send it? */
-
- * channels/chan_sip.c: Fix issues where repeated messages where
- ignored, but retransmitted reliably instead of unreliably.
- Reported by: johan Patches: 12746.txt uploaded by oej (license
- 306) Tested by: johan (issue #12746)
-
-2008-07-08 00:01 +0000 [r128812-128856] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: Check for non-NULL before stripping
- characters. (closes issue #12954) Reported by: bfsworks Patches:
- 20080701__bug12954.diff.txt uploaded by Corydon76 (license 14)
- Tested by: deti
-
- * apps/app_voicemail.c: Stop using deprecated method, as requested
- by Kevin.
-
-2008-07-07 22:41 +0000 [r128795] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix handling of when a pvt disappears.
- Properly return the pvt locked and don't hold the pvt lock while
- destroying the ast_channel. (closes issue #13014) Reported by:
- jpgrayson Patches: chan_iax2_ast_iax2_new2.patch uploaded by
- jpgrayson (license 492)
-
-2008-07-07 20:47 +0000 [r128737] Sean Bright <sean.bright@gmail.com>
-
- * channels/chan_iax2.c: Remove spurious trailing whitespace from
- log messages and fix a spelling error in a log message. (closes
- issue #13017) Reported by: jpgrayson Patches:
- chan_iax2_space_after_newline.patch uploaded by jpgrayson
- (license 492) chan_iax2_spelling.patch uploaded by jpgrayson
- (license 492)
-
-2008-07-07 17:02 +0000 [r128639] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_iax2.c: By using the iaxdynamicthreadcount to
- identify a thread, it was possible for thread identifiers to be
- duplicated. By using a globally-unique monotonically- increasing
- integer, this is now avoided. (closes issue #13009) Reported by:
- jpgrayson Patches: chan_iax2_dyn_threadnum.patch uploaded by
- jpgrayson (license 492)
-
-2008-07-07 16:51 +0000 [r128637] Kevin P. Fleming <kpfleming@digium.com>
-
- * configure, configure.ac: use tzafrir's patch to fix this problem
- properly... i made the previous set of changes without thoroughly
- testing them, doh! (closes issue #12911) Reported by: tzafrir
- Patches: custum_dahdi_configure_2.diff uploaded by tzafrir
- (license 46) Tested by: tzafrir
-
-2008-07-04 16:11 +0000 [r127973-128029] Tilghman Lesher <tlesher@digium.com>
-
- * pbx/pbx_config.c: Move the free down one
-
- * main/pbx.c, include/asterisk/pbx.h, pbx/pbx_config.c: Fix the
- 'dialplan remove extension' logic, so that it a) works with
- cidmatch, and b) completes contexts correctly when the extension
- is ambiguous. (closes issue #12980) Reported by: licedey Patches:
- 20080703__bug12980.diff.txt uploaded by Corydon76 (license 14)
- Tested by: Corydon76
-
-2008-07-03 22:20 +0000 [r127754-127895] Kevin P. Fleming <kpfleming@digium.com>
-
- * apps/Makefile: remove this, it has been moved to the main
- Makefile
-
- * Makefile, main/editline/np/vis.c: a couple of small
- Solaris-related fixes (closes issue #11885) Reported by: snuffy,
- asgaroth
-
- * configure, main/Makefile, configure.ac, acinclude.m4: ensure that
- DAHDI_INCLUDE and ZAPTEL_INCLUDE are added in all the places
- needed improve AST_EXT_LIB_CHECK to accept (and remember)
- additional CFLAGS data like it does in trunk already (closes
- issue #12911) Reported by: tzafrir
-
-2008-07-03 00:16 +0000 [r127663] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, main/channel.c, channels/chan_dahdi.c, main/pbx.c,
- channels/chan_sip.c, res/res_features.c, include/asterisk/cdr.h:
- The CDRfix4/5/6 omnibus cdr fixes. (closes issue #10927) Reported
- by: murf Tested by: murf, deeperror (closes issue #12907)
- Reported by: falves11 Tested by: murf, falves11 (closes issue
- #11849) Reported by: greyvoip As to 11849, I think these changes
- fix the core problems brought up in that bug, but perhaps not the
- more global problems created by the limitations of CDR's
- themselves not being oriented around transfers. Reopen if necc,
- but bug reports are not the best medium for enhancement
- discussions. We need to start a second-generation CDR
- standardization effort to cover transfers. (closes issue #11093)
- Reported by: rossbeer Tested by: greyvoip, murf
-
-2008-07-02 20:47 +0000 [r127560] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_agent.c: Fix thread-safety of some of the
- pbx_builtin_getvar_helper calls
-
-2008-07-02 19:47 +0000 [r127501] Tilghman Lesher <tlesher@digium.com>
-
- * main/acl.c: Merged revisions 127466 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r127466 |
- tilghman | 2008-07-02 13:31:11 -0500 (Wed, 02 Jul 2008) | 6 lines
- Solaris fix (closes issue #12949) Reported by: snuffy Patches:
- bug_12949.diff uploaded by snuffy (license 35) ........
-
-2008-07-01 23:36 +0000 [r127244] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Add error message to failed open(2) calls
- inside the copy() function of app_voicemail. This idea came as
- part of my work in helping to resolve issue #12764.
-
-2008-07-01 20:25 +0000 [r126999-127133] Tilghman Lesher <tlesher@digium.com>
-
- * build_tools/cflags.xml, channels/chan_iax2.c: Disable the old,
- slow search for matching callno in chan_iax2 (but allow it to be
- reenabled for debugging)
-
- * channels/chan_iax2.c: Oops
-
- * channels/chan_iax2.c: Change around how we schedule pings and
- lagrqs, and fix a reason why the jobs were not getting properly
- cancelled. (closes issue #12903) Reported by: stevedavies
- Patches: 20080620__bug12903__2.diff.txt uploaded by Corydon76
- (license 14) Tested by: stevedavies
-
- * channels/chan_iax2.c: Suppress annoying warning by finding the
- remaining cases where the callno is not in the hash.
-
-2008-07-01 14:59 +0000 [r126735-126902] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Use domain part of SIP uri in register=
- configuration as fromdomain. Reported by: one47 Patches:
- sip-reg-fromdom2.dpatch uploaded by one47 (license 23) (closes
- issue #12474)
-
- * channels/chan_sip.c: Handle escaped URI's in call pickups. Patch
- by oej and IgorG. Reported by: IgorG Patches:
- bug12299-11062-v2.patch uploaded by IgorG (license 20) Tested by:
- IgorG, oej (closes issue #12299)
-
- * configs/sip.conf.sample: Clear up documentation on "domain="
- setting in sip.conf Reported by: davidw (closes issue #12413)
-
- * channels/chan_sip.c: Report 200 OK to all in-dialog OPTIONs
- requests (to confirm that the dialog exist). Don't bother
- checking the request URI. (closes issue #11264) Reported by: ibc
-
- * channels/chan_sip.c: Fix bad XML for hold notification. Reported
- by: gowen72 Patches: hold.patch uploaded by gowen72 (license 432)
- (closes issue #12942)
-
-2008-06-30 23:11 +0000 [r126680] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c: Load the proper channel configuration file
- based on which driver was detected.
-
-2008-06-30 22:30 +0000 [r126674] Tilghman Lesher <tlesher@digium.com>
-
- * configs/zapata.conf.sample: Add note about other names for
- EuroISDN
-
-2008-06-30 16:05 +0000 [r126573] Russell Bryant <russell@digium.com>
-
- * include/asterisk/lock.h: Fix a typo in the non-DEBUG_THREADS
- version of the recently added DEADLOCK_AVOIDANCE() macro. This
- caused the lock to not actually be released, and as a result, not
- avoid deadlocks at all. This resolves the issues reported in the
- last while about Asterisk locking up all over the place (and most
- commonly, in chan_iax2). (closes issue #12927) (closes issue
- #12940) (closes issue #12925) (potentially closes others ...)
-
-2008-06-30 12:50 +0000 [r126516] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Send all responses to an INVITE reliably, so
- that we retransmit if we don't get an ACK and also fail if we
- don't get the very same precious ACK. Based on patch by tsearle,
- with my own additions. (closes issue #12951) Reported by: tsearle
- Patches: busy_retransmit.patch uploaded by tsearle (license 373)
-
-2008-06-29 18:05 +0000 [r126395] Kevin P. Fleming <kpfleming@digium.com>
-
- * pbx/Makefile: ignore warnings for prototypes in GTK headers
-
-2008-06-27 22:01 +0000 [r125740-126056] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_sip.c: When we get a 408 Timeout, don't stop trying
- to re-register. (closes issue #12863) Reported by: ricvil
-
- * include/asterisk/tonezone_compat.h: Since HAVE_DAHDI is defined
- to HAVE_ZAPTEL in dahdi_compat.h, we must first check for
- HAVE_ZAPTEL. (closes issue #12938) Reported by: opticron Patches:
- tonezone_compat.diff uploaded by opticron (license 267)
-
- * main/utils.c, include/asterisk/lock.h: In this debugging
- function, copy to a buffer instead of using potentially unsafe
- pointers.
-
- * channels/chan_local.c: Add proper deadlock avoidance. (closes
- issue #12914) Reported by: ozan Patches:
- 20080625__bug12914.diff.txt uploaded by Corydon76 (license 14)
- Tested by: ozan
-
-2008-06-26 23:03 +0000 [r125587] Jason Parker <jparker@digium.com>
-
- * main/utils.c: Make sure to unlock the lock_info lock (huh?).
- Possible deadlock?
-
-2008-06-26 22:52 +0000 [r125476-125585] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Add the interface of a queue member to the
- output of the "queue show" command so that it can easily be
- associated with a queue member's name. This helps so that the
- appropriate queue member can be removed or paused since the
- interface is required, not the member's name. (closes issue
- #12783) Reported by: davevg Patches: app_queue.diff uploaded by
- davevg (license 209) with small mod from me
-
- * apps/app_queue.c: Backport of attended transfer queue_log patch
- from trunk. This patch allows for attended transfers to be logged
- in the queue_log the same way that blind transfers have always
- been. It was decided by popular opinion on the asterisk-dev
- mailing list that this should be backported to 1.4. Thanks to
- everyone who gave an opinion.
-
- * apps/app_queue.c: Prior to this patch, the "queue show" command
- used cached information for realtime queues instead of giving
- up-to-date info. Now realtime is queried for the latest and
- greatest in queue info. (closes issue #12858) Reported by: bcnit
- Patches: queue_show.patch uploaded by putnopvut (license 60)
-
-2008-06-26 16:32 +0000 [r125384] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Add support for peer realm based auth (a few
- missing lines, the rest is well documented but never worked)
-
-2008-06-26 15:30 +0000 [r125327] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_dahdi.c: ensure that (whenever possible) if we
- generate a log message because an ioctl() call to DAHDI/Zaptel
- failed, that we include the reason it failed by including the
- stringified error number (issue AST-80)
-
-2008-06-26 11:01 +0000 [r125218-125276] Tilghman Lesher <tlesher@digium.com>
-
- * main/rtp.c: Check for rtcp structure before trying to delete
- schedule. (closes issue #12872) Reported by: destiny6628 Patches:
- 20080621__bug12872.diff.txt uploaded by Corydon76 (license 14)
- Tested by: destiny6628
-
- * configs/agents.conf.sample: Document ackcall=always. (closes
- issue #12852) Reported by: davidw
-
-2008-06-25 22:21 +0000 [r125132] Kevin P. Fleming <kpfleming@digium.com>
-
- * apps/app_rpt.c, include/asterisk/dahdi_compat.h,
- channels/chan_dahdi.c, configure,
- include/asterisk/tonezone_compat.h (added), configure.ac: allow
- tonezone to live in a different place than DAHDI/Zaptel, since
- dahdi-tools and dahdi-linux are now separate packages and can be
- installed in different places don't include tonezone.h in
- dahdi_compat.h, because only a couple of modules need it get
- app_rpt building again after the DAHDI changes (closes issue
- #12911) Reported by: tzafrir
-
-2008-06-25 00:46 +0000 [r124908-124965] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_dahdi.c: Pvt deadlock causes some channels to get
- stuck in Reserved status. (closes issue #12621) Reported by:
- fabianoheringer Patches: 20080612__bug12621.diff.txt uploaded by
- Corydon76 (license 14) Tested by: fabianoheringer
-
- * apps/app_voicemail.c: Occasionally control characters find their
- way into CallerID. These need to be stripped prior to placing
- CallerID in the headers of an email. (closes issue #12759)
- Reported by: RobH Patches: 20080602__bug12759__2.diff.txt
- uploaded by Corydon76 (license 14) Tested by: RobH
-
- * channels/chan_sip.c: Don't access the pvt structure if unable to
- acquire the lock. (closes issue #12162) Reported by: norman
- Patches: 12162-lockfail.diff uploaded by qwell (license 4)
-
-2008-06-23 21:22 +0000 [r124743] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_iax2.c: emit a warning if the old IAX2 call
- searching code finds a call when the new code did not... so that
- we can get rid of the old code in 2-3 months
-
-2008-06-22 02:54 +0000 [r124540] Steve Murphy <murf@digium.com>
-
- * apps/app_forkcdr.c: (closes issue #12910) Reported by: chris-mac
- Sorry, my testing did not contain the simple case of forkCDR(v),
- I am much embarrassed to admit. If I had, I would have more
- solidly initialized the opts element for varset.
-
-2008-06-20 23:12 +0000 [r124395-124450] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_rpt.c: usleep with a value over 1,000,000 is
- nonportable. Changing to use sleep() instead. (closes issue
- #12814) Reported by: pputman Patches: app_rtp_sleep.patch
- uploaded by pputman (license 81)
-
- * main/app.c: If the last character in a string to be parsed is the
- delimiter, then we should count that final empty string as an
- additional argument.
-
-2008-06-20 21:14 +0000 [r124372] Jeff Gehlbach <jeffg@opennms.org>
-
- * doc/asterisk-mib.txt, doc/digium-mib.txt: Fix issues in
- digium-mib.txt and asterisk-mib.txt to placate smilint - bug
- 12905
-
-2008-06-20 20:16 +0000 [r124182-124315] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_local.c: When using a Local channel, started by a
- call file, with a destination of an AGI script, the AGI script
- does not always get notified of a hangup if the underlying
- channel hangs up early. (closes issue #11833) Reported by: IgorG
- Patches: local_hangup-v1.diff uploaded by IgorG (license 20)
-
- * channels/chan_dahdi.c: It's possible for a hangup to be received,
- even just after the initial cid spill. (closes issue #12453)
- Reported by: Alex728 Patches: 20080604__bug12453.diff.txt
- uploaded by Corydon76 (license 14)
-
-2008-06-19 20:28 +0000 [r124112] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Fix IMAP forwarding so that messages are
- sent to the proper mailbox. (closes issue #12897) Reported by:
- jaroth Patches: destination_forward.patch uploaded by jaroth
- (license 50)
-
-2008-06-19 19:55 +0000 [r124066] Brett Bryant <bbryant@digium.com>
-
- * main/utils.c: Merge revision 124064 from trunk. Add errors that
- report any locks held by threads when they are being closed.
-
-2008-06-19 16:58 +0000 [r123710-123930] Tilghman Lesher <tlesher@digium.com>
-
- * main/channel.c: Change informative messages to use the _multiple
- variant when multiple formats are possible. (Closes issue #12848)
- Reported by klaus3000
-
- * build_tools/strip_nonapi: Only process 40 arguments (20 files) at
- once with xargs, because some older shells may force xargs to
- separate on an odd boundary. (Closes issue #12883) Reported by
- Nik Soggia
-
- * configs/sip.conf.sample: Correct description of notifyringing
- option. (Closes issue #12890) Reported by gminet
-
- * main/asterisk.c: The RDTSC instruction was introduced on the
- Pentium line of microprocessors, and is not compatible with
- certain 586 clones, like Cyrix. Hence, asking for i386
- compatibility was always incorrect. See
- http://en.wikipedia.org/wiki/RDTSC (Closes issue #12886) Reported
- by tecnoxarxa
-
- * main/say.c, doc/lang (added), doc/lang/hebrew.ods (added): Add
- support for saying numbers in Hebrew. (closes issue #11662)
- Reported by: greenfieldtech Patches: say.c.patch-12042008
- uploaded by greenfieldtech (license 369) Hebrew-Sounds.ods
- uploaded by greenfieldtech (with signficant changes to the
- spreadsheet by me)
-
- * pbx/pbx_spool.c: Set the variables top-down, so that if a script
- sets a variable more than once, the last one will take
- precedence. (closes issue #12673) Reported by: phber Patches:
- 20080519__bug12673.diff.txt uploaded by Corydon76 (license 14)
-
-2008-06-17 20:26 +0000 [r123485] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: Make chan_sip build under dev mode with
- compilers >= GCC 4.2 Thanks to jpeeler for alerting me of this
-
-2008-06-17 18:56 +0000 [r123391] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_iax2.c: Fix 3 more places where not locking the
- structure could cause the wrong lock to be unlocked. (Closes
- issue #12795)
-
-2008-06-17 18:09 +0000 [r123274-123333] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: Cisco BTS sends SIP responses with a tab
- between the Cseq number and SIP request method in the Cseq:
- header. Asterisk did not handle this properly, but with this
- patch, all is well. (closes issue #12834) Reported by: tobias_e
- Patches: 12834.patch uploaded by putnopvut (license 60) Tested
- by: tobias_e
-
- * apps/app_queue.c: davidw pointed out that the holdtime
- calculation used by app_queue does not use "boxcar" filtering as
- the comments say. The term "boxcar" means that the number of
- samples used to calculate stays constant, with new samples
- replacing the oldest ones. The queue holdtime calculation uses
- all holdtime samples collected since the queue was loaded, so the
- comment has been changed to be accurate. (closes issue #12781)
- Reported by: davidw
-
-2008-06-17 15:48 +0000 [r123271] Russell Bryant <russell@digium.com>
-
- * main/astobj2.c: Fix a memory leak in astobj2 that was pointed out
- by seanbright. When a container got destroyed, the underlying
- bucket list entry for each object that was in the container at
- that time did not get free'd.
-
-2008-06-16 19:50 +0000 [r123110-123113] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_mgcp.c, channels/chan_dahdi.c,
- channels/chan_skinny.c, channels/chan_h323.c,
- channels/chan_iax2.c: Port "hasvoicemail" change from SIP to
- other channel drivers
-
- * channels/chan_sip.c: People expect that if "hasvoicemail" is set
- in users.conf, even if "mailbox" isn't set, that SIP will detect
- a mailbox. (closes issue #12855) Reported by: PLL Patches:
- 20080614__bug12855__2.diff.txt uploaded by Corydon76 (license 14)
- Tested by: PLL
-
-2008-06-16 12:31 +0000 [r122869-122919] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Only compare the first 15 characters so that
- even if the charset is specified we still accept it as SDP.
- (closes issue #12803) Reported by: lanzaandrea Patches:
- chan_sip.c.diff uploaded by lanzaandrea (license 496)
-
- * channels/chan_sip.c: Don't send a BYE on a dialog that is already
- gone during a REFER. (closes issue #12865) Reported by: flefoll
- Patches: chan_sip.c.br14.121495.patch-ALREADYGONE uploaded by
- flefoll (license 244)
-
-2008-06-13 21:44 +0000 [r122713] Mark Michelson <mmichelson@digium.com>
-
- * main/autoservice.c: Short circuit the loop in autoservice_run if
- there are no channels to poll. If we continued, then the result
- would be calling poll() with a NULL pollfd array. While this is
- fine with POSIX's poll(2) system call, those who use Asterisk's
- internal poll mechanism (Darwin systems) would have a failed
- assertion occur when poll is called. (related to issue #10342)
-
-2008-06-13 18:57 +0000 [r122663] Jeff Peeler <jpeeler@digium.com>
-
- * include/asterisk/dahdi_compat.h, res/res_musiconhold.c: fixed
- dahdi compatability header from assuming either dahdi or zaptel
- is installed (may not have either)
-
-2008-06-13 17:45 +0000 [r122617] Terry Wilson <twilson@digium.com>
-
- * apps/app_dial.c: Remove extra option from previous solution
- attempt
-
-2008-06-13 17:36 +0000 [r122613] Jeff Peeler <jpeeler@digium.com>
-
- * configure, configure.ac: (closes issue #12846) Reported by:
- Netview Tested by: jpeeler Use correct location to search for
- tonezone.
-
-2008-06-13 16:29 +0000 [r122589] Terry Wilson <twilson@digium.com>
-
- * apps/app_dial.c, res/res_features.c: This should fix the behavior
- of the 'T' dial feature being passed incorrectly to the
- transferee when builtin_atxfers are used. Also, doing a
- builtin_atxfer to parking was broken and is fixed here as well.
- (closes issue #11898) Reported by: sergee Tested by: otherwiseguy
-
-2008-06-12 19:08 +0000 [r122314] Jeff Peeler <jpeeler@digium.com>
-
- * main/indications.c, include/asterisk/dahdi_compat.h (added),
- main/loader.c, main/channel.c, channels/chan_dahdi.c (added),
- configure, apps/app_zapscan.c (removed), apps/app_zapras.c
- (removed), main/app.c, include/asterisk/options.h,
- apps/app_rpt.c, channels/chan_mgcp.c, apps/app_read.c,
- channels/chan_zap.c (removed), apps/app_page.c,
- include/asterisk/indications.h, apps/app_dahdiras.c (added),
- configure.ac, apps/app_disa.c, include/asterisk/channel.h,
- apps/app_getcpeid.c, apps/app_queue.c, apps/app_zapbarge.c
- (removed), channels/chan_misdn.c, apps/app_flash.c,
- build_tools/menuselect-deps.in, funcs/func_channel.c,
- main/file.c, res/snmp/agent.c, contrib/utils/zones2indications.c,
- codecs/codec_dahdi.c (added), res/res_indications.c,
- pbx/pbx_config.c, makeopts.in, apps/app_chanspy.c,
- main/asterisk.c, apps/app_dahdibarge.c (added),
- apps/app_meetme.c, include/asterisk/autoconfig.h.in,
- apps/app_dahdiscan.c (added), acinclude.m4,
- res/res_musiconhold.c, codecs/codec_zap.c (removed),
- channels/chan_iax2.c: Adds DAHDI support alongside Zaptel. DAHDI
- usage favored, but all Zap stuff should continue working. Release
- announcement to follow.
-
-2008-06-12 18:50 +0000 [r122311] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Properly play a holdtime message if the
- announce-holdtime option is set to "once." (closes issue #12842)
- Reported by: ramonpeek Patches: patch001.diff uploaded by
- ramonpeek (license 266)
-
-2008-06-12 18:22 +0000 [r122259] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix some race conditions that cause
- ast_assert() to report that chan_iax2 tried to remove an entry
- that wasn't in the scheduler
-
-2008-06-12 15:46 +0000 [r122208] Jeff Peeler <jpeeler@digium.com>
-
- * apps/app_parkandannounce.c, res/res_features.c: (closes issue
- #12193) Reported by: davidw Patch by: Corydon76, modified by me
- to work properly with ParkAndAnnounce app
-
-2008-06-12 15:18 +0000 [r122130-122137] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_meetme.c: Flipflop the sections for two options, since
- the section for 'X' (exit context) may otherwise absorb
- keypresses meant for 's' (admin/user menu). (closes issue #12836)
- Reported by: blitzrage Patches: 20080611__bug12836.diff.txt
- uploaded by Corydon76 (license 14) Tested by: blitzrage
-
- * main/channel.c: Occasionally, the alertpipe loses its nonblocking
- status, so detect and correct that situation before it causes a
- deadlock. (Reported and tested by ctooley via #asterisk-dev)
-
-2008-06-12 14:51 +0000 [r122127] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, apps/app_forkcdr.c: Arkadia tried to warn me, but the
- code added to ast_cdr_busy, _failed, and _noanswer was redundant.
- Didn't spot it until I was resolving conflicts in trunk. Ugh.
- Redundant code removed. It wasn't harmful. Just dumb.
-
-2008-06-12 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.21 released.
-
-2008-06-06 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.21-rc2 released.
-
-2008-06-05 18:03 +0000 [r120731-120735] Russell Bryant <russell@digium.com>
-
- * UPGRADE-1.2.txt: fix filename
-
- * UPGRADE-1.2.txt (added), UPGRADE.txt: Add the UPGRADE.txt file
- from Asterisk 1.2, for handy reference.
-
-2008-06-05 16:56 +0000 [r120675] Philippe Sultan <philippe.sultan@gmail.com>
-
- * res/res_jabber.c: Ignore appended resource when comparing JIDs.
-
-2008-06-05 16:38 +0000 [r120671] Russell Bryant <russell@digium.com>
-
- * doc/smdi.txt, res/res_smdi.c: It turns out that searching on the
- forwarding station isn't very useful for most people, so pull in
- the changes that allow searching for SMDI messages based on other
- components of the SMDI message. Also, update the SMDI
- documentation.
-
-2008-06-04 22:05 +0000 [r120513] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Make sure that the string we set will survive
- the unref of the queue member. Thanks to Russell, who pointed
- this out.
-
-2008-06-04 18:35 +0000 [r120425] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_zap.c: If we fail to setup the PRI request channel,
- don't continue, exit with an error. (closes issue #11989)
- Reported by: Corydon76 Patches: 20080213__zap_memleak.diff.txt
- uploaded by Corydon76 (license 14)
-
-2008-06-04 16:26 +0000 [r120371] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_config.c: Make the "dialplan remove include" CLI command
- actually work. Also, tweak some formatting, and make the success
- message a little bit more clear. (closes AST-52)
-
-2008-06-04 14:11 +0000 [r120285] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Tab completion when removing a member should
- give the member's interface, not the name, since the interface is
- what is expected for the command. (closes issue #12783) Reported
- by: davevg
-
-2008-06-04 13:31 +0000 [r120282] Joshua Colp <jcolp@digium.com>
-
- * main/pbx.c, pbx/pbx_config.c: Fix a log message and add a message
- for when the dialplan is done reloading. (closes issue #12716)
- Reported by: chappell Patches: dialplan_reload_2.diff uploaded by
- chappell (license 8)
-
-2008-06-03 22:41 +0000 [r120226] Tilghman Lesher <tlesher@digium.com>
-
- * pbx/pbx_loopback.c: Due to incorrect use of the
- AST_LIST_INSERT_HEAD() macro the loopback switch cannot perform
- any translation on the extension number before searching for it
- in the target context. (closes issue #12473) Reported by:
- chappell Patches: pbx_loopback.c.diff uploaded by chappell
- (license 8)
-
-2008-06-03 22:15 +0000 [r120173] Jeff Peeler <jpeeler@digium.com>
-
- * main/config.c: (closes issue #11594) Reported by: yem Tested by:
- yem This change decreases the buffer size allocated on the stack
- substantially in config_text_file_load when LOW_MEMORY is turned
- on. This change combined with the fix from revision 117462
- (making mkintf not copy the zt_chan_conf structure) was enough to
- prevent the crash.
-
-2008-06-03 21:34 +0000 [r120168] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix another place where peer->callno could
- change at a very bad time, and also fix a place where a peer was
- used after the reference was released. (inspired by rev 120001)
-
-2008-06-03 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.21-rc1 released.
-
-2008-06-03 18:23 +0000 [r120001-120061] Tilghman Lesher <tlesher@digium.com>
-
- * main/manager.c: When listing the manager users, managers in
- users.conf are not shown, even though they are allowed to
- connect. (closes issue #12594) Reported by: bkruse Patches:
- 12594-managerusers-2.diff uploaded by qwell (license 4) Tested
- by: bkruse
-
- * channels/chan_iax2.c: Save the callno when we're poking, because
- our peer structure could change during deadlock avoidance (and
- thus we unlock the wrong callno, causing a cascade failure).
- (closes issue #12717) Reported by: gewfie Patches:
- 20080525__bug12717.diff.txt uploaded by Corydon76 (license 14)
- Tested by: gewfie
-
-2008-06-03 15:26 +0000 [r119929-119966] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18,
- pbx/ael/ael-test/ref.ael-vtest13,
- pbx/ael/ael-test/ref.ael-vtest17,
- pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
- pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-test5,
- pbx/ael/ael-test/ref.ael-test15: Updated the regressions on AEL.
- Hadn't updated this for the changes I made to preserve ${EXTEN}
- in switches, which affected several tests because it adds extra
- priorities, and at least one needed to be updated because of the
- removal of the empty extension warning message.
-
- * pbx/pbx_ael.c: as per
- http://lists.digium.com/pipermail/asterisk-users/2008-June/212934.html,
- which is a message from Philipp Kempgen, requesting that the
- WARNING that an extension is empty be reduced to a NOTICE or
- less, as empty extensions are syntactically possible, and no big
- deal. With which I agree, and have removed that WARNING message
- entirely. I think it is not necessary to see this message. It
- didn't state that a NoOp() was inserted automatically on your
- behalf, and really, as users, who cares? Why freak out dialplan
- writers with unnecessary warnings? The details of the
- machinations a compiler goes thru to produce working assembly
- code is of little interest to most programmers-- we will follow
- the unix principal of doing our work silently.
-
-2008-06-03 14:46 +0000 [r119926] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Treat ECONNREFUSED as an error that will
- stop further retransmissions. (issue #AST-58, patch from
- Switchvox)
-
-2008-06-02 20:08 +0000 [r119742-119838] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Revert a change made for issue #12479. This
- change caused a regression such that a dial string such as
- (IAX2/foo) did not automatically fall back to dialing the 's'
- extension anymore. (closes issue #12770) Reported by: dagmoller
-
- * main/manager.c: Improve CLI command blacklist checking for the
- command manager action. Previously, it did not handle case or
- whitespace properly. This made it possible for blacklisted
- commands to get executed anyway. (closes issue #12765)
-
-2008-06-02 14:32 +0000 [r119740] Philippe Sultan <philippe.sultan@gmail.com>
-
- * channels/chan_gtalk.c, res/res_jabber.c: Do not link the guest
- account with any configured XMPP client (in jabber.conf). The
- actual connection is made when a call comes in Asterisk. Fix the
- ast_aji_get_client function that was not able to retrieve an XMPP
- client from its JID. (closes issue #12085) Reported by: junky
- Tested by: phsultan
-
-2008-06-02 12:30 +0000 [r119687] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Even of the first PING or LAGRQ doesn't get
- sent because it comes up too soon, make sure to reschedule so it
- gets sent later.
-
-2008-06-02 09:29 +0000 [r119585-119636] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c: fixed compile issue when dev-mode is
- enabled
-
- * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h: Added
- counter for unhandled_bmsg Print, this prevents the logs to be
- flooded to fast and save CPU in this error scenario. Added
- 'last_used' element to bc structure, when a bchannel changes from
- used to free this exact time will be marked in last_used. When a
- new channel is requested the find_free_chan function will check
- if the new empty channel was used within the last second, if yes
- it will search for the next channel, if no it will return this
- channel. This simple mechanism has prooven to prevent race
- conditions where the NT and TE tried to allocate the exact same
- channel at the same time (RELEASE cause: 44).
-
-2008-06-02 01:06 +0000 [r119530-119533] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Change a debug message to an actual debug
- message
-
- * apps/app_dial.c: Fix another typo in documentation
-
-2008-06-01 20:47 +0000 [r119478] Michiel van Baak <michiel@vanbaak.info>
-
- * apps/app_dial.c: small typo fix 'retires' => 'retries'
-
-2008-05-30 21:17 +0000 [r119404] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_queue.c: When joinempty=strict, it only failed on join
- if there were busy members. If all members were logged out OR
- paused, then it (incorrectly) let callers join the queue. (closes
- issue #12451) Reported by: davidw
-
-2008-05-30 19:46 +0000 [r119354] Joshua Colp <jcolp@digium.com>
-
- * main/autoservice.c: Fix a bug I found while testing for another
- issue.
-
-2008-05-30 16:44 +0000 [r119301] Michiel van Baak <michiel@vanbaak.info>
-
- * contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk,
- contrib/init.d/rc.debian.asterisk,
- contrib/init.d/rc.mandrake.asterisk,
- contrib/init.d/rc.redhat.asterisk,
- contrib/init.d/rc.gentoo.asterisk,
- contrib/init.d/rc.slackware.asterisk: dont use a bashism way to
- check the $VERSION variable. The rc/init.d scripts, and
- safe_asterisk work on normal sh now again. Tested on: OpenBSD 4.2
- (me) Debian etch (me) Ubuntu Hardy (me and loloski) FC9 (loloski)
- (closes issue #12687) Reported by: loloski Patches:
- 20080529-12687-safe_asterisk-fixversion.diff.txt uploaded by
- mvanbaak (license 7) Tested by: loloski, mvanbaak
-
-2008-05-30 12:55 +0000 [r119076-119238] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 119237 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r119237 | russell | 2008-05-30 07:49:39 -0500 (Fri, 30
- May 2008) | 7 lines - Instead of only enforcing destination call
- number checking on an ACK, check all full frames except for PING
- and LAGRQ, which may be sent by older versions too quickly to
- contain the destination call number. (As suggested by Tim Panton
- on the asterisk-dev list) - Merge changes from
- team/russell/iax2-frame-race, which prevents PING and LAGRQ from
- being sent before the destination call number is known. ........
-
- * main/autoservice.c: Fix a race condition in channel autoservice.
- There was still a small window of opportunity for a DTMF frame,
- or some other deferred frame type, to come in and get dropped.
- (closes issue #12656) (closes issue #12656) Reported by: dimas
- Patches: v3-12656.patch uploaded by dimas (license 88) -- with
- some modifications by me
-
- * include/asterisk/audiohook.h: Oddly enough, all of the contents
- of audiohook.h were in there twice. I have removed the second
- copy.
-
-2008-05-29 20:24 +0000 [r119071] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_zap.c: Call waiting tone occurs too often, because
- it's getting serviced by both subchannels. (closes issue #11354)
- Reported by: cahen Patches: 20080512__bug11354.diff.txt uploaded
- by Corydon76 (license 14)
-
-2008-05-29 19:04 +0000 [r118956-119012] Russell Bryant <russell@digium.com>
-
- * apps/app_milliwatt.c: - Fix a typo in the argument to Playtones -
- use ast_safe_sleep() instead of calling the wait application
- (thanks to tilghman for pointing these out!)
-
- * /, channels/chan_iax2.c: Merged revisions 119008 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r119008 | russell | 2008-05-29 13:45:21 -0500 (Thu, 29
- May 2008) | 7 lines Merge changes from
- team/russell/iax2-another-fix-to-the-fix As described in the
- following post to the asterisk-dev mailing list, only enforce
- destination call numbers when processing an ACK.
- http://lists.digium.com/pipermail/asterisk-dev/2008-May/033217.html
- (closes issue #12631) ........
-
- * apps/app_milliwatt.c: - Mark app_milliwatt dependent on
- res_indications (thanks to jsmith) - fix a typo in a log message
- (thanks to qwell)
-
- * apps/app_milliwatt.c: Change milliwatt to use the proper tone by
- default (1004 Hz) instead of 1000 Hz. An option is there to use
- 1000 Hz for anyone that might want it.
-
-2008-05-29 17:33 +0000 [r118953-118954] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/lock.h: Define also when not DEBUG_THREADS
-
- * channels/chan_mgcp.c, channels/chan_zap.c, channels/chan_sip.c,
- channels/chan_agent.c, channels/chan_alsa.c, main/utils.c,
- include/asterisk/lock.h, channels/chan_iax2.c: Add some debugging
- code that ensures that when we do deadlock avoidance, we don't
- lose the information about how a lock was originally acquired.
-
-2008-05-29 00:25 +0000 [r118858] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, apps/app_forkcdr.c: (closes issue #10668) (closes
- issue #11721) (closes issue #12726) Reported by: arkadia Tested
- by: murf These changes: 1. revert the changes made via bug 10668;
- I should have known that such changes, even tho they made sense
- at the time, seemed like an omission, etc, were actually integral
- to the CDR system via forkCDR. It makes sense to me now that
- forkCDR didn't natively end any CDR's, but rather depended on
- natively closing them all at hangup time via traversing and
- closing them all, whether locked or not. I still don't completely
- understand the benefits of setvar and answer operating on locked
- cdrs, but I've seen enough to revert those changes also, and stop
- messing up users who depended on that behavior. bug 12726 found
- reverting the changes fixed his changes, and after a long review
- and working on forkCDR, I can see why. 2. Apply the suggested
- enhancements proposed in 10668, but in a completely compatible
- way. ForkCDR will behave exactly as before, but now has new
- options that will allow some actions to be taken that will
- slightly modify the outcome and side-effects of forkCDR. Based on
- conversations I've had with various people, these small tweaks
- will allow some users to get the behavior they need. For
- instance, users executing forkCDR in an AGI script will find the
- answer time set, and DISPOSITION set, a situation not covered
- when the routines were first written. 3. A small problem in the
- cdr serializer would output answer and end times even when they
- were not set. This is now fixed.
-
-2008-05-28 16:10 +0000 [r118716] Brett Bryant <bbryant@digium.com>
-
- * channels/chan_iax2.c: merge revision 118702 from trunk to 1.4 --
- Fixes a bug in chan_iax that uses send_command to poke a peer
- while a channel is unlocked in some cases, and because it can
- cause seemingly random failures could be related to some bugs in
- the tracker...
-
-2008-05-28 14:23 +0000 [r118558-118646] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add an
- option to use the source IP address of RTP as the destination IP
- address of UDPTL when a specific option is enabled. If the remote
- side is properly configured (ports forwarded) then UDPTL will
- flow. (closes issue #10417) Reported by: cstadlmann
-
- * channels/chan_sip.c: Fix an issue where codec preferences were
- not set on dialogs that were not authenticated via a user or peer
- and allow framing to work without rtpmap in the SDP. (closes
- issue #12501) Reported by: slimey
-
-2008-05-27 19:15 +0000 [r118551] Tilghman Lesher <tlesher@digium.com>
-
- * main/cli.c: When showing an error message for a command, don't
- shorten the command output, as it tends to confuse the user (it's
- fine for suggesting other commands, however). Reported by:
- seanbright (on #asterisk-dev) Fixed by: me
-
-2008-05-27 19:07 +0000 [r118509] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_chanspy.c: Russell noted to me that in the case that
- separate threads use their own addressing system, the fix I made
- for issue 12376 does not guarantee uniqueness to the datastores'
- uids. Though I know of no system that works this way, I am going
- to change this right now to prevent trying to track down some
- future bug that may occur and cause untold hours of debugging
- time to track down. The change involves using a global counter
- which increases with each new chanspy_ds which is created. This
- guarantees uniqueness.
-
-2008-05-27 18:58 +0000 [r118465] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c: NULL character should terminate only commands
- back to the core, not log messages to the console. (closes issue
- #12731) Reported by: seanbright Patches:
- 20080527__bug12731.diff.txt uploaded by Corydon76 (license 14)
- Tested by: seanbright
-
-2008-05-27 17:17 +0000 [r118416] Michiel van Baak <michiel@vanbaak.info>
-
- * apps/app_voicemail.c: small update to the g() option of
- app_voicemail to note that gain changes only work on zap channels
- right now. issue #12578 shows it's not clear right now.
-
-2008-05-27 16:38 +0000 [r118365] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_chanspy.c: Add a unique id to the datastore allocated in
- app_chanspy since it is possible that multiple spies may be
- listening to the same channel. (closes issue #12376) Reported by:
- DougUDI Patches: 12376_chanspy_uid.diff uploaded by putnopvut
- (license 60) Tested by: destiny6628 (closes issue #12243)
- Reported by: atis
-
-2008-05-27 15:45 +0000 [r118358] Tilghman Lesher <tlesher@digium.com>
-
- * configs/queues.conf.sample: Add a note that pbx_config.so is
- needed for Local channels. (Closes issue #12671)
-
-2008-05-25 16:02 +0000 [r118251] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_sip.c: Realtime flag affects construction in
- multiple ways, so consulting whether rtcachefriends was set was
- done too soon (needed to be done inside build_peer, not just as a
- flag to build_peer). Also, fullcontact needed to be
- reconstructed, because realtime separates the embedded ';' into
- multiple fields. (closes issue #12722) Reported by: barthpbx
- Patches: 20080525__bug12722.diff.txt uploaded by Corydon76
- (license 14) Tested by: barthpbx (Much of the discussion happened
- on #asterisk-dev for diagnosing this issue)
-
-2008-05-23 21:21 +0000 [r118163] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_zap.c: Fix a few things I missed to ensure
- zt_chan_conf structure is not modified in mkintf
-
-2008-05-23 13:18 +0000 [r118052-118055] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/utils.h: Add format type checking for recently
- de-inlined function
-
- * doc/cli.txt (added), doc/00README.1st: Add information on using
- the Asterisk console, including tab command line completion.
- (Closes issue #12681)
-
-2008-05-23 12:30 +0000 [r118048] Russell Bryant <russell@digium.com>
-
- * include/asterisk/utils.h, main/utils.c: Don't declare a function
- that takes variable arguments as inline, because it's not valid,
- and on some compilers, will emit a warning.
- http://gcc.gnu.org/onlinedocs/gcc/Inline.html#Inline (closes
- issue #12289) Reported by: francesco_r Patches by Tilghman, final
- patch by me
-
-2008-05-22 18:53 +0000 [r117809-117899] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c: Also remove preamble from asynchronous events
- (reported by jsmith on #asterisk-dev)
-
- * funcs/func_realtime.c: Take into account the length of delimiters
- when calculating result string length. (closes issue #12696)
- Reported by: adomjan Patches: func_realtime.c-longdelimiter.patch
- uploaded by adomjan (license 487)
-
-2008-05-21 20:11 +0000 [r117582] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_zap.c: Ensure that passed in zt_chan_conf structure
- is not modified in mkintf.
-
-2008-05-21 19:38 +0000 [r117574] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Apply the autoframing setting to dialogs
- that do not get matched against a user or peer.
-
-2008-05-21 18:44 +0000 [r117519-117523] Tilghman Lesher <tlesher@digium.com>
-
- * pbx/pbx_spool.c: Revert accidental commit of the last change
-
- * main/asterisk.c, pbx/pbx_spool.c: Strip the preamble from the
- output also when -rx is not being used (Related to issue #12702)
-
-2008-05-21 18:28 +0000 [r117479-117514] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c: Don't filter the magic character in the network
- verboser. It gets filtered once it reaches the client. (related
- to issue #12702, pointed out by tilghman)
-
- * main/asterisk.c, pbx/pbx_gtkconsole.c: 1) Don't print the verbose
- marker in front of every message from ast_verbose() being sent to
- remote consoles. 2) Fix pbx_gtkconsole to filter out the verbose
- marker. (related to issue #12702)
-
- * main/asterisk.c: Don't display the verbose marker for calls to
- ast_verbose() that do not include a VERBOSE_PREFIX in front of
- the message. (closes issue #12702) Reported by: johnlange Patched
- by me
-
-2008-05-21 16:58 +0000 [r117462] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_zap.c: Pass a pointer for the conf parameter to the
- function mkintf rather than the whole zt_chan_conf structure.
-
-2008-05-20 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.20 released.
-
-2008-05-14 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.20-rc3 released.
-
-2008-05-14 12:51 +0000 [r116230] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Accept text messages even with Content-Type:
- text/plain;charset=Södermanländska
-
-2008-05-13 23:47 +0000 [r116088] Mark Michelson <mmichelson@digium.com>
-
- * main/channel.c, include/asterisk/lock.h: A change to the way
- channel locks are handled when DEBUG_CHANNEL_LOCKS is defined.
- After debugging a deadlock, it was noticed that when
- DEBUG_CHANNEL_LOCKS is enabled in menuselect, the actual origin
- of channel locks is obscured by the fact that all channel locks
- appear to happen in the function ast_channel_lock(). This code
- change redefines ast_channel_lock to be a macro which maps to
- __ast_channel_lock(), which then relays the proper file name,
- line number, and function name information to the core lock
- functions so that this information will be displayed in the case
- that there is some sort of locking error or core show locks is
- issued.
-
-2008-05-13 21:17 +0000 [r115990-116038] Russell Bryant <russell@digium.com>
-
- * channels/chan_local.c: Fix a deadlock involving channel
- autoservice and chan_local that was debugged and fixed by
- mmichelson and me. We observed a system that had a bunch of
- threads stuck in ast_autoservice_stop(). The reason these threads
- were waiting around is because this function waits to ensure that
- the channel list in the autoservice thread gets rebuilt before
- the stop() function returns. However, the autoservice thread was
- also locked, so the autoservice channel list was never getting
- rebuilt. The autoservice thread was stuck waiting for the channel
- lock on a local channel. However, the local channel was locked by
- a thread that was stuck in the autoservice stop function. It
- turned out that the issue came down to the local_queue_frame()
- function in chan_local. This function assumed that one of the
- channels passed in as an argument was locked when called.
- However, that was not always the case. There were multiple cases
- in which this channel was not locked when the function was
- called. We fixed up chan_local to indicate to this function
- whether this channel was locked or not. The previous assumption
- had caused local_queue_frame() to improperly return with the
- channel locked, where it would then never get unlocked. (closes
- issue #12584) (related to issue #12603)
-
- * main/autoservice.c: Fix an issue that I noticed in autoservice
- while mmichelson and I were debugging a different problem. I
- noticed that it was theoretically possible for two threads to
- attempt to start the autoservice thread at the same time. This
- change makes the process of starting the autoservice thread,
- thread-safe.
-
-2008-05-13 20:28 +0000 [r115944] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_alsa.c: Use the right flag to open the audio in
- non-blocking. (closes issue #12616) Reported by:
- nicklewisdigiumuser
-
-2008-05-13 18:36 +0000 [r115884] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c: If the socket dies (read returns 0=EOF), return
- immediately. (Closes issue #12637)
-
-2008-05-12 17:51 +0000 [r115735] Mark Michelson <mmichelson@digium.com>
-
- * main/utils.c: If a thread holds no locks, do not print any
- information on the thread when issuing a core show locks command.
- This will help to de-clutter output somewhat. Russell said it
- would be fine to place this improvement in the 1.4 branch, so
- that's why it's going here too.
-
-2008-05-09 16:34 +0000 [r115579] Joshua Colp <jcolp@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac:
- Improve res_ninit and res_ndestroy autoconf logic on the Darwin
- platform.
-
-2008-05-08 19:19 +0000 [r115545-115568] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Remove debug output.
-
- * /, channels/chan_iax2.c: Merged revisions 115564 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r115564 | russell | 2008-05-08 14:14:04 -0500 (Thu, 08
- May 2008) | 25 lines Fix a race condition that bbryant just found
- while doing some IAX2 testing. He was running Asterisk trunk
- running IAX2 calls through a few Asterisk boxes, however, the
- audio was extremely choppy. We looked at a packet trace and saw a
- storm of INVAL and VNAK frames being sent from one box to
- another. It turned out that what had happened was that one box
- tried to send a CONTROL frame before the 3 way handshake had
- completed. So, that frame did not include the destination call
- number, because it didn't have it yet. Part of our recent work
- for security issues included an additional check to ensure that
- frames that are supposed to include the destination call number
- have the correct one. This caused the frame to be rejected with
- an INVAL. The frame would get retransmitted for forever, rejected
- every time ... This race condition exists in all versions that
- got the security changes, in theory. However, it is really only
- likely that this would cause a problem in Asterisk trunk. There
- was a control frame being sent (SRCUPDATE) at the _very_
- beginning of the call, which does not exist in 1.2 or 1.4.
- However, I am fixing all versions that could potentially be
- affected by the introduced race condition. These changes are what
- bbryant and I came up with to fix the issue. Instead of simply
- dropping control frames that get sent before the handshake is
- complete, the code attempts to wait a little while, since in most
- cases, the handshake will complete very quickly. If it doesn't
- complete after yielding for a little while, then the frame gets
- dropped. ........
-
- * channels/chan_sip.c: Don't give up on attempting an outbound
- registration if we receive a 408 Timeout. (closes issue #12323)
-
- * contrib/scripts/postgres_cdr.sql (removed): remove
- postgres_cdr.sql, as the CDR schema is in realtime_pgsql.sql, as
- well (closes issue #9676)
-
- * contrib/init.d/rc.debian.asterisk: Don't exit the script if
- Asterisk is not running. (closes issue #12611)
-
- * main/pbx.c: Don't use a channel before checking for channel
- allocation failure. (closes issue #12609) Reported by: edantie
-
- * contrib/init.d/rc.debian.asterisk: Use the same method for
- executing Asterisk as the rest of the script. (closes issue
- #12611) Reported by: b_plessis
-
-2008-05-07 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.20-rc2 released.
-
-2008-05-07 18:17 +0000 [r115512-115517] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Track peer references when stored in the
- sip_pvt struct as the peer related to a qualify ping or a
- subscription. This fixes some realtime related crashes. (closes
- issue #12588) (closes issue #12555)
-
-2008-05-06 19:55 +0000 [r115418-115422] Jason Parker <jparker@digium.com>
-
- * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 115421
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r115421 | qwell | 2008-05-06 14:54:57 -0500 (Tue, 06 May 2008) |
- 7 lines read requires an argument on some non-bash shells (closes
- issue #12593) Reported by: bkruse Patches:
- getilbc.sh_12593_v1.diff uploaded by bkruse (license 132)
- ........
-
- * res/res_musiconhold.c: Switch to using ast_random() rather than
- just rand(). This does not fix the bug reported, but I believe it
- is correct. (from issue #12446) Patches: bug_12446.diff uploaded
- by snuffy (license 35)
-
-2008-05-06 19:31 +0000 [r115415] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c: Don't print the terminating NUL. (Closes issue
- #12589)
-
-2008-05-06 13:54 +0000 [r115341] Joshua Colp <jcolp@digium.com>
-
- * configure, configure.ac: Add in missing argument.
-
-2008-05-05 22:50 +0000 [r115333] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c, main/logger.c: Separate verbose output from CLI
- output, by using a preamble. (closes issue #12402) Reported by:
- Corydon76 Patches: 20080410__no_verbose_in_rx_output.diff.txt
- uploaded by Corydon76 (license 14)
- 20080501__no_verbose_in_rx_output__1.4.diff.txt uploaded by
- Corydon76 (license 14)
-
-2008-05-05 22:10 +0000 [r115327] Joshua Colp <jcolp@digium.com>
-
- * build_tools/menuselect-deps.in, configure,
- include/asterisk/autoconfig.h.in, codecs/codec_speex.c,
- configure.ac: Make sure that either the main speex library
- contains preprocess functions or that speexdsp does. If both fail
- then speex stuff can not be built.
-
-2008-05-05 21:41 +0000 [r115320] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Don't consider a caller "handled" until the
- caller is bridged with a queue member. There was too much of an
- opportunity for the member to hang up (either during a delay,
- announcement, or overly long agi) between the time that he
- answered the phone and the time when he actually was bridged with
- the caller. The consequence of this was that if the member hung
- up in that interval, then proper abandonment details would not be
- noted in the queue log if the caller were to hang up at any point
- after the member hangup. (closes issue #12561) Reported by:
- ablackthorn
-
-2008-05-05 20:17 +0000 [r115308-115312] Tilghman Lesher <tlesher@digium.com>
-
- * Makefile: Reverse order, such that user configs override default
- selections
-
- * include/asterisk/res_odbc.h: Err, the documentation on the return
- value of ast_odbc_backslash_is_escape is exactly backwards.
-
-2008-05-05 19:49 +0000 [r115297-115304] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Avoid putting opaque="" in Digest
- authentication. This patch came from switchvox. It fixes
- authentication with Primus in Canada, and has been in use for a
- very long time without causing problems with any other providers.
- (closes issue AST-36)
-
-2008-05-05 03:22 +0000 [r115285] Tilghman Lesher <tlesher@digium.com>
-
- * contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk,
- contrib/init.d/rc.debian.asterisk,
- contrib/init.d/rc.mandrake.asterisk,
- contrib/init.d/rc.redhat.asterisk,
- contrib/init.d/rc.gentoo.asterisk,
- contrib/init.d/rc.slackware.asterisk: When starting Asterisk, bug
- out if Asterisk is already running. (closes issue #12525)
- Reported by: explidous Patches: 20080428__bug12525.diff.txt
- uploaded by Corydon76 (license 14) Tested by: mvanbaak
-
-2008-05-04 02:09 +0000 [r115276-115282] Joshua Colp <jcolp@digium.com>
-
- * configure, acinclude.m4: Expand the test function for GCC
- attributes so that more complex attributes are properly
- recognized.
-
- * include/asterisk/compiler.h: For my next trick I will make these
- work with what our autoconf header file gives us.
-
- * configure, acinclude.m4: Treat warnings as errors when checking
- if a GCC attribute exists. We have to do this as GCC will just
- ignore the attribute and pop up a warning, it won't actually fail
- to compile.
-
-2008-05-02 20:25 +0000 [r115257] Brett Bryant <bbryant@digium.com>
-
- * channels/chan_zap.c, configure, include/asterisk/autoconfig.h.in,
- configure.ac, CHANGES: Add new "pri show version" command to show
- the libpri version for support reasons.
-
-2008-05-02 14:28 +0000 [r115196] Mark Michelson <mmichelson@digium.com>
-
- * include/asterisk/sched.h: Clarify a comment that was, well, just
- wrong. It turns out that ignoring the way that macros expand.
- Instead, I have clarified in the comment why the macro will work
- even if the scheduler id for the task to be deleted changes
- during the execution of the macro.
-
-2008-05-01 23:20 +0000 [r115017-115102] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/res_odbc.h: Change the comment of deprecated to
- an actual compiler deprecation
-
- * main/utils.c: '#' is another reserved character for URIs that
- also needs to be escaped. (closes issue #10543) Reported by:
- blitzrage Patches: 20080418__bug10543.diff.txt uploaded by
- Corydon76 (license 14)
-
-2008-05-01 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.20-rc1 released.
-
-2008-04-30 16:30 +0000 [r114891] Russell Bryant <russell@digium.com>
-
- * include/asterisk/dlinkedlists.h (added), channels/chan_iax2.c:
- Merge changes from team/russell/iax2_find_callno and
- iax2_find_callno_1.4 These changes address a critical performance
- issue introduced in the latest release. The fix for the latest
- security issue included a change that made Asterisk randomly
- choose call numbers to make them more difficult to guess by
- attackers. However, due to some inefficient (this is by far, an
- understatement) code, when Asterisk chose high call numbers,
- chan_iax2 became unusable after just a small number of calls. On
- a small embedded platform, it would not be able to handle a
- single call. On my Intel Core 2 Duo @ 2.33 GHz, I couldn't run
- more than about 16 IAX2 channels. Ouch. These changes address
- some performance issues of the find_callno() function that have
- bothered me for a very long time. On every incoming media frame,
- it iterated through every possible call number trying to find a
- matching active call. This involved a mutex lock and unlock for
- each call number checked. So, if the random call number chosen
- was 20000, then every media frame would cause 20000 locks and
- unlocks. Previously, this problem was not as obvious since
- Asterisk always chose the lowest call number it could. A second
- container for IAX2 pvt structs has been added. It is an astobj2
- hash table. When we know the remote side's call number, the pvt
- goes into the hash table with a hash value of the remote side's
- call number. Then, lookups for incoming media frames are a very
- fast hash lookup instead of an absolutely insane array traversal.
- In a quick test, I was able to get more than 3600% more IAX2
- channels on my machine with these changes.
-
-2008-04-30 16:23 +0000 [r114890] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Don't crash on bad SIP replys. Fix created
- in Huntsville together with Mark M (putnopvut) (closes issue
- #12363) Reported by: jvandal Tested by: putnopvut, oej
-
-2008-04-30 14:46 +0000 [r114875-114880] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/iax2.h, channels/chan_iax2.c: use the ARRAY_LEN macro
- for indexing through the iaxs/iaxsl arrays so that the size of
- the arrays can be adjusted in one place, and change the size of
- the arrays from 32768 calls to 2048 calls when LOW_MEMORY is
- defined
-
- * Makefile.rules: pay attention to *all* header files for
- dependency tracking, not just the local ones (inspired by r578 of
- asterisk-addons by tilghman)
-
-2008-04-29 19:40 +0000 [r114848] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Use the MACRO_CONTEXT and MACRO_EXTEN channel
- variables instead of the channel's macrocontext and macroexten
- fields. This is needed because if macros are daisy-chained, the
- incorrect context and extension are placed on the new channel. I
- also added locking to the channel prior to accessing these
- variables as noted in trunk's janitor project file. (closes issue
- #12549) Reported by: darren1713 Patches:
- app_queue.c.macroextenpatch uploaded by darren1713 (license 116)
- (with modifications from me) Tested by: putnopvut
-
-2008-04-29 17:08 +0000 [r114829] Jason Parker <jparker@digium.com>
-
- * res/res_config_pgsql.c: Change warning message to debug, since
- there are cases where 0 results is perfectly fine.
-
-2008-04-29 12:53 +0000 [r114823] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 114822
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r114822 | kpfleming | 2008-04-29 07:52:32 -0500 (Tue, 29 Apr
- 2008) | 2 lines stop script from appending source code if run
- multiple times ........
-
-2008-04-28 04:47 +0000 [r114708] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c, channels/chan_gtalk.c: When modules are
- embedded, they take on a different name, without the ".so"
- extension. Specifically check for this name, when we're checking
- if a module is loaded. (Closes issue #12534)
-
-2008-04-27 01:26 +0000 [r114695] Sean Bright <sean.bright@gmail.com>
-
- * configure, configure.ac: When we don't explicitly pass a path to
- the --with-tds configure option, we may end up finding tds.h in
- /usr/local/include instead of /usr/include. If this happens, the
- grep that looks for the version (from tdsver.h) will fail and
- we'll have some problems during the build.
-
-2008-04-26 13:15 +0000 [r114689] Tilghman Lesher <tlesher@digium.com>
-
- * contrib/scripts/vmail.cgi: Clicking forward without selecting a
- message leaves an errant .lock file. (closes issue #12528)
- Reported by: pukepail Patches: patch.diff uploaded by pukepail
- (license 431)
-
-2008-04-25 21:54 +0000 [r114673] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Use consistent logic for checking to see if
- a call number has been chosen yet. Also, remove some redundant
- logic I recently added in a fix.
-
-2008-04-25 19:32 +0000 [r114662] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_chanspy.c: Move the unlock of the spyee channel to
- outside the start_spying() function so that the channel is not
- unlocked twice when using whisper mode.
-
-2008-04-25 15:53 +0000 [r114649] Tilghman Lesher <tlesher@digium.com>
-
- * configs/zapata.conf.sample, configs/iax.conf.sample,
- configs/iaxprov.conf.sample, configs/sip.conf.sample: Reference
- documentation files that actually exist. (closes issue #12516)
- Reported by: linuxmaniac Patches: diff_rev114611.patch uploaded
- by linuxmaniac (license 472)
-
-2008-04-24 21:35 +0000 [r114624-114632] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: Re-invite RTP during a masquerade so that,
- for instance, an AMI redirect of two channels which are natively
- bridged will preserve audio on both channels. This prevents a
- problem with Asterisk not re-inviting due to one of the channels
- having being a zombie. (closes issue #12513) Reported by:
- mneuhauser Patches:
- asterisk-1.4-114602_restore-RTP-on-fixup.patch uploaded by
- mneuhauser (license 425)
-
- * apps/app_queue.c: Output of channel variables when
- eventwhencalled=vars was set was being truncated two characters.
- This patch corrects the problem. (closes issue #12493) Reported
- by: davidw
-
- * channels/chan_local.c: Resolve a deadlock in chan_local by
- releasing the channel lock temporarily. (closes issue #11712)
- Reported by: callguy Patches: 11712.patch uploaded by putnopvut
- (license 60) Tested by: acunningham
-
-2008-04-24 19:53 +0000 [r114621] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_local.c: Ensure that when we set the accountcode,
- it actually shows up in the CDR. (Fix for AMI Originate) (Closes
- issue #12007)
-
-2008-04-24 15:55 +0000 [r114608] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix a silly mistake in a change I made
- yesterday that caused chan_iax2 to blow up very quickly. (issue
- #12515)
-
-2008-04-24 14:55 +0000 [r114603] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Only have one max-forwards header in
- outbound REFERs. Discovered in the Asterisk SIP Masterclass in
- Orlando. Thanks Joe!
-
-2008-04-23 22:18 +0000 [r114597-114600] Russell Bryant <russell@digium.com>
-
- * main/http.c: Improve some broken cookie parsing code. Previously,
- manager login over HTTP would only work if the mansession_id
- cookie was first. Now, the code builds a list of all of the
- cookies in the Cookie header. This fixes a problem observed by
- users of the Asterisk GUI. (closes AST-20)
-
- * apps/app_chanspy.c, main/http.c: Fix an issue that caused getting
- the correct next channel to not always work. Also, remove setting
- the amount of time to wait for a digit from 5 seconds back down
- to 1/10 of a second. I believe this was so the beep didn't get
- played over and over really fast, but a while back I put in
- another fix for that issue. (closes issue #12498) Reported by:
- jsmith Patches: app_chanspy_channel_walk.trunk.patch uploaded by
- jsmith (license 15)
-
-2008-04-23 18:28 +0000 [r114594] Jason Parker <jparker@digium.com>
-
- * res/res_musiconhold.c: Fix reload/unload for res_musiconhold
- module. (closes issue #11575) Reported by: sunder Patches:
- M11575_14_rev3.diff uploaded by junky (license 177)
- bug11575_trunk.diff.txt uploaded by jamesgolovich (license 176)
-
-2008-04-23 17:55 +0000 [r114587-114591] Russell Bryant <russell@digium.com>
-
- * main/manager.c, include/asterisk/manager.h: Store the manager
- session ID explicitly as 4 byte ID instead of a ulong. The
- mansession_id cookie is coded to be limited to 8 characters of
- hex, and this could break logins from 64-bit machines in some
- cases. (inspired by AST-20)
-
- * channels/chan_iax2.c: Fix find_callno_locked() to actually return
- the callno locked in some more cases.
-
-2008-04-23 16:51 +0000 [r114584] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Add 502 support for both directions, not
- only one... (see r114571)
-
-2008-04-23 14:54 +0000 [r114579] Joshua Colp <jcolp@digium.com>
-
- * main/pbx.c: Instead of stopping dialplan execution when SayNumber
- attempts to say a large number that it can not print out a
- message informing the user and continue on. (closes issue #12502)
- Reported by: bcnit
-
-2008-04-22 23:51 +0000 [r114571] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_sip.c: Treat a 502 just like a 503, when it comes
- to processing a response code
-
-2008-04-22 22:15 +0000 [r114522-114558] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: When we receive a full frame that is
- supposed to contain our call number, ensure that it has the
- correct one. (closes issue #10078) (AST-2008-006)
-
- * main/rtp.c, main/channel.c, formats/format_pcm.c, main/file.c: I
- thought I was going to be able to leave 1.4 alone, but that was
- not the case. I ran into some problems with G.722 in 1.4, so I
- have merged in all of the fixes in this area that I have made in
- trunk/1.6.0, and things are happy again.
-
- * res/res_musiconhold.c: Trivial change to read the number of
- samples from a frame before calling ast_write()
-
- * res/res_features.c: After a parked call times out, allow the call
- back to the parker to time out. (closes issue #10890)
-
- * channels/chan_iax2.c: If the dial string passed to the call
- channel callback does not indicate an extension, then consider
- the extension on the channel before falling back to the default.
- (closes issue #12479) Reported by: darren1713 Patches:
- exten_dial_fix_chan_iax2.c.patch uploaded by darren1713 (license
- 116)
-
- * channels/chan_sip.c, include/asterisk/sched.h: Merge changes from
- team/russell/issue_9520 These changes make sure that the
- reference count for sip_peer objects properly reflects the fact
- that the peer is sitting in the scheduler for a scheduled
- callback for qualifying peers or for expiring registrations.
- Without this, it was possible for these callbacks to happen at
- the same time that the peer was being destroyed. This was
- especially likely to happen with realtime peers, and for people
- making use of the realtime prune CLI command. (closes issue
- #9520) Reported by: kryptolus Committed patch by me
-
-2008-04-21 14:39 +0000 [r114322] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Only drop audio if we receive it without a
- progress indication. We allow other frames through such as DTMF
- because they may be needed to complete the call. (closes issue
- #12440) Reported by: aragon
-
-2008-04-19 13:57 +0000 [r114297-114299] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_playback.c: Ensure that help text terminates with a
- newline
-
- * res/res_musiconhold.c: MOH usage information needs a terminating
- newline, or else "asterisk -rx 'help moh reload'" will hang.
- Reported via -dev list, fixed by me.
-
-2008-04-18 21:48 +0000 [r114275-114284] Russell Bryant <russell@digium.com>
-
- * main/manager.c: Don't destroy a manager session if poll() returns
- an error of EAGAIN.
-
- * Makefile: ensure directories are created before we try to install
- stuff into them
-
- * Makefile: SUBDIRS_INSTALL is already listed as a subtarget for
- bininstall
-
-2008-04-18 17:44 +0000 [r114257] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_zap.c, main/callerid.c: Clearing up error messages
- so they make a bit more sense. Also removing a redundant error
- message. Issue AST-15
-
-2008-04-18 15:24 +0000 [r114248] Russell Bryant <russell@digium.com>
-
- * channels/chan_agent.c: Ensure that we don't ast_strdupa(NULL)
- (closes issue #12476) Reported by: davidw Patch by me
-
-2008-04-18 13:33 +0000 [r114245] Sean Bright <sean.bright@gmail.com>
-
- * channels/chan_sip.c: Only complete the SIP channel name once for
- 'sip show channel <channel>'
-
-2008-04-18 06:49 +0000 [r114242] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_setcallerid.c: For consistency sake, ensure that the
- values that ${CALLINGPRES} returns are valid as an input to
- SetCallingPres. (Closes issue #12472)
-
-2008-04-17 22:15 +0000 [r114230] Russell Bryant <russell@digium.com>
-
- * main/autoservice.c: Remove redundant safety net. The check for
- the autoservice channel list state accomplishes the same goal in
- a better way. (issue #12470) Reported By: atis
-
-2008-04-17 21:03 +0000 [r114207-114226] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_chanspy.c: Declaration of the peer channel in this scope
- was making it so the peer variable defined in the outer scope was
- never set properly, therefore making iterating through the
- channel list always restart from the beginning. This bug would
- have affected anyone who called chanspy without specifying a
- first argument. (closes issue #12461) Reported by: stever28
-
- * main/frame.c, include/asterisk/dsp.h: Add prototype for
- ast_dsp_frame_freed. I'm not sure how this was compiling
- before...
-
- * main/dsp.c, main/frame.c, include/asterisk/frame.h: It was
- possible for a reference to a frame which was part of a freed DSP
- to still be referenced, leading to memory corruption and eventual
- crashes. This code change ensures that the dsp is freed when we
- are finished with the frame. This change is very similar to a
- change Russell made with translators back a month or so ago.
- (closes issue #11999) Reported by: destiny6628 Patches:
- 11999.patch uploaded by putnopvut (license 60) Tested by:
- destiny6628, victoryure
-
-2008-04-17 16:23 +0000 [r114204] Russell Bryant <russell@digium.com>
-
- * Makefile: Fix the bininstall target to install from subdirs, as
- well. (closes issue AST-8, patch from bmd at switchvox)
-
-2008-04-17 13:42 +0000 [r114198] Philippe Sultan <philippe.sultan@gmail.com>
-
- * res/res_jabber.c: Use keepalives effectively in order diagnose
- bug #12432.
-
-2008-04-17 12:56 +0000 [r114195] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_agi.c: Add special case for when the agi cannot be
- executed, to comply with the documentation that we return failure
- in that case. (closes issue #12462) Reported by: fmueller
- Patches: 20080416__bug12462.diff.txt uploaded by Corydon76
- (license 14) Tested by: fmueller
-
-2008-04-17 10:51 +0000 [r114191] Sean Bright <sean.bright@gmail.com>
-
- * apps/app_chanspy.c: Make sure we have enough room for the
- recording's filename.
-
-2008-04-16 20:46 +0000 [r114184] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c: use the ZT_SET_DIALPARAMS ioctl properly by
- initializing the structure to all zeroes in case it contains
- fields that we don't write values into (which it does as of
- Zaptel 1.4.10) (closes issue #12456) Reported by: fnordian
-
-2008-04-16 19:59 +0000 [r114180] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_vpb.cc: Backport revisions for latest vpb drivers
- to 1.4 (Closes issue #12457)
-
-2008-04-16 17:30 +0000 [r114173] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c: Fix "fallthrough" behavior here, so config
- options in a previously configured user don't override settings
- in general. (closes issue #12458) Reported by: tzafrir Patches:
- chanzap_users_sections.diff uploaded by tzafrir (license 46)
-
-2008-04-16 14:10 +0000 [r114167] Joshua Colp <jcolp@digium.com>
-
- * apps/app_meetme.c: Include the proper headers for using mkdir on
- FreeBSD. (closes issue #12430) Reported by: ys Patches:
- app_meetme.c.diff uploaded by ys (license 281)
-
-2008-04-15 20:26 +0000 [r114148] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Handle subscribe queues in all situations...
- Thanks to festr_ on irc for telling me about this bug.
-
-2008-04-15 17:17 +0000 [r114120-114138] Jason Parker <jparker@digium.com>
-
- * contrib/scripts/autosupport: Update Digium autosupport script,
- for more useful information. (closes issue #12452) Reported by:
- angler Patches: autosupport.diff uploaded by angler (license 106)
-
- * apps/app_queue.c: Allow autofill to work in the general section
- of queues.conf. Additionally, don't try to (re)set options when
- they have empty values in realtime (all unset columns would have
- an empty value). (closes issue #12445) Reported by: atis Patches:
- 12445-autofill.diff uploaded by qwell (license 4)
-
- * channels/chan_h323.c: The call_token on the pvt can occasionally
- be NULL, causing a crash. If it is NULL, we can skip this
- channel, since it can't the one we're looking for. (closes issue
- #9299) Reported by: vazir
-
-2008-04-14 17:41 +0000 [r114106-114117] Mark Michelson <mmichelson@digium.com>
-
- * main/channel.c: Increase the retry count when attempting to show
- channels. This apparently cleared an issue someone was seeing
- when attempting to show channels when the load was high. (closes
- issue #11667) Reported by: falves11 Patches: 11677.txt uploaded
- by russell (license 2) Tested by: falves11
-
- * apps/app_dial.c, apps/app_queue.c: If the datastore has been
- moved to another channel due to a masquerade, then freeing the
- datastore here causes an eventual double free when the new
- channel hangs up. We should only free the datastore if we were
- able to successfully remove it from the channel we are
- referencing (i.e. the datastore was not moved). (closes issue
- #12359) Reported by: pguido
-
- * main/channel.c: Save a local copy of the generate callback prior
- to unlocking the channel in case the generate callback goes NULL
- on us after the channel is unlocked. Thanks to Russell for
- pointing this need out to me.
-
-2008-04-14 14:52 +0000 [r114100-114103] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: It is possible for the remote side to say
- they want T38 but not give any capabilities. (closes issue
- #12414) Reported by: MVF
-
- * main/rtp.c: Don't change the SSRC when a new source comes into
- play, this might happen quite often and depending on the remote
- side... they might not like this. (closes issue #12353) Reported
- by: dimas
-
-2008-04-11 22:32 +0000 [r114083] Terry Wilson <twilson@digium.com>
-
- * channels/chan_iax2.c: Several places in the code called
- find_callno() (which releases the lock on the pvt structure) and
- then immediately locked the call and did things with it.
- Unfortunately, the call can disappear between the find_callno and
- the lock, causing Bad Stuff(tm) to happen. Added
- find_callno_locked() function to return the callno withtout
- unlocking for instances that it is needed. (issue #12400)
- Reported by: ztel
-
-2008-04-11 21:35 +0000 [r114072] Jason Parker <jparker@digium.com>
-
- * main/pbx.c: It's possible that a channel can have an async goto
- on the successful execution of an application as well. Closes
- issue #12172.
-
-2008-04-11 15:44 +0000 [r114045-114063] Mark Michelson <mmichelson@digium.com>
-
- * res/res_features.c: Fix a race condition that may happen between
- a sip hangup and a "core show channel" command. This patch adds
- locking to prevent the resulting crash. (closes issue #12155)
- Reported by: tsearle Patches: show_channels_crash2.patch uploaded
- by tsearle (license 373) Tested by: tsearle
-
- * main/utils.c, include/asterisk/lock.h: Fix 1.4 build when
- LOW_MEMORY is enabled.
-
- * channels/chan_sip.c: Be sure that we're not about to set
- bridgepvt NULL prior to dereferencing it. (closes issue #11775)
- Reported by: fujin
-
-2008-04-10 17:26 +0000 [r114035] Jason Parker <jparker@digium.com>
-
- * main/file.c: Only try to prefix language if we are not using an
- absolute path (suffix it otherwise).
- en/var/lib/asterisk/sounds/blah.gsm is a very silly path. (closes
- issue #12379) Reported by: kuj Patches: 12379-absolutepath.diff
- uploaded by qwell (license 4) Tested by: kuj, qwell
-
-2008-04-10 15:58 +0000 [r114021-114032] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c: Forgot the 1.4 branch for russian language
- fix. (closes issue #12404) Reported by: IgorG Patches:
- voicemail_ru_hardcoded-v1.patch uploaded by IgorG (license 20)
-
- * apps/app_meetme.c: Create the directory where name recordings
- will go if it does not exist. (closes issue #12311) Reported by:
- rkeene Patches: 12311-mkdir.diff uploaded by qwell (license 4)
-
- * channels/chan_sip.c: Don't add custom URI options if they don't
- exist OR they are empty. (closes issue #12407) Reported by:
- homesick Patches: uri_options-1.4.diff uploaded by homesick
- (license 91)
-
-2008-04-09 20:54 +0000 [r113927] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: We need to set the persistant_route [sic]
- parameter for the sip_pvt during the initial INVITE, no matter if
- we're building the route set from an INVITE request or response.
- (closes issue #12391) Reported by: benjaminbohlmann Tested by:
- benjaminbohlmann
-
-2008-04-09 18:57 +0000 [r113874] Tilghman Lesher <tlesher@digium.com>
-
- * cdr/cdr_csv.c, configs/cdr.conf.sample: If the [csv] section does
- not exist in cdr.conf, then an unload/load sequence is needed to
- correct the problem. Track whether the load succeeded with a
- variable, so we can fix this with a simple reload event, instead.
-
-2008-04-09 16:50 +0000 [r113784] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: If we receive an AUTHREQ from the remote
- server and we are unable to reply (for example they have a secret
- configured, but we do not) then queue a hangup frame on the
- Asterisk channel. This will cause the channel to hangup and a
- HANGUP to be sent via IAX2 to the remote side which is the proper
- thing to do in this scenario. (closes issue #12385) Reported by:
- viraptor
-
-2008-04-09 14:40 +0000 [r113681] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: If Asterisk receives a 488 on an INVITE (not
- a reinvite), then we should not send a BYE. (closes issue #12392)
- Reported by: fnordian Patches: chan_sip.patch uploaded by
- fnordian (license 110) with small modification from me
-
-2008-04-09 01:34 +0000 [r113596] Terry Wilson <twilson@digium.com>
-
- * channels/chan_iax2.c: Initialize fr->cacheable to make valgrind
- happy
-
-2008-04-08 19:07 +0000 [r113507] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_parkandannounce.c: Fix potential buffer overflow that
- could happen if more than 100 announce files were specified when
- calling ParkAndAnnounce. This overflow is not exploitable
- remotely and so there is no need for a security advisory. (closes
- issue #12386) Reported by: davidw
-
-2008-04-08 18:48 +0000 [r113402-113504] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Add a little more that is required for
- previously added devices.
-
- * channels/chan_skinny.c: Add support for several new(ish) devices
- - most notably, 7942/7945, 7962/7965, 7975. Thanks to Greg Oliver
- for providing me the required information.
-
- * main/asterisk.c: Work around some silliness caused by
- sys/capability.h - this should fix compile errors a number of
- users have been experiencing.
-
-2008-04-08 16:51 +0000 [r113348-113399] Tilghman Lesher <tlesher@digium.com>
-
- * contrib/scripts/astgenkey.8: Add security note on astgenkey's
- manpage. (closes issue #12373) Reported by: lmamane Patches:
- 20080406__bug12373.diff.txt uploaded by Corydon76 (license 14)
-
- * channels/chan_sip.c: Move check for still-bridged channels out a
- little further, to avoid possible deadlocks. (Closes issue
- #12252) Reported by: callguy Patches: 20080319__bug12252.diff.txt
- uploaded by Corydon76 (license 14) Tested by: callguy
-
-2008-04-08 15:03 +0000 [r113296] Joshua Colp <jcolp@digium.com>
-
- * include/asterisk/slinfactory.h, main/slinfactory.c,
- main/audiohook.c: If audio suddenly gets fed into one side of a
- channel after a lapse of frames flush the other factory so that
- old audio does not remain in the factory causing the sync code to
- not execute. (closes issue #12296) Reported by: jvandal
-
-2008-04-07 21:34 +0000 [r113240] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_sip.c: (closes issue #12362) [redo of 113012] This
- fixes a for loop (in realtime_peer) to check all the
- ast_variables the loop was intending to test rather than just the
- first one. The change exposed the problem of calling memcpy on a
- NULL pointer, in this case the passed in sockaddr_in struct which
- is now checked.
-
-2008-04-07 18:00 +0000 [r113118] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c, configs/skinny.conf.sample: Allow
- playback with noanswer (and add earlyrtp option). (closes issue
- #9077) Reported by: pj Patches: earlyrtp.diff uploaded by wedhorn
- (license 30) Tested by: pj, qwell, DEA, wedhorn
-
-2008-04-07 17:51 +0000 [r113117] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_strings.c: Force ast_mktime() to check for DST, since
- strptime(3) does not. (Closes issue #12374)
-
-2008-04-07 16:08 +0000 [r113065] Mark Michelson <mmichelson@digium.com>
-
- * main/channel.c: This fix prevents a deadlock that was experienced
- in chan_local. There was deadlock prevention in place in
- chan_local, but it would not work in a specific case because the
- channel was recursively locked. By unlocking the channel prior to
- calling the generator's generate callback in
- ast_read_generator_actions(), we prevent the recursive locking,
- and therefore the deadlock. (closes issue #12307) Reported by:
- callguy Patches: 12307.patch uploaded by putnopvut (license 60)
- Tested by: callguy
-
-2008-04-07 15:16 +0000 [r113012] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_sip.c: (closes issue #12362) (closes issue #12372)
- Reported by: vinsik Tested by: tecnoxarxa This one line change
- makes an if inside a for loop (in realtime_peer) check all the
- ast_variables the loop was intending to test rather than just the
- first one.
-
-2008-04-04 19:26 +0000 [r112766-112820] Philippe Sultan <philippe.sultan@gmail.com>
-
- * channels/chan_gtalk.c: Free newly allocated channel before
- returning
-
- * channels/chan_gtalk.c: Prevent call connections when codecs don't
- match. (closes issue #10604) Reported by: keepitcool Patches:
- branch-1.4-10604-2.diff uploaded by phsultan (license 73) Tested
- by: phsultan
-
-2008-04-04 00:52 +0000 [r112709-112711] Joshua Colp <jcolp@digium.com>
-
- * main/Makefile: Pass in the path to Zaptel for systems that
- install Zaptel headers in a separate location.
-
- * main/asterisk.c: One thing at a time... let's get 1.4 building.
-
-2008-04-03 23:57 +0000 [r112689] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * main/asterisk.c: add a Zaptel timer check to verify the timer is
- responding when Zaptel support is compiled into Asterisk and
- Zaptel drivers are loaded. This will help people not waste their
- valuable time debugging side effects.
-
-2008-04-03 14:32 +0000 [r112393-112599] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_zap.c: Fix the testing of the "res" variable so
- that it is more logically correct and makes the correct warning
- and debug messages print. (closes issue #12361) Reported by:
- one47 Patches: chan_zap_deferred_digit.patch uploaded by one47
- (license 23)
-
- * main/manager.c: Fix a race condition in the manager. It is
- possible that a new manager event could be appended during a
- brief time when the manager is not waiting for input. If an event
- comes during this period, we need to set an indicator that there
- is an event pending so that the manager doesn't attempt to wait
- forever for an event that already happened. (closes issue #12354)
- Reported by: bamby Patches: manager_race_condition.diff uploaded
- by bamby (license 430) (comments added by me)
-
- * apps/app_queue.c: Ensure that there is no timeout if none is
- specified. (closes issue #12349) Reported by: johnlange
-
-2008-04-01 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.19 released.
-
-2008-03-28 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.19-rc4 released.
-
-2008-03-28 16:19 +0000 [r111658] Jason Parker <jparker@digium.com>
-
- * formats/format_wav_gsm.c: The file size of WAV49 does not need to
- be an even number. (closes issue #12128) Reported by: mdu113
- Patches: 12128-noevenlength.diff uploaded by qwell (license 4)
- Tested by: qwell, mdu113
-
-2008-03-28 14:35 +0000 [r111442-111605] Tilghman Lesher <tlesher@digium.com>
-
- * doc/valgrind.txt: Update debugging text, since Valgrind
- eliminated the --log-file-exactly option. (Closes issue #12320)
-
- * main/acl.c: For FreeBSD, at least, the ifa_addr element could be
- NULL. (closes issue #12300) Reported by: festr Patches:
- acl.c.patch uploaded by festr (license 443)
-
-2008-03-27 13:03 +0000 [r111341-111391] Steve Murphy <murf@digium.com>
-
- * apps/app_playback.c, main/pbx.c: These small documentation
- updates made in response to a query in asterisk-users, where a
- user was using Playback, but needed the features of Background,
- and had no idea that Background existed, or that it might provide
- the features he needed. I thought the best way to avert these
- kinds of queries was to provide "See Also" references in all
- three of "Background", "Playback", "WaitExten". Perhaps a project
- to do this with all related apps is in order.
-
- * pbx/pbx_ael.c, include/asterisk/ael_structs.h: (closes issue
- #12302) Reported by: pj Tested by: murf These changes will set a
- channel variable ~~EXTEN~~ just before generating code for a
- switch, with the value of ${EXTEN}. The exten is marked as having
- a switch, and ever after that, till the end of the exten, we
- substitute any ${EXTEN} with ${~~EXTEN~~} instead in application
- arguments; (and the ${EXTEN: also). The reason for this, is that
- because switches are coded using separate extensions to provide
- pattern matching, and jumping to/from these switch extensions
- messes up the ${EXTEN} value, which blows the minds of users.
-
-2008-03-27 00:25 +0000 [r111245-111280] Jason Parker <jparker@digium.com>
-
- * main/frame.c: Put this flag back so we don't change the API.
-
- * main/frame.c: Remove excessive smoother optimization that was
- causing audio glitches (small "pops") after (about 200ms later)
- an "incorrectly" sized frame was received. While it would be very
- nice to keep this as optimized as possible, it makes no sense for
- the smoother to be dropping random bits of audio like this. Isn't
- that the whole point of a smoother? Closes issue #12093.
-
-2008-03-26 19:55 +0000 [r111129] Joshua Colp <jcolp@digium.com>
-
- * contrib/scripts/autosupport: Update autosupport script. (closes
- issue #12310) Reported by: angler Patches: autosupport.diff
- uploaded by angler (license 106)
-
-2008-03-26 19:51 +0000 [r111126] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, UPGRADE.txt: Merged revisions 111125 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r111125 | kpfleming | 2008-03-26 14:49:30 -0500 (Wed, 26 Mar
- 2008) | 2 lines update UPGRADE notes to document usage of the
- script ........
-
-2008-03-26 19:37 +0000 [r111049-111121] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: This code change is made just for
- clarification. It does exactly the same thing as before. It just
- doesn't look as wrong.
-
- * apps/app_voicemail.c: Add a lock to the vm_state structure and
- use the lock around mail_open calls to prevent concurrent access
- of the same mailstream. This, along with trunk's ability to
- configure TCP timeouts for IMAP storage will help to prevent
- crashes and hangs when using voicemail with IMAP storage. (closes
- issue #10487) Reported by: ewilhelmsen
-
-2008-03-26 19:06 +0000 [r111024] Kevin P. Fleming <kpfleming@digium.com>
-
- * codecs/ilbc, /, contrib/scripts/get_ilbc_source.sh (added):
- Merged revisions 111019 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r111019 | kpfleming | 2008-03-26 13:58:37 -0500 (Wed, 26 Mar
- 2008) | 2 lines add a script to make getting the iLBC source code
- simple for end users ........
-
-2008-03-26 19:04 +0000 [r111014-111020] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: If we are requested to authenticate a
- reinvite make sure that it contains T38 SDP if need be. (closes
- issue #11995) Reported by: fall
-
- * channels/chan_iax2.c: Make sure that full video frames are sent
- whenever the 15 bit timestamp rolls over. (closes issue #11923)
- Reported by: mihai Patches: asterisk-fullvideo.patch uploaded by
- mihai (license 94)
-
-2008-03-26 17:43 +0000 [r110880-110962] Kevin P. Fleming <kpfleming@digium.com>
-
- * UPGRADE.txt: add note that the user will need to enable
- codec_ilbc to get it to build
-
- * codecs/ilbc/StateConstructW.h (removed),
- codecs/ilbc/libilbc.vcproj (removed), codecs/ilbc/packing.h
- (removed), codecs/ilbc/getCBvec.c (removed),
- codecs/ilbc/LPCdecode.c (removed), codecs/ilbc/enhancer.c
- (removed), codecs/ilbc/lsf.c (removed), codecs/ilbc/iLBC_encode.c
- (removed), codecs/ilbc/getCBvec.h (removed),
- codecs/ilbc/LPCdecode.h (removed), codecs/ilbc/enhancer.h
- (removed), codecs/ilbc/FrameClassify.c (removed),
- codecs/ilbc/iLBC_define.h (removed), codecs/ilbc/lsf.h (removed),
- codecs/ilbc/iLBC_encode.h (removed), codecs/ilbc/FrameClassify.h
- (removed), codecs/ilbc/helpfun.c (removed), codecs/ilbc/doCPLC.c
- (removed), codecs/ilbc/anaFilter.c (removed),
- codecs/ilbc/helpfun.h (removed), codecs/ilbc/createCB.c
- (removed), codecs/ilbc/doCPLC.h (removed),
- codecs/ilbc/anaFilter.h (removed), UPGRADE.txt,
- codecs/ilbc/iLBC_decode.c (removed), codecs/ilbc/constants.c
- (removed), codecs/ilbc/createCB.h (removed), CHANGES,
- codecs/ilbc/iLBC_decode.h (removed), codecs/ilbc/constants.h
- (removed), codecs/Makefile, codecs/ilbc/iCBSearch.c (removed),
- codecs/ilbc/filter.c (removed), codecs/ilbc/hpInput.c (removed),
- codecs/ilbc/gainquant.c (removed), codecs/ilbc/hpOutput.c
- (removed), codecs/ilbc/iCBSearch.h (removed),
- codecs/ilbc/filter.h (removed), codecs/ilbc/hpInput.h (removed),
- codecs/ilbc/gainquant.h (removed), codecs/ilbc/LPCencode.c
- (removed), codecs/ilbc/hpOutput.h (removed),
- codecs/ilbc/StateSearchW.c (removed), codecs/codec_ilbc.c,
- codecs/ilbc/LPCencode.h (removed), codecs/ilbc/StateSearchW.h
- (removed), codecs/ilbc/iCBConstruct.c (removed),
- codecs/ilbc/syntFilter.c (removed), /, codecs/ilbc/iCBConstruct.h
- (removed), codecs/ilbc/syntFilter.h (removed),
- codecs/ilbc/StateConstructW.c (removed), codecs/ilbc/packing.c
- (removed): Merged revisions 110869 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r110869 | kpfleming | 2008-03-26 08:53:46 -0700 (Wed, 26 Mar
- 2008) | 2 lines due to licensing restrictions, we cannot
- distribute the source code for iLBC encoding and decoding... so
- remove it, and add instructions on how the user can obtain it
- themselves ........
-
-2008-03-25 22:51 +0000 [r110779] Jason Parker <jparker@digium.com>
-
- * cdr/cdr_custom.c: Make file access in cdr_custom similar to
- cdr_csv. Fixes issue #12268. Patch borrowed from r82344
-
-2008-03-25 20:03 +0000 [r110727] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_sip.c: This one line change makes an if inside a
- for loop (in realtime_peer) check all the ast_variables the loop
- was intending to test rather than just the first one.
-
-2008-03-25 15:40 +0000 [r110635] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: When reverting a commit, I accidentally left
- in this bit which was an experiment to see what would happen. It
- passed the compile test, and I didn't notice I had left this
- change in too. So this is a revert of a revert...sort of.
-
-2008-03-25 14:37 +0000 [r110628] Joshua Colp <jcolp@digium.com>
-
- * include/asterisk/options.h, main/asterisk.c, Makefile,
- main/app.c: Add an option (transmit_silence) which transmits
- silence during both Record() and DTMF generation. The reason this
- is an option is that in order to transmit silence we have to
- setup a translation path. This may not be needed/wanted in all
- cases. (closes issue #10058) Reported by: tracinet
-
-2008-03-24 19:17 +0000 [r110618] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: This is a revert for revision 108288. The
- reason is that that revision was not for an actual bug fix per
- se, and so it really should not have been in 1.4 in the first
- place. Plus, people who compile with DO_CRASH are more likely to
- encounter a crash due to this change. While I think the usage of
- DO_CRASH in ast_sched_del is a bit absurd, this sort of change is
- beyond the scope of 1.4 and should be done instead in a developer
- branch based on trunk so that all scheduler functions are fixed
- at once. I also am reverting the change to trunk and 1.6 since
- they also suffer from the DO_CRASH potential. (closes issue
- #12272) Reported by: qq12345
-
-2008-03-24 17:34 +0000 [r110614] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Turn a NOTICE into a DEBUG message.
-
-2008-03-21 14:32 +0000 [r110474] Jason Parker <jparker@digium.com>
-
- * codecs/gsm/Makefile: Don't attempt to do optimizations of gsm on
- mips platforms either. (closes issue #12270) Reported by:
- zandbelt Patches: 026-gsm-mips.patch uploaded by zandbelt
- (license 33)
-
-2008-03-20 23:13 +0000 [r110163-110395] Russell Bryant <russell@digium.com>
-
- * main/autoservice.c: Shorten the ast_waitfor() timeout from 500 ms
- to 50 ms in the autoservice thread. This really should not make a
- difference except in very rare cases. That case would be that all
- of the channels in autoservice are not generating any frames. In
- that case, this change reduces the potential amount of time that
- a thread waits in ast_autoservice_stop() for the autoservice
- thread to wrap back around to the beginning of its loop. (closes
- issue #12266, reported by dimas)
-
- * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions
- 110335 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r110335 | russell | 2008-03-20 16:53:27 -0500 (Thu, 20 Mar 2008)
- | 6 lines Fix some very broken code that was introduced in 1.2.26
- as a part of the security fix. The dnsmgr is not appropriate
- here. The dnsmgr takes a pointer to an address structure that a
- background thread continuously updates. However, in these cases,
- a stack variable was passed. That means that the dnsmgr thread
- would be continuously writing to bogus memory. ........
-
- * apps/app_meetme.c: Fix a bug where when calls on the trunk side
- hang up while on hold, the state is not properly reflected.
- (closes issue #11990, reported by anakaoka, patched by me)
-
-2008-03-19 20:33 +0000 [r110083] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_chanspy.c: Add a missing unlock in the case that memory
- allocation fails in app_chanspy. Thanks to Russell for confirming
- that this was an issue.
-
-2008-03-19 19:11 +0000 [r110019-110035] Joshua Colp <jcolp@digium.com>
-
- * res/res_musiconhold.c: Add sanity checking for position resuming.
- We *have* to make sure that the position does not exceed the
- total number of files present, and we have to make sure that the
- position's filename is the same as previous. These values can
- change if a music class is reloaded and give unpredictable
- behavior. (closes issue #11663) Reported by: junky
-
- * main/rtp.c: Make sure that the mark bit does not incorrectly
- cause video frame timestamps to be calculated as if they are
- audio frames. (closes issue #11429) Reported by: sperreault
- Patches: 11429-frametype.diff uploaded by qwell (license 4)
-
-2008-03-19 17:12 +0000 [r109973] Jason Parker <jparker@digium.com>
-
- * Makefile, build_tools/cflags.xml, build_tools/cflags-devmode.xml
- (added): People report bugs about Asterisk crashing with DO_CRASH
- enabled was getting a little silly... Now we only show certain
- cflags when you run configure with --enable-dev-mode
- (corresponding menuselect change to follow)
-
-2008-03-19 15:41 +0000 [r109908] Steve Murphy <murf@digium.com>
-
- * main/config.c: (closes issue #11442) Reported by: tzafrir
- Patches: 11442.patch uploaded by murf (license 17) Tested by:
- murf I didn't give tzafrir very much time to test this, but if he
- does still have remaining issues, he is welcome to re-open this
- bug, and we'll do what is called for. I reproduced the problem,
- and tested the fix, so I hope I am not jumping by just going
- ahead and committing the fix. The problem was with what file_save
- does with templates; firstly, it tended to print out multiple
- options: [my_category](!)(templateref) instead of
- [my_category](!,templateref) which is fixed by this patch.
- Nextly, the code to suppress output of duplicate declarations
- that would occur because the reader copies inherited declarations
- down the hierarchy, was not working. Thus: [master-template](!)
- mastervar = bar [template](!,master-template) tvar = value
- [cat](template) catvar = val would be rewritten as: ;! ;!
- Automatically generated configuration file ;! Filename:
- experiment.conf (/etc/asterisk/experiment.conf) ;! Generator:
- Manager ;! Creation Date: Tue Mar 18 23:17:46 2008 ;!
- [master-template](!) mastervar = bar
- [template](!,master-template) mastervar = bar tvar = value
- [cat](template) mastervar = bar tvar = value catvar = val This
- has been fixed. Since the config reader 'explodes' inherited vars
- into the category, users may, in certain circumstances, see
- output different from what they originally entered, but it should
- be both correct and equivalent.
-
-2008-03-19 04:06 +0000 [r109763-109838] Russell Bryant <russell@digium.com>
-
- * main/utils.c: Tweak spacing in a recent change because I'm very
- picky.
-
- * apps/app_chanspy.c: Fix one place where the chanspy datastore
- isn't removed from a channel. (issue #12243, reported by atis,
- patch by me)
-
-2008-03-18 20:52 +0000 [r109713] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: This patch makes it so that all queue member
- status changes are handled through device state code. This
- removes several problems people were seeing where their queue
- members would get into an "unknown" state. Huge props go to atis
- on this one since he was the one who found the code section that
- was causing the problem and proposed the solution. I just wrote
- what he suggested :) (closes issue #12127) Reported by: atis
- Patches: 12127v3.patch uploaded by putnopvut (license 60) Tested
- by: atis, jvandal
-
-2008-03-18 19:23 +0000 [r109648] Jason Parker <jparker@digium.com>
-
- * codecs/log2comp.h: Allow codecs that use log2comp (g726) to
- compile correctly on x86 with gcc4 optimizations. (closes issue
- #12253) Reported by: fossil Patches: log2comp.patch uploaded by
- fossil (license 140)
-
-2008-03-18 17:58 +0000 [r109575] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_agent.c: Make sure an agent doesn't try to send
- dtmf to a NULL channel closes issue #12242 Reported by Yourname
-
-2008-03-18 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.19-rc3 released.
-
-2008-03-18 16:25 +0000 [r109482] Terry Wilson <twilson@digium.com>
-
- * include/asterisk/astobj.h: Fix character string being treated ad
- format string
-
-2008-03-18 15:10 +0000 [r109393] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 109391 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r109391 | qwell | 2008-03-18 10:08:41 -0500 (Tue, 18 Mar 2008) |
- 3 lines Do not return with a successful authentication if the
- From header ends up empty. (AST-2008-003) ........
-
-2008-03-18 14:58 +0000 [r109386] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c, channels/chan_sip.c: Put a maximum limit on the
- number of payloads accepted, and also make sure a given payload
- does not exceed our maximum value. (AST-2008-002)
-
-2008-03-18 06:37 +0000 [r109309] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael-test/ael-ntest23 (added),
- pbx/ael/ael-test/ael-ntest23/t1/a.ael (added),
- pbx/ael/ael-test/ael-ntest23/t1/b.ael (added),
- pbx/ael/ael-test/ael-ntest23/t1/c.ael (added),
- pbx/ael/ael-test/ael-ntest23/t2/d.ael (added),
- pbx/ael/ael-test/ael-ntest23/t2/e.ael (added),
- pbx/ael/ael-test/ael-ntest23/t2/f.ael (added),
- pbx/ael/ael-test/ref.ael-ntest23 (added), pbx/ael/ael_lex.c,
- pbx/ael/ael-test/ael-ntest23/t3/g.ael (added),
- pbx/ael/ael-test/ael-ntest23/t3/h.ael (added),
- pbx/ael/ael-test/ael-ntest23/t3/i.ael (added), pbx/ael/ael.flex,
- pbx/ael/ael-test/ael-ntest23/t3/j.ael (added),
- pbx/ael/ael-test/ael-ntest23/qq.ael (added),
- pbx/ael/ael-test/ael-ntest23/t1 (added),
- pbx/ael/ael-test/ael-ntest23/t2 (added),
- pbx/ael/ael-test/ael-ntest23/t3 (added),
- pbx/ael/ael-test/ael-ntest23/extensions.ael (added): (closes
- issue #11903) Reported by: atis Many thanks to atis for spotting
- this problem and reporting it. The fix was to straighten out how
- items are placed on and removed from the file stack. Regressions
- as well as the provided test case helped to straighten out all
- code paths. valgrind was used to make sure all memory allocated
- was freed. Sorry for not solving this earlier. I got distracted.
- Added the ntest23 regression test, which is mainly a copy of
- ntest22, but with a few juicy errors thrown in, to replicate the
- kind of error that atis spotted.
-
-2008-03-17 22:05 +0000 [r109226] Mark Michelson <mmichelson@digium.com>
-
- * main/utils.c: Fix a logic flaw in the code that stores lock info
- which is displayed via the "core show locks" command. The idea
- behind this section of code was to remove the previous lock from
- the list if it was a trylock that had failed. Unfortunately,
- instead of checking the status of the previous lock, we were
- referencing the index immediately following the previous lock in
- the lock_info->locks array. The result of this problem, under the
- right circumstances, was that the lock which we currently in the
- process of attempting to acquire could "overwrite" the previous
- lock which was acquired. While this does not in any way affect
- typical operation, it *could* lead to misleading "core show
- locks" output.
-
-2008-03-17 17:55 +0000 [r109171] Michiel van Baak <michiel@vanbaak.info>
-
- * channels/chan_skinny.c: Update the directory of placed calls on
- skinny phones when dialing a channel that does not provide
- progress (analog ZAP lines) The phone does handle the double
- update on calls to channels that do provide progress and wont
- insert duplicate items (closes issue #12239) Reported by: DEA
- Patches: chan_skinny-call-log.txt uploaded by DEA (license 3)
-
-2008-03-17 16:24 +0000 [r109107] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: 200 OKs in response to a reinvite need to be
- sent reliably. If the remote side does not receive one the dialog
- will be torn down. (closes issue #12208) Reported by: atrash
-
-2008-03-17 15:15 +0000 [r109057] Jason Parker <jparker@digium.com>
-
- * main/file.c: Backport revision 106439 from trunk. I didn't
- realize this was broken in 1.4 as well. Closes issue #12222.
-
-2008-03-17 14:18 +0000 [r109012] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_chanspy.c: Make sure that we release the lock on the
- spyee channel if the spyee or spy has hung up (closes issue
- #12232) Reported by: atis
-
-2008-03-16 21:47 +0000 [r108961] Michiel van Baak <michiel@vanbaak.info>
-
- * main/dial.c: add missing break to case AST_CONTROL_SRCUPDATE
- (closes issue #12228) Reported by: andrew Patches: SRC.patch
- uploaded by andrew (license 240)
-
-2008-03-14 20:09 +0000 [r108792-108796] Russell Bryant <russell@digium.com>
-
- * channels/chan_oss.c: Fix a channel name issue. chan_oss registers
- the "Console" channel type, but it created channels with an "OSS"
- prefix. (closes issue #12194, reported by davidw, patched by me)
-
- * contrib/init.d/rc.suse.asterisk: Update the SuSE init script to
- start networking before asterisk, as well. (closes issue #12200,
- reported by and change suggested by reinerotto)
-
-2008-03-14 16:44 +0000 [r108737] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: Fix a race condition in the SIP packet
- scheduler which could cause a crash. chan_sip uses the scheduler
- API in order to schedule retransmission of reliable packets (such
- as INVITES). If a retransmission of a packet is occurring, then
- the packet is removed from the scheduler and retrans_pkt is
- called. Meanwhile, if a response is received from the packet as
- previously transmitted, then when we ACK the response, we will
- remove the packet from the scheduler and free the packet. The
- problem is that both the ACK function and retrans_pkt attempt to
- acquire the same lock at the beginning of the function call. This
- means that if the ACK function acquires the lock first, then it
- will free the packet which retrans_pkt is about to read from and
- write to. The result is a crash. The solution: 1. If the ACK
- function fails to remove the packet from the scheduler and the
- retransmit id of the packet is not -1 (meaning that we have not
- reached the maximum number of retransmissions) then release the
- lock and yield so that retrans_pkt may acquire the lock and
- operate. 2. Make absolutely certain that the ACK function does
- not recursively lock the lock in question. If it does, then
- releasing the lock will do no good, since retrans_pkt will still
- be unable to acquire the lock. (closes issue #12098) Reported by:
- wegbert (closes issue #12089) Reported by: PTorres Patches:
- 12098-putnopvutv3.patch uploaded by putnopvut (license 60) Tested
- by: jvandal
-
-2008-03-14 14:29 +0000 [r108682] Jason Parker <jparker@digium.com>
-
- * res/res_musiconhold.c: Fix a potential segfault if chan (or
- chan->music_state) is NULL. Closes issue #12210, credit to
- edantie for pointing this out.
-
-2008-03-13 21:38 +0000 [r108469-108583] Russell Bryant <russell@digium.com>
-
- * apps/app_chanspy.c, main/channel.c, include/asterisk/channel.h:
- Fix another issue that was causing crashes in chanspy. This
- introduces a new datastore callback, called chan_fixup(). The
- concept is exactly like the fixup callback that is used in the
- channel technology interface. This callback gets called when the
- owning channel changes due to a masquerade. Before this was
- introduced, if a masquerade happened on a channel being spyed on,
- the channel pointer in the datastore became invalid. (closes
- issue #12187) (reported by, and lots of testing from atis) (props
- to file for the help with ideas)
-
- * channels/chan_sip.c: Make a tweak that gets the LEDs on polycom
- phones to blink when an extension that has been subscribed to
- goes on hold. Otherwise, they just stay on like it does when an
- extension is in use. (closes issue #11263) Reported by: russell
- Patches: notify_hold.rev1.txt uploaded by russell (license 2)
- Tested by: russell
-
- * apps/app_followme.c: Fix a couple uses of sprintf. The second one
- could actually cause an overflow of a stack buffer. It's not a
- security issue though, it only depends on your configuration.
-
-2008-03-12 21:53 +0000 [r108227-108288] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: Change AST_SCHED_DEL use to ast_sched_del
- for autocongestion in chan_sip. The scheduler callback will
- always return 0. This means that this id is never rescheduled, so
- it makes no sense to loop trying to delete the id from the
- scheduler queue. If we fail to remove the item from the queue
- once, it will fail every single time. (Yes I realize that in this
- case, the macro would exit early because the id is set to -1 in
- the callback, but it still makes no sense to use that macro in
- favor of calling ast_sched_del once and being done with it) This
- is the first of potentially several such fixes.
-
- * include/asterisk/sched.h: Added a large comment before the
- AST_SCHED_DEL macro to explain its purpose as well as when it is
- appropriate and when it is not appropriate to use it. I also
- removed the part of the debug message that mentions that this is
- probably a bug because there are some perfectly legitimate places
- where ast_sched_del may fail to delete an entry (e.g. when the
- scheduler callback manually reschedules with a new id instead of
- returning non-zero to tell the scheduler to reschedule with the
- same idea). I also raised the debug level of the debug message in
- AST_SCHED_DEL since it seems like it could come up quite
- frequently since the macro is probably being used in several
- places where it shouldn't be. Also removed the redundant line,
- file, and function information since that is provided by ast_log.
-
-2008-03-12 19:57 +0000 [r108135] Russell Bryant <russell@digium.com>
-
- * apps/app_chanspy.c, main/channel.c: (closes issue #12187,
- reported by atis, fixed by me after some brainstorming on the
- issue with mmichelson) - Update copyright info on app_chanspy. -
- Fix a race condition that caused app_chanspy to crash. The issue
- was that the chanspy datastore magic that was used to ensure that
- spyee channels did not disappear out from under the code did not
- completely solve the problem. It was actually possible for
- chanspy to acquire a channel reference out of its datastore to a
- channel that was in the middle of being destroyed. That was
- because datastore destruction in ast_channel_free() was done near
- the end. So, this left the code in app_chanspy accessing a
- channel that was partially, or completely invalid because it was
- in the process of being free'd by another thread. The following
- sort of shows the code path where the race occurred:
- =============================================================================
- Thread 1 (PBX thread for spyee chan) || Thread 2 (chanspy)
- --------------------------------------||-------------------------------------
- ast_channel_free() || - remove channel from channel list || -
- lock/unlock the channel to ensure || that no references retrieved
- from || the channel list exist. ||
- --------------------------------------||-------------------------------------
- || channel_spy() - destroy some channel data || - Lock chanspy
- datastore || - Retrieve reference to channel || - lock channel ||
- - Unlock chanspy datastore
- --------------------------------------||-------------------------------------
- - destroy channel datastores || - call chanspy datastore d'tor ||
- which NULL's out the ds' || - Operate on the channel ...
- reference to the channel || || - free the channel || || || -
- unlock the channel
- --------------------------------------||-------------------------------------
- =============================================================================
-
-2008-03-12 19:16 +0000 [r108086] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_sip.c: if we receive an INVITE with a
- Content-Length that is not a valid number, or is zero, then don't
- process the rest of the message body looking for an SDP closes
- issue #11475 Reported by: andrebarbosa
-
-2008-03-12 18:26 +0000 [r108083] Joshua Colp <jcolp@digium.com>
-
- * apps/app_mixmonitor.c, include/asterisk/audiohook.h,
- main/audiohook.c: Add a trigger mode that triggers on both read
- and write. The actual function that returns the combined audio
- frame though will wait until both sides have fed in audio, or
- until one side stops (such as the case when you call Wait).
- (closes issue #11945) Reported by: xheliox
-
-2008-03-12 16:59 +0000 [r108031] Russell Bryant <russell@digium.com>
-
- * main/channel.c: Destroy the channel lock after the channel
- datastores. (inspired by issue #12187)
-
-2008-03-12 01:52 +0000 [r107877] Tilghman Lesher <tlesher@digium.com>
-
- * contrib/scripts/iax-friends.sql, contrib/scripts/sip-friends.sql:
- Document all of the possible realtime fields
-
-2008-03-11 23:37 +0000 [r107714-107826] Jason Parker <jparker@digium.com>
-
- * doc/voicemail_odbc_postgresql.txt: Update documentation for pgsql
- ODBC voicemail. (closes issue #12186) Reported by: jsmith
- Patches: vm_pgsql_doc_update.patch uploaded by jsmith (license
- 15)
-
- * channels/chan_gtalk.c: Copy voicemail dependency logic for
- res_adsi to chan_gtalk (for jabber). (closes issue #12014)
- Reported by: junky
-
-2008-03-11 20:48 +0000 [r107713] Kevin P. Fleming <kpfleming@digium.com>
-
- * Makefile.rules, channels/Makefile: get chan_vpb to build properly
- in dev mode
-
-2008-03-11 20:47 +0000 [r107712] Jason Parker <jparker@digium.com>
-
- * apps/app_voicemail.c: Add a newline on a log
-
-2008-03-11 19:20 +0000 [r107582-107646] Joshua Colp <jcolp@digium.com>
-
- * res/res_features.c: Make sure the visible indication is on the
- right channel so when the masquerade happens the proper
- indication is enacted. (closes issue #11707) Reported by: iam
-
- * apps/app_meetme.c: Add an additional check for setting conference
- parameter when using the marked user options. It was possible for
- it to return to a no listen/no talk state if a masquerade
- happened. (closes issue #12136) Reported by: aragon
-
- * apps/app_exec.c: Fix a minor spelling error. (closes issue
- #12183) Reported by: darrylc
-
-2008-03-11 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.19-rc2 released.
-
-2008-03-11 15:18 +0000 [r107352-107472] Kevin P. Fleming <kpfleming@digium.com>
-
- * apps/app_rpt.c: backport a fix from trunk
-
- * channels/misdn/isdn_lib.c, codecs/Makefile,
- channels/chan_misdn.c: fix various other problems found by gcc
- 4.3
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac,
- apps/app_sms.c: stop checking for mktime() in the configure
- script... we don't use it, and the test is buggy under gcc 4.3
-
- * configure, main/Makefile, configure.ac, makeopts.in: check for
- compiler support for -fno-strict-overflow before using it (tested
- with Debian's gcc 4.3, 4.1 and 3.4) (closes issue #12179)
- Reported by: Netview
-
- * configure, configure.ac: fix small bug in IMAP toolkit testing
-
- * main/udptl.c, utils/Makefile, main/Makefile,
- main/editline/readline.c, pbx/Makefile: fix up various compiler
- warnings found with gcc-4.3: - the output of flex includes a
- static function called 'input' that is not used, so for the
- moment we'll stop having the compiler tell us about unused
- variables in the flex source files (a better fix would be to
- improve our flex post-processing to remove the unused function) -
- main/stdtime/localtime.c makes assumptions about signed integer
- overflow, and gcc-4.3's improved optimizer tries to take
- advantage of handling potential overflow conditions at compile
- time; for now, suppress these optimizations until we can fiure
- out if the code needs improvement - main/udptl.c has some
- references to uninitialized variables; in one case there was no
- bug, but in the other it was certainly possibly for unexpected
- behavior to occur - main/editline/readline.c had an unused
- variable
-
-2008-03-11 00:59 +0000 [r107290] Terry Wilson <twilson@digium.com>
-
- * channels/chan_sip.c: If we fail to alloc a channel, we should
- re-lock the pvt structure before returning.
-
-2008-03-10 21:32 +0000 [r107230] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c: Use non-global storage for eswitch
-
-2008-03-10 20:27 +0000 [r107173] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c: Make sure to reenable echo can after a
- "failed" (canceled, etc) three-way call. (closes issue #11335)
- Reported by: rebuild
-
-2008-03-10 20:17 +0000 [r107099-107161] Russell Bryant <russell@digium.com>
-
- * main/pbx.c: Fix another bug specifically related to asynchronous
- call origination. Once the PBX is started on the channel using
- ast_pbx_start(), then the ownership of the channel has been
- passed on to another thread. We can no longer access it in this
- code. If the channel gets hung up very quickly, it is possible
- that we could access a channel that has been free'd. (inspired by
- BE-386)
-
- * main/pbx.c: Fix some bugs related to originating calls. If the
- code failed to start a PBX on the channel (such as if you set a
- call limit based on the system's load average), then there were
- cases where a channel that has already been free'd using
- ast_hangup() got accessed. This caused weird memory corruption
- and crashes to occur. (fixes issue BE-386) (much debugging credit
- goes to twilson, final patch written by me)
-
- * main/channel.c: Resolve a compiler warning.
-
- * main/channel.c: Fix a race condition where the generator can go
- away (closes issue #12175, reported by edantie, patched by me)
-
-2008-03-10 14:33 +0000 [r107016] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c, main/cdr.c, include/asterisk/cdr.h: Move where
- unanswered CDRs are dropped to the CDR core, not everything uses
- app_dial. (closes issue #11516) Reported by: ys Patches:
- branch_1.4_cdr.diff uploaded by ys (license 281) Tested by:
- anest, jcapp, dartvader
-
-2008-03-08 15:59 +0000 [r106945] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c: don't generate D-Channel "up" and "down"
- messages unless the channel state is actually changing; also,
- generate the "up" message when an implicit "up" occurs due to
- reception of a normal event when we thought the channel was
- "down"
-
-2008-03-07 22:51 +0000 [r106895] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c: Only start the SLA thread if SLA has actually
- been configured.
-
-2008-03-07 22:14 +0000 [r106842] Jason Parker <jparker@digium.com>
-
- * main/editline/Makefile.in: Fix hardcoded grep in editline, were
- GNU grep is required. (closes issue #12124) Reported by: dmartin
-
-2008-03-07 19:32 +0000 [r106788] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: Ignore source update control frame. (closes issue
- #12168) Reported by: plack
-
-2008-03-07 17:16 +0000 [r106704] Russell Bryant <russell@digium.com>
-
- * include/asterisk/sched.h: Change a warning message to a debug
- message. This is happening quite frequently, and it is not worth
- spamming users with these messages unless we are pretty confident
- that it should never happen. As it stands today, it _will_ and
- _does_ happen and until that gets cleaned up a reasonable amount
- on the development side, let's not spam the logs of everyone
- else. (closes issue #12154)
-
-2008-03-07 16:22 +0000 [r106552-106635] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: Warn the user when a temporary greeting
- exists (Closes issue #11409)
-
- * main/rtp.c: Properly initialize rtp->schedid (Closes issue
- #12154)
-
- * apps/app_chanspy.c, apps/app_rpt.c, main/asterisk.c,
- apps/app_speech_utils.c, apps/app_voicemail.c, main/channel.c,
- funcs/func_enum.c, channels/chan_misdn.c, main/frame.c,
- main/manager.c: Safely use the strncat() function. (closes issue
- #11958) Reported by: norman Patches: 20080209__bug11958.diff.txt
- uploaded by Corydon76 (license 14)
-
-2008-03-06 22:10 +0000 [r106437] Mark Michelson <mmichelson@digium.com>
-
- * main/pbx.c: Quell an annoying message that is likely to print
- every single time that ast_pbx_outgoing_app is called. The reason
- is that __ast_request_and_dial allocates the cdr for the channel,
- so it should be expected that the channel will have a cdr on it.
- Thanks to joetester on IRC for pointing this out
-
-2008-03-06 04:40 +0000 [r106328] Tilghman Lesher <tlesher@digium.com>
-
- * sounds/Makefile: Upgrade to the next release of sounds
-
-2008-03-05 22:37 +0000 [r106237] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix a potential deadlock and a few
- different potential crashes. (closes issue #12145, reported by
- thiagarcia, patched by me)
-
-2008-03-05 22:32 +0000 [r106235] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_oss.c, main/rtp.c, channels/chan_mgcp.c,
- apps/app_dial.c, main/channel.c, channels/chan_phone.c,
- main/dial.c, channels/chan_zap.c, channels/chan_sip.c,
- channels/chan_skinny.c, channels/chan_h323.c, main/file.c,
- channels/chan_alsa.c, apps/app_followme.c,
- include/asterisk/frame.h: Add a control frame to indicate the
- source of media has changed. Depending on the underlying
- technology it may need to change some things. (closes issue
- #12148) Reported by: jcomellas
-
-2008-03-05 21:12 +0000 [r106178] Michiel van Baak <michiel@vanbaak.info>
-
- * doc/realtime.txt: document var_metric so no bugreports will come
- in when it's actually a configuration issue. (issue #12151)
- Reported and patched by: caio1982 1.4 patch by me
-
-2008-03-05 15:32 +0000 [r106038] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c: when a PRI call must be moved to a different
- B channel at the request of the other endpoint, ensure that any
- DSP active on the original channel is moved to the new one
- (closes issue #11917) Reported by: mavetju Tested by: mavetju
-
-2008-03-05 15:17 +0000 [r106015] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_sip.c, include/asterisk/sched.h: Correctly
- initialize retransid in SIP, and ensure that the warning when
- failing to delete a schedule entry can actually hit the log.
- (closes issue #12140) Reported by: slavon Patches: sch2.patch
- uploaded by slavon (license 288) (Patch slightly modified by me)
-
-2008-03-05 01:52 +0000 [r105932] Russell Bryant <russell@digium.com>
-
- * main/rtp.c, main/translate.c, include/asterisk/frame.h: Fix a bug
- that I just noticed in the RTP code. The calculation for setting
- the len field in an ast_frame of audio was wrong when G.722 is in
- use. The len field represents the number of ms of audio that the
- frame contains. It would have set the value to be twice what it
- should be.
-
-2008-03-04 18:10 +0000 [r105674-105676] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: In addition to setting the marker bit let's change
- our ssrc so they know for sure it is a different source.
-
- * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: When a
- new source of audio comes in (such as music on hold) make sure
- the marker bit gets set. (closes issue #10355) Reported by:
- wdecarne Patches: 10355.diff uploaded by file (license 11)
- (closes issue #11491) Reported by: kanderson
-
-2008-03-04 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.19-rc1 released.
-
-2008-03-04 04:31 +0000 [r105591] Russell Bryant <russell@digium.com>
-
- * main/pbx.c: Backport a minor bug fix from trunk that I found
- while doing random code cleanup. Properly break out of the loop
- when a context isn't found when verify that includes are valid.
-
-2008-03-03 18:06 +0000 [r105572] Jason Parker <jparker@digium.com>
-
- * res/snmp/agent.c: Fix type for astNumChannels. (closes issue
- #12114) Reported by: jeffg Patches: 12114.patch uploaded by jeffg
- (license 192)
-
-2008-03-03 17:16 +0000 [r105563-105570] Russell Bryant <russell@digium.com>
-
- * channels/chan_local.c: In the case of an ast_channel allocation
- failure, take the local_pvt out of the pvt list before destroying
- it.
-
- * channels/chan_local.c: Fix a potential memory leak of the
- local_pvt struct when ast_channel allocation fails. Also, in
- passing, centralize the code necessary to destroy a local_pvt.
-
- * main/autoservice.c: Update the copyright information for
- autoservice. Most of the code in this file now is stuff that I
- have written recently ...
-
- * main/asterisk.c, main/channel.c, include/asterisk.h,
- main/autoservice.c: Merge in some changes from
- team/russell/autoservice-nochans-1.4 These changes fix up some
- dubious code that I came across while auditing what happens in
- the autoservice thread when there are no channels currently in
- autoservice. 1) Change it so that autoservice thread doesn't keep
- looping around calling ast_waitfor_n() on 0 channels twice a
- second. Instead, use a thread condition so that the thread
- properly goes to sleep and does not wake up until a channel is
- put into autoservice. This actually fixes an interesting bug, as
- well. If the autoservice thread is already running (almost always
- is the case), then when the thread goes from having 0 channels to
- have 1 channel to autoservice, that channel would have to wait
- for up to 1/2 of a second to have the first frame read from it.
- 2) Fix up the code in ast_waitfor_nandfds() for when it gets
- called with no channels and no fds to poll() on, such as was the
- case with the previous code for the autoservice thread. In this
- case, the code would call alloca(0), and pass the result as the
- first argument to poll(). In this case, the 2nd argument to
- poll() specified that there were no fds, so this invalid pointer
- shouldn't actually get dereferenced, but, this code makes it
- explicit and ensures the pointers are NULL unless we have valid
- data to put there. (related to issue #12116)
-
-2008-03-03 15:28 +0000 [r105557-105560] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: It is possible for no audio to pass between the
- current digit and next digit so expand logic that clears
- emulation to AST_FRAME_NULL. (closes issue #11911) Reported by:
- edgreenberg Patches: v1-11911.patch uploaded by dimas (license
- 88) Tested by: tbsky
-
- * channels/chan_sip.c: Add a comment to describe some logic.
- (closes issue #12120) Reported by: flefoll Patches:
- chan_sip.c.br14.patch-just-a-comment uploaded by flefoll (license
- 244)
-
-2008-02-29 23:34 +0000 [r105409] Russell Bryant <russell@digium.com>
-
- * main/autoservice.c: Fix a major bug in autoservice. There was a
- race condition in the handling of the list of channels in
- autoservice. The problem was that it was possible for a channel
- to get removed from autoservice and destroyed, while the
- autoservice thread was still messing with the channel. This led
- to memory corruption, and caused crashes. This explains multiple
- backtraces I have seen that have references to autoservice, but
- do to the nature of the issue (memory corruption), could cause
- crashes in a number of areas. (fixes the crash in BE-386) (closes
- issue #11694) (closes issue #11940) The following issues could be
- related. If you are the reporter of one of these, please update
- to include this fix and try again. (potentially fixes issue
- #11189) (potentially fixes issue #12107) (potentially fixes issue
- #11573) (potentially fixes issue #12008) (potentially fixes issue
- #11189) (potentially fixes issue #11993) (potentially fixes issue
- #11791)
-
-2008-02-29 14:47 +0000 [r105326] Philippe Sultan <philippe.sultan@gmail.com>
-
- * res/res_jabber.c: Fix a potential memory leak
-
-2008-02-29 14:34 +0000 [r105296] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: If the message file does not exist, just
- return harmlessly, instead of crashing. (Closes issue #12108)
-
-2008-02-29 13:48 +0000 [r105261] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c: Bump up the size of the uniqueid variable.
- (closes issue #12107) Reported by: asgaroth
-
-2008-02-29 13:05 +0000 [r105209] Philippe Sultan <philippe.sultan@gmail.com>
-
- * res/res_jabber.c: Automatically create new buddy upon reception
- of a presence stanza of type subscribed. (closes issue #12066)
- Reported by: ffadaie Patches: branch-1.4-12066-1.diff uploaded by
- phsultan (license 73) trunk-12066-1.diff uploaded by phsultan
- (license 73) Tested by: ffadaie, phsultan
-
-2008-02-28 22:23 +0000 [r105116] Russell Bryant <russell@digium.com>
-
- * main/utils.c, include/asterisk/lock.h: Fix a bug in the lock
- tracking code that was discovered by mmichelson. The issue is
- that if the lock history array was full, then the functions to
- mark a lock as acquired or not would adjust the stats for
- whatever lock is at the end of the array, which may not be
- itself. So, do a sanity check to make sure that we're updating
- lock info for the proper lock. (This explains the bizarre stats
- on lock #63 in BE-396, thanks Mark!)
-
-2008-02-28 21:56 +0000 [r105113] Tilghman Lesher <tlesher@digium.com>
-
- * contrib/init.d/rc.debian.asterisk: Update init script for LSB
- compat (closes issue #9843) Reported by: ibc Patches:
- rc.debian.asterisk.patch uploaded by ibc (license 211) Tested by:
- paravoid
-
-2008-02-28 20:11 +0000 [r105059] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: When using autofill, members who are in use
- should be counted towards the number of available members to call
- if ringinuse is set to yes. Thanks to jmls who brought this issue
- up on IRC
-
-2008-02-28 19:20 +0000 [r104920-105005] Jason Parker <jparker@digium.com>
-
- * main/cdr.c, main/pbx.c: Make pbx_exec pass an empty string into
- applications, if we get NULL. This protects against possible
- segfaults in applications that may try to use data before
- checking length (ast_strdupa'ing it, for example) (closes issue
- #12100) Reported by: foxfire Patches: 12100-nullappargs.diff
- uploaded by qwell (license 4)
-
- * channels/chan_skinny.c: According to a video at www.cisco.com,
- the 7921G supports 6 line appearances.
-
-2008-02-28 00:05 +0000 [r104868] Tilghman Lesher <tlesher@digium.com>
-
- * main/Makefile, build_tools/strip_nonapi: Compatibility fix for
- PPC64 (closes issue #12081) Reported by: jcollie Patches:
- asterisk-1.4.18-funcdesc.patch uploaded by jcollie (license 412)
- Tested by: jcollie, Corydon76
-
-2008-02-27 21:49 +0000 [r104841] Mark Michelson <mmichelson@digium.com>
-
- * main/dial.c: Two fixes: 1. Make the list of ast_dial_channels a
- lockable list. This is because in some cases, the ast_dial may
- exist in multiple threads due to asynchronous execution of its
- application, and I found some cases where race conditions could
- exist. 2. Check in ast_dial_join to be sure that the channel
- still exists before attempting to lock it, since it could have
- gotten hung up but the is_running_app flag on the
- ast_dial_channel may not have been cleared yet. (closes issue
- #12038) Reported by: jvandal Patches: 12038v2.patch uploaded by
- putnopvut (license 60) Tested by: jvandal
-
-2008-02-27 20:56 +0000 [r104787] Joshua Colp <jcolp@digium.com>
-
- * apps/app_chanspy.c: Don't loop around infinitely trying to spy on
- our own channel, and don't forget to free/detach the datastore
- upon hangup of the spy.
-
-2008-02-27 20:36 +0000 [r104783] Mark Michelson <mmichelson@digium.com>
-
- * main/file.c: Bump a couple of more buffers up by 2 so that
- annoying warnings aren't generated like crazy on every
- fileexists_core call.
-
-2008-02-27 18:15 +0000 [r104704] Tilghman Lesher <tlesher@digium.com>
-
- * main/manager.c: Ensure the session ID can't be 0.
-
-2008-02-27 17:41 +0000 [r104665] Joshua Colp <jcolp@digium.com>
-
- * main/file.c: Bump up the buffer by 2.
-
-2008-02-27 17:33 +0000 [r104625] Russell Bryant <russell@digium.com>
-
- * apps/app_chanspy.c: Fix a problem in ChanSpy where it could get
- stuck in an infinite loop without being able to detect that the
- calling channel hung up. (closes issue #12076, reported by junky,
- patched by me)
-
-2008-02-27 17:26 +0000 [r104598] Jason Parker <jparker@digium.com>
-
- * res/res_features.c: Inherit language from the transfering channel
- on a blind transfer. (closes issue #11682) Reported by: caio1982
- Patches: local_atxfer_lang3-1.4.diff uploaded by caio1982
- (license 22) Tested by: caio1982, victoryure
-
-2008-02-27 17:07 +0000 [r104596] Joshua Colp <jcolp@digium.com>
-
- * main/loader.c: Use the lock (which already existed, it just
- wasn't used) on the updaters list to protect the contents instead
- of the overall module list lock. (closes issue #12080) Reported
- by: ChaseVenters
-
-2008-02-27 16:53 +0000 [r104593] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/file.c: fallback to standard English prompts properly when
- using new prompt directory layout (closes issue #11831) Reported
- by: IgorG Patches: fallbacken.v1.diff uploaded by IgorG (license
- 20) (modified by me to improve code and conform rest of function
- to coding guidelines)
-
-2008-02-27 16:45 +0000 [r104591] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c: When we receive a known alarm, make sure
- that the unknown alarm flag is not still set to make sure that
- when we come back out of alarm, it gets reported in the log and
- manager interface (after discussion with tzafrir on the -dev
- list)
-
-2008-02-27 15:52 +0000 [r104536] Joshua Colp <jcolp@digium.com>
-
- * res/res_smdi.c: Only stop the MWI monitor thread if it was
- actually started. (closes issue #12086) Reported by: francesco_r
-
-2008-02-27 01:15 +0000 [r104332-104334] Russell Bryant <russell@digium.com>
-
- * apps/app_chanspy.c: Avoid some recursion in the cleanup code for
- the chanspy datastore (closes issue #12076, reported by junky,
- patched by me)
-
- * channels/chan_zap.c: Zaptel 1.4 now exposes FXO battery state as
- an alarm. However, Asterisk 1.4 does not know what to do with
- these alarms. Only Asterisk 1.6 cares about it. So, if we get an
- unknown alarm in chan_zap, don't generate confusing log messages
- about it.
-
-2008-02-26 18:26 +0000 [r104132-104141] Jason Parker <jparker@digium.com>
-
- * Makefile: Add badshell to .PHONY target (thanks Kevin)
-
- * Makefile: Since all shells aren't as awesome as bash, we have to
- fail if somebody tries to use a literal "~" in DESTDIR.
-
- * sounds/Makefile: Revert previous abspath change. ...abspath is
- new in GNU make 3.81. I feel so...defeated. Must find new fix!
-
- * sounds/Makefile: Fix a very bizarre issue we were seeing with our
- buildbot when using a DESTDIR that wasn't an absolute path (such
- as DESTDIR=~/asterisk-1.4). Apparently what was happening, was
- that some of the targets were being expanded to the full path, so
- $@ ended up being /root/asterisk-1.4/[...]/ rather than
- ~/asterisk-1.4/[...]/ It appears that this may be a new "feature"
- in GNU make. (*cough*
- http://en.wikipedia.org/wiki/Principle_of_least_surprise *cough*)
-
-2008-02-26 00:25 +0000 [r104119] Russell Bryant <russell@digium.com>
-
- * include/asterisk/smdi.h, apps/app_voicemail.c,
- channels/chan_zap.c, res/res_smdi.c, configs/smdi.conf.sample:
- Merge changes from team/russell/smdi-1.4 This commit brings in a
- significant set of changes to the SMDI support in Asterisk. There
- were a number of bugs in the current implementation, most notably
- being that it was very likely on busy systems to pop off the
- wrong message from the SMDI message queue. So, this set of
- changes fixes the issues discovered as well as introducing some
- new ways to use the SMDI support which are required to avoid the
- bugs with grabbing the wrong message off of the queue. This code
- introduces a new interface to SMDI, with two dialplan functions.
- First, you get an SMDI message in the dialplan using
- SMDI_MSG_RETRIEVE() and then you access details in the message
- using the SMDI_MSG() function. A side benefit of this is that it
- now supports more than just chan_zap. For example, with this
- implementation, you can have some FXO lines being terminated on a
- SIP gateway, but the SMDI link in Asterisk. Another issue with
- the current implementation is that it is quite common that the
- station ID that comes in on the SMDI link is not necessarily the
- same as the Asterisk voicemail box. There are now additional
- directives in the smdi.conf configuration file which let you map
- SMDI station IDs to Asterisk voicemail boxes. Yet another issue
- with the current SMDI support was related to MWI reporting over
- the SMDI link. The current code could only report a MWI change
- when the change was made by someone calling into voicemail. If
- the change was made by some other entity (such as with IMAP
- storage, or with a web interface of some kind), then the MWI
- change would never be sent. The SMDI module can now poll for MWI
- changes if configured to do so. This work was inspired by and
- primarily done for the University of Pennsylvania. (also related
- to issue #9260)
-
-2008-02-26 00:03 +0000 [r104111] Jason Parker <jparker@digium.com>
-
- * channels/chan_h323.c: IPTOS_MINCOST is not defined on Solaris.
- (closes issue #12050) Reported by: asgaroth Patches: 12050.patch
- uploaded by putnopvut (license 60)
-
-2008-02-25 23:42 +0000 [r104102-104106] Russell Bryant <russell@digium.com>
-
- * apps/app_chanspy.c: This patch fixes some pretty significant
- problems with how app_chanspy handles pointers to channels that
- are being spied upon. It was very likely that a crash would occur
- if the channel being spied upon hung up. This was because the
- current ast_channel handling _requires_ that the object is locked
- or else it could disappear at any time (except in the owning
- channel thread). So, this patch uses some channel datastore magic
- on the spied upon channel to be able to detect if and when the
- channel goes away. (closes issue #11877) (patch written by me,
- but thanks to kpfleming for the idea, and to file for review)
-
- * main/utils.c: Improve the lock tracking code a bit so that a
- bunch of old locks that threads failed to lock don't sit around
- in the history. When a lock is first locked, this checks to see
- if the last lock in the list was one that was failed to be
- locked. If it is, then that was a lock that we're no longer
- sitting in a trylock loop trying to lock, so just remove it.
- (inspired by issue #11712)
-
-2008-02-25 21:37 +0000 [r104095] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Make it so a users.conf user creates both a
- SIP peer and a SIP user. The user will be used for inbound
- authentication for the device, and peer will be used for placing
- calls to the device. (closes issue #9044) Reported by: queuetue
- Patches: sip-gui-friend.diff uploaded by qwell (license 4)
-
-2008-02-25 21:31 +0000 [r104094] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: If the destination folder is full, don't
- delete a message when exiting. (closes issue #12065) Reported by:
- selsky Patch by: (myself)
-
-2008-02-25 20:49 +0000 [r104092] Jason Parker <jparker@digium.com>
-
- * main/config.c: Allow the use of #include and #exec in situations
- where the max include depth was only 1. Specifically, this fixes
- using #include and #exec in extconfig.conf. This was basically
- caused because the config file itself raises the include level to
- 1. I opted not to raise the include limit, because recursion here
- could cause very bizarre behavior. Pointed out, and tested by
- jmls (closes issue #12064)
-
-2008-02-25 18:38 +0000 [r104086] Russell Bryant <russell@digium.com>
-
- * channels/chan_agent.c: Ensure that the channel doesn't disappear
- in agent_logoff(). If it does, it could cause a crash. (fixes the
- crash reported in BE-396)
-
-2008-02-25 16:16 +0000 [r104082-104084] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: If a resubscription comes in for a dialog we
- no longer know about tell the remote side that the dialog does
- not exist so they subscribe again using a new dialog. (closes
- issue #10727) Reported by: s0l4rb03 Patches: 10727-2.diff
- uploaded by file (license 11)
-
- * channels/chan_sip.c: Due to recent changes tag will no longer be
- NULL if not present so we have to use ast_strlen_zero to see if
- it's actually blank. (closes issue #12061) Reported by: flefoll
- Patches: chan_sip.c.br14.patch_pedantic_no_totag uploaded by
- flefoll (license 244)
-
-2008-02-22 22:45 +0000 [r104037] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_sip.c: Backwards debug message. (closes issue
- #12052) Reported by: flefoll Patches:
- chan_sip.c.br14.patch_found-notfound uploaded by flefoll (license
- 244)
-
-2008-02-21 21:05 +0000 [r104026-104027] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_zap.c: And as a followup to revision 104026,
- completely remove event-related calls from a section of code
- where we know there was no event to handle or get.
-
- * channels/chan_zap.c: Remove an incorrect debug message. It
- reported that it had received a specific event and tried to
- report which event was received. What actually was happening was
- that it was reporting the number of bytes returned from a call to
- read(). Thanks to Jared Smith for bringing the issue up on IRC
-
-2008-02-21 14:33 +0000 [r104015] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/manager.c: reduce the likelihood that HTTP Manager session
- ids will consist of primarily '1' bits
-
-2008-02-20 22:32 +0000 [r103956] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Clear up confusion when viewing the
- QUEUE_WAITING_COUNT of a "dead" realtime queue. Since from the
- user's perspective, the queue does exist, we shouldn't tell them
- we couldn't find the queue. Instead since it is a dead queue,
- report a 0 waiting count This issue was brought up on IRC by jmls
-
-2008-02-20 22:06 +0000 [r103953] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_zap.c: Don't wait for additional digits when
- overlap dialing is enabled if the setup message contains the
- sending_complete information element. (closes issue #11785)
- Reported by: klaus3000 Patches:
- sending_complete_overlap_asterisk-1.4.17.patch.txt uploaded by
- klaus3000 (license 65)
-
-2008-02-20 21:40 +0000 [r103904] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_local.c: Fix a crash if the channel becomes NULL
- while attempting to lock it. (closes issue #12039) Reported by:
- danpwi
-
-2008-02-20 17:53 +0000 [r103845] Tilghman Lesher <tlesher@digium.com>
-
- * main/stdtime/localtime.c: Compat fix for Solaris (closes issue
- #12022) Reported by: asgaroth Patches:
- 20080219__bug12022.diff.txt uploaded by Corydon76 (license 14)
- Tested by: asgaroth
-
-2008-02-19 20:28 +0000 [r103823] Joshua Colp <jcolp@digium.com>
-
- * channels/h323/ast_h323.cxx: Send CallerID Name in setup message.
- (closes issue #11241) Reported by: tusar Patches:
- h323id_as_callerid_name.patch uploaded by tusar (license 344)
-
-2008-02-19 20:02 +0000 [r103821] Russell Bryant <russell@digium.com>
-
- * channels/chan_local.c: Account for the fact that the "other"
- channel can disappear while the local pvt is not locked. (fixes a
- problem introduced in rev 100581) (closes issue #12012) Reported
- by: stevedavies Patch by me
-
-2008-02-19 17:31 +0000 [r103807-103812] Joshua Colp <jcolp@digium.com>
-
- * configure, configure.ac: Don't look for launchd when cross
- compiling. (closes issue #12029) Reported by: ovi
-
- * channels/chan_sip.c: Fix building of chan_sip.
-
-2008-02-19 10:27 +0000 [r103806] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Make sure we send error replies correctly by
- checking the via header.
-
-2008-02-18 23:56 +0000 [r103801] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: Ensure that emulated DTMFs do not get interrupted
- by another begin frame. (closes issue #11740) Reported by: gserra
- Patches: v1-11740.patch uploaded by dimas (license 88) (closes
- issue #11955) Reported by: tsearle (closes issue #10530) Reported
- by: xmarksthespot
-
-2008-02-18 22:28 +0000 [r103790-103795] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c: Fix previous commit so that we actually
- disable echocanbridged if echocancel is off.
-
- * channels/chan_zap.c: Correct a message when echocancelwhenbridged
- is on, but echocancel is not. Issue #12019
-
-2008-02-18 20:52 +0000 [r103786] Mark Michelson <mmichelson@digium.com>
-
- * main/app.c: There was an invalid assumption when calculating the
- duration of a file that the filestream in question was created
- properly. Unfortunately this led to a segfault in the situation
- where an unknown format was specified in voicemail.conf and a
- voicemail was recorded. Now, we first check to be sure that the
- stream was written correctly or else assume a zero duration.
- (closes issue #12021) Reported by: jakep Tested by: putnopvut
-
-2008-02-18 17:31 +0000 [r103780] Tilghman Lesher <tlesher@digium.com>
-
- * main/rtp.c, channels/chan_sip.c: When a SIP channel is being
- auto-destroyed, it's possible for it to still be in bridge code.
- When that happens, we crash. Delay the RTP destruction until the
- bridge is ended. (closes issue #11960) Reported by: norman
- Patches: 20080215__bug11960__2.diff.txt uploaded by Corydon76
- (license 14) Tested by: norman
-
-2008-02-18 16:37 +0000 [r103770] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_zap.c: Fix a linked list corruption that under the
- right circumstances could lead to a looped list, meaning it will
- traverse forever. (closes issue #11818) Reported by: michael-fig
- Patches: 11818.patch uploaded by putnopvut (license 60) Tested
- by: michael-fig
-
-2008-02-18 16:11 +0000 [r103763-103768] Joshua Colp <jcolp@digium.com>
-
- * main/asterisk.c: Backport fix from issue #9325. (closes issue
- #11980) Reported by: rbrunka
-
- * channels/chan_sip.c: Don't care if the extension given doesn't
- exist for subscription based MWI.
-
-2008-02-15 23:31 +0000 [r103726-103741] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix a crash in chan_iax2 due to a race
- condition (closes issue #11780) Reported by: guillecabeza
- Patches: bug_iax2_jb_1.4.patch uploaded by guillecabeza (license
- 380) bug_iax2_jb_trunk.patch uploaded by guillecabeza (license
- 380)
-
- * main/loader.c: In the case that you try to directly reload a
- module has returned AST_MODULE_LOAD_DECLINE, log a message
- indicating that the module is not fully initialized and must be
- initialized using "module load".
-
- * main/loader.c: Don't attempt to execute the reload callback for a
- module that returned AST_MODULE_LOAD_DECLINE. This fixes a crash
- that was reported against chan_console in trunk. (closes issue
- #11953, reported by junky, fixed by me)
-
-2008-02-15 17:26 +0000 [r103688-103722] Mark Michelson <mmichelson@digium.com>
-
- * doc/imapstorage.txt, configure, configure.ac: Final round of
- changes for configure script logic for IMAP Now if a directory is
- specified, then we will search that directory for a source
- installation of the IMAP toolkit. If none is found, then we will
- use that directory as the basis for detecting a package
- installation of the IMAP c-client. If that check fails, then
- configure will fail.
-
- * configure, configure.ac: Fix a bit of wrong logic in the
- configure script that caused problems when trying to configure
- without IMAP. Patch suggestion from phsultan, but I modified it
- slightly. (closes issue #12003) Reported by: pj Tested by:
- putnopvut
-
- * doc/imapstorage.txt, configure, configure.ac: I apparently
- misunderstood one of the requirements of this configure change.
- Now, if a source directory is specified with the --with-imap
- option, and a valid source installation is not detected there,
- then configure will fail and will not check for a package
- installation.
-
- * doc/imapstorage.txt: Make a small clarification in the
- documentation
-
- * doc/imapstorage.txt: Update documentation regarding configuration
- of IMAP
-
- * apps/app_voicemail.c, configure,
- include/asterisk/autoconfig.h.in, configure.ac: Change to the
- configure logic regarding IMAP. Prior to this commit, if you
- wished to configure Asterisk with IMAP support, you would use the
- --with-imap configure switch in one of the following two ways:
- --with-imap=/some/directory would look in the directory specified
- for a UW IMAP source installation --with-imap would assume that
- you had imap-2004g installed in .. relative to the Asterisk
- source With this set of changes the two above options still work
- the same, but there are two new behaviors, too.
- --with-imap=system will assume that you have -libc-client.so
- where you store your shared objects and will attempt to find
- c-client headers in your include path either in the imap or
- c-client directory. If either of the two original methods of
- specifying the imap option should fail, then the check for
- --with-imap =system will be performed in addition. It is only
- after this "system" check that failure can happen.
-
- * apps/app_voicemail.c: Fix build for non-IMAP builds
-
- * apps/app_voicemail.c: Fix the new message count if delete=yes
- when using IMAP storage. (closes issue #11406) Reported by:
- jaroth Patches: deleteflag_v2.patch uploaded by jaroth (license
- 50) Tested by: jaroth
-
-2008-02-14 19:51 +0000 [r103683-103684] Jason Parker <jparker@digium.com>
-
- * funcs/func_cdr.c: swap location for this..
-
- * funcs/func_cdr.c: Document the 'l' option to the CDR() function.
- (Thanks voipgate for pointing out the option, and Leif for
- providing text for it.) Closes issue #11695.
-
-2008-02-13 06:25 +0000 [r103556-103607] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_agent.c: We aren't talking to ourselves; we're
- talking to someone else. (closes issue #11771) Reported by:
- msetim Patches: ami_agent_talkingto-1.4.diff uploaded by caio1982
- (license 22) Tested by: caio1982, msetim
-
- * apps/app_voicemail.c: Refuse to load app_voicemail if res_adsi is
- not loaded (which is a symbol dependency) (closes issue #11760)
- Reported by: non-poster Patches: 20080114__bug11760.diff.txt
- uploaded by Corydon76 (license 14) Tested by: Corydon76,
- non-poster, jamesgolovich
-
-2008-02-12 22:24 +0000 [r103503-103504] Jason Parker <jparker@digium.com>
-
- * main/asterisk.c: revert accidental change from last commit. oops
-
- * contrib/scripts/safe_asterisk, main/asterisk.c: Remove condition
- that was impossible.
-
-2008-02-12 15:09 +0000 [r103324-103385] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Even if no CallerID name or number has been
- provided by the remote party still use the configured sip.conf
- ones. (closes issue #11977) Reported by: pj
-
- * apps/app_meetme.c: If entering a conference with the 'w' option
- ensure that we can't listen or speak until the marked user
- appears. (closes issue #11835) Reported by: alanmcmillan
-
-2008-02-11 17:05 +0000 [r103315] Kevin P. Fleming <kpfleming@digium.com>
-
- * configs/zapata.conf.sample: improve 2BCT documentation a bit
- (thanks Jared)
-
-2008-02-09 06:23 +0000 [r103197] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: Commit fix for being unable to send
- voicemail from VoiceMailMain Reported by: William F Acker (via
- the -users mailing list) Patch by: Corydon76 (license 14)
-
-2008-02-08 18:48 +0000 [r103070-103120] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Prevent a potential three-thread deadlock. Also
- added a comment block to explicitly state the locking order
- necessary inside app_queue. (closes issue #11862) Reported by:
- flujan Patches: 11862.patch uploaded by putnopvut (license 60)
- Tested by: flujan
-
- * channels/chan_iax2.c: Yield the thread and return -1 if the ioctl
- fails for Zaptel timing device. (closes issue #11891) Reported
- by: tzafrir
-
-2008-02-08 15:08 +0000 [r102968] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: Make sure the presence of dbsecret is
- factored into user scoring. (closes issue #11952) Reported by:
- bbhoss
-
-2008-02-07 19:53 +0000 [r102858] Jason Parker <jparker@digium.com>
-
- * res/res_features.c: Specify which digit string was matched in
- debug message. (closes issue #11949) Reported by: dimas Patches:
- v1-feature-debug.patch uploaded by dimas (license 88)
-
-2008-02-07 16:41 +0000 [r102807] Kevin P. Fleming <kpfleming@digium.com>
-
- * configs/zapata.conf.sample: document usage of 'transfer'
- configuration option for ISDN PRI switch-side transfers
-
-2008-02-06 17:59 +0000 [r102653-102725] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Only consider a T.38-only INVITE compatible
- if we have both a joint capability between us and them and if
- they provided T.38.
-
- * main/global_datastores.c: Add missing header file and
- ASTERISK_FILE_VERSION usage. (closes issue #11936) Reported by:
- snuffy
-
-2008-02-06 15:19 +0000 [r102651] Russell Bryant <russell@digium.com>
-
- * configs/features.conf.sample: Clarify setting DYNAMIC_FEATURES so
- that it gets inherited by outbound channels. (due to a discussion
- between me and a user via email)
-
-2008-02-06 11:48 +0000 [r102627] Kevin P. Fleming <kpfleming@digium.com>
-
- * pbx/Makefile, res/Makefile: ensure that all remaining
- multi-object modules are built using their proper CFLAGS and
- include directory paths
-
-2008-02-06 00:26 +0000 [r102576] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: Move around some defines to unbreak ODBC
- storage. (closes issue #11932) Reported by: snuffy
-
-2008-02-05 20:02 +0000 [r102453] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_mgcp.c: Clear the DTMF buffer on hangup. (closes
- issue #11919) Reported by: eferro Patches:
- mgcp_dtmfclean_on_hangup.diff uploaded by eferro (license 337)
- Tested by: eferro
-
-2008-02-05 19:52 +0000 [r102450] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: If a REGISTER attempt comes in that is a
- retransmission of a previous REGISTER do not create a new nonce
- value. (issue #BE-381)
-
-2008-02-05 17:15 +0000 [r102425] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/Makefile: ensure that components of chan_misdn.so are
- built using any special build options that the configure script
- generated (reported by Philipp Kempgen on asterisk-dev)
-
-2008-02-05 15:09 +0000 [r102378] Joshua Colp <jcolp@digium.com>
-
- * res/res_clioriginate.c: Perform dialing asynchronously when using
- the originate CLI command so the CLI does not appear to block.
- (closes issue #11927) Reported by: bbhoss
-
-2008-02-04 21:06 +0000 [r102214-102323] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c, utils/muted.c, configure,
- include/asterisk/autoconfig.h.in, configure.ac: Cross-platform
- fix: OS X now deprecates the use of the daemon(3) API. (closes
- issue #11908) Reported by: oej Patches:
- 20080204__bug11908.diff.txt uploaded by Corydon76 (license 14)
- Tested by: Corydon76
-
- * funcs/func_strings.c: Missing braces. (closes issue #11912)
- Reported by: dimas Patches: sprintf.patch uploaded by dimas
- (license 88)
-
-2008-02-03 16:38 +0000 [r102090-102142] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Use the same CSEQ on CANCEL as on INVITE
- (according to RFC 3261) (closes issue #9492) Reported by:
- kryptolus Patches: bug9492.txt uploaded by oej (license 306)
- Tested by: oej
-
- * channels/chan_sip.c: Handle ACK and CANCEL in an invite
- transaction - even if we get INFO transactions during the actual
- call setup. (closes issue #10567) Reported by: jacksch Tested by:
- oej Patch by: oej inspired by suggestions from neutrino88 in the
- bug tracker
-
-2008-02-01 23:06 +0000 [r101989] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Change the SDP_SAMPLE_RATE macro. It turns
- out that even though G.722 is 16 kHz, it is supposed to specified
- as 8 kHz in the RTP, and RTP timestamps are supposed to be
- calculated based on 8 kHz. (Apparently this is due to a bug in a
- spec, but people follow it anyway, because it's the spec ...)
-
-2008-02-01 21:54 +0000 [r101894-101942] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: Fix the VM_DUR variable for forwarded
- voicemail, and fixed several other bugs while I'm in the area.
- (closes issue #11615) Reported by: jamessan Patches:
- 20071226__bug11615__2.diff.txt uploaded by Corydon76 (license 14)
- Tested by: Corydon76, jamessan
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac,
- acinclude.m4: Change detection of getifaddrs to use
- AST_C_COMPILE_CHECK, backported from trunk (as suggested by
- kpfleming)
-
-2008-02-01 17:41 +0000 [r101822] Jason Parker <jparker@digium.com>
-
- * apps/app_authenticate.c: Remove a needless (and incorrect) call
- to feof() after fgets(). This would have exited the loop early if
- you had an authentication file with no newline at the end.
-
-2008-02-01 17:27 +0000 [r101818-101820] Russell Bryant <russell@digium.com>
-
- * apps/app_authenticate.c: off by one error
-
- * apps/app_authenticate.c: Don't overwrite the last character of a
- line if it's not a newline. This would happen if the last line in
- the file doesn't have a newline. (pointed out by Qwell)
-
-2008-02-01 15:55 +0000 [r101772] Tilghman Lesher <tlesher@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac,
- main/acl.c: Compatibility fix for OpenWRT (reported by Brian
- Capouch via the mailing list)
-
-2008-02-01 00:32 +0000 [r101693] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Add some more sanity checking on IAX2 dial
- strings for the case that no peer or hostname was provided, which
- is the one part of the dial string that is absolutely required.
- If it's not there, bail out. (closes issue #11897) Reported by
- sokhapkin Patch by me
-
-2008-02-01 00:06 +0000 [r101649] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_amd.c: From bugtracker: "fix totalAnalysisTime to handle
- periods of no channel activity" (closes issue #9256) Reported by:
- cmaj Patches: amd-dont-wait-too-long-for-frames-take3.diff.txt
- uploaded by cmaj (license 111) Tested by: cmaj, skygreg, ZX81,
- rjain
-
-2008-01-31 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.18 released.
-
-2008-01-31 23:10 +0000 [r101601] Russell Bryant <russell@digium.com>
-
- * main/translate.c, main/file.c: Fix a couple of places where
- ast_frfree() was not called on a frame that came from a
- translator. This showed itself by g729 decoders not getting
- released. Since the flag inside the translator frame never got
- unset by freeing the frame to indicate it was no longer in use,
- the translators never got destroyed, and thus the g729 licenses
- were not released. (closes issue #11892) Reported by: xrg
- Patches: 11892.diff uploaded by russell (license 2) Tested by:
- xrg, russell
-
-2008-01-31 21:00 +0000 [r101531] Mark Michelson <mmichelson@digium.com>
-
- * res/res_monitor.c: 1. Prevent the addition of an extra '/' to the
- beginning of an absolute pathname. 2. If ast_monitor_change_fname
- is called and the new filename is the same as the old, then exit
- early and don't set the filename_changed field in the monitor
- structure. Setting it in this case was causing ast_monitor_stop
- to erroneously delete them. (closes issue #11741) Reported by:
- garlew Tested by: putnopvut
-
-2008-01-31 19:52 +0000 [r101482] Jason Parker <jparker@digium.com>
-
- * channels/chan_sip.c, channels/chan_iax2.c: Solaris compat fixes
- for struct in_addr funkiness. Issue #11885, patch by snuffy.
-
-2008-01-31 19:30 +0000 [r101480] Steve Murphy <murf@digium.com>
-
- * main/pbx.c: closes issue #11845; that's the one where there's a
- 1004 byte cdr leak with every AMI Redirect to a zap channel
-
-2008-01-31 19:17 +0000 [r101413-101433] Russell Bryant <russell@digium.com>
-
- * channels/chan_agent.c: Add more missing locking of the agents
- list ...
-
- * channels/chan_agent.c: Move the locking from find_agent() into
- the agent dialplan function handler to ensure that the agent
- doesn't disappear while we're looking at it.
-
- * channels/chan_agent.c: Add missing locking to the find_agent()
- function.
-
-2008-01-30 15:41 +0000 [r101222] Joshua Colp <jcolp@digium.com>
-
- * main/slinfactory.c: Fix an issue where if a frame of higher
- sample size preceeded a frame of lower sample size and
- ast_slinfactory_read was called with a sample size of the
- combined values or higher a crash would happen. (closes issue
- #11878) Reported by: stuarth
-
-2008-01-30 15:34 +0000 [r101219] Jason Parker <jparker@digium.com>
-
- * configs/extensions.conf.sample: Change default config to use
- descending channel order of groups, rather than ascending. Fixes
- a potential source of confusion in glare-type situations. Issue
- 11875, reported by JimVanM.
-
-2008-01-30 15:23 +0000 [r101216] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Fix a logic error with regards to autofill.
- Prior to this change, it was possible for a caller to go out of
- turn if autofill were enabled and callers ahead in the queue were
- attempting to call a member. This change fixes this.
-
-2008-01-30 11:20 +0000 [r101152] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Stop musiconhold on attended transfer.
- (closes issue #11872) Reported by: gareth Patches:
- svn-101018.patch uploaded by gareth (license 208)
-
-2008-01-29 23:50 +0000 [r101080] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * build_tools/make_version: updated build_tools to handle the
- autotag directory structure changes; changes related to BE-353.
- Patch by The Russell and reviewed by The Me.
-
-2008-01-29 23:02 +0000 [r100973-101035] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Remove a memory leak from updating realtime
- queues
-
- * apps/app_queue.c: Fixing an erroneous return value returned when
- attempting to pause or unpause a queue member fails. Fixes
- BE-366, thanks to John Bigelow for writing the patch.
-
-2008-01-29 17:57 +0000 [r100934] Joshua Colp <jcolp@digium.com>
-
- * apps/app_mixmonitor.c: Don't forget to record the channel so we
- know whether it is bridged or not later. (closes issue #11811)
- Reported by: slavon
-
-2008-01-29 17:43 +0000 [r100932] Russell Bryant <russell@digium.com>
-
- * main/Makefile: Fix the last couple of issues related to building
- from a path that contains spaces. (closes issue #11834)
-
-2008-01-29 17:41 +0000 [r100930] Jason Parker <jparker@digium.com>
-
- * channels/misdn_config.c: Initialize an array to 0s if config
- option not specified. (closes issue #11860) Patches:
- misdn_get_config.v1.diff uploaded by IgorG (license 20)
-
-2008-01-29 17:21 +0000 [r100882-100922] Russell Bryant <russell@digium.com>
-
- * Makefile: Use GNU make magic instead of shell magic to escape
- spaces in the working directory. (related to issue #11834)
-
- * Makefile: Fix building Asterisk when the working path has spaces
- in it. (closes issue #11834) Reported by: spendergrass Patched
- by: me
-
-2008-01-29 16:10 +0000 [r100835] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c: Allow zap groups above 30 to work properly.
- (closes issue #11590) Reported by: tbsky
-
-2008-01-29 10:36 +0000 [r100793] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c: fixed potential segfault in misdn show
- channels CLI command
-
-2008-01-29 08:26 +0000 [r100740] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: (closes issue #11736) Reported by: MVF
- Patches: bug11736-2.diff uploaded by oej (license 306) Tested by:
- oej, MVF, revolution (russellb: This was the showstopper for the
- release.)
-
-2008-01-28 21:02 +0000 [r100675] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c: WaitExten didn't handle AbsoluteTimeout properly
- (went to 't' instead of 'T')
-
-2008-01-28 20:55 +0000 [r100673] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_vpb.cc, UPGRADE.txt: Undoing the deprecation of
- chan_vpb. It is alive and well.
-
-2008-01-28 20:42 +0000 [r100672] Jason Parker <jparker@digium.com>
-
- * apps/app_voicemail.c: When using ODBC_STORAGE, make sure we put
- greeting files into the database like we do with the others.
- Issue #11795 Reported by: dimas Patches: vmgreet.patch uploaded
- by dimas (license 88)
-
-2008-01-28 18:34 +0000 [r100626-100629] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: For some reason, the use of this strdupa()
- is leading to memory corruption on freebsd sparc64. This trivial
- workaround fixes it. (closes issue #10300, closes issue #11857,
- reported by mattias04 and Home-of-the-Brave)
-
- * res/res_features.c: Fix a crash in ast_masq_park_call() (issue
- #11342) Reported by: DEA Patches: res_features-park.txt uploaded
- by DEA (license 3)
-
-2008-01-28 18:23 +0000 [r100624] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c: Correct a comment which made little/no
- sense.
-
-2008-01-28 17:15 +0000 [r100581] Russell Bryant <russell@digium.com>
-
- * main/channel.c, channels/chan_local.c,
- include/asterisk/channel.h: Make some deadlock related fixes.
- These bugs were discovered and reported internally at Digium by
- Steve Pitts. - Fix up chan_local to ensure that the channel lock
- is held before the local pvt lock. - Don't hold the channel lock
- when executing the timing function, as it can cause a deadlock
- when using chan_local. This actually changes the code back to be
- how it was before the change for issue #10765. But, I added some
- other locking that I think will prevent the problem reported
- there, as well.
-
-2008-01-27 21:59 +0000 [r100465] Tilghman Lesher <tlesher@digium.com>
-
- * main/rtp.c, channels/chan_mgcp.c, main/cdr.c,
- channels/chan_misdn.c, main/dnsmgr.c, channels/chan_sip.c,
- channels/chan_h323.c, include/asterisk/sched.h, main/file.c,
- pbx/pbx_dundi.c, channels/chan_iax2.c: When deleting a task from
- the scheduler, ignoring the return value could possibly cause
- memory to be accessed after it is freed, which causes all sorts
- of random memory corruption. Instead, if a deletion fails, wait a
- bit and try again (noting that another thread could change our
- taskid value). (closes issue #11386) Reported by: flujan Patches:
- 20080124__bug11386.diff.txt uploaded by Corydon76 (license 14)
- Tested by: Corydon76, flujan, stuarth`
-
-2008-01-25 22:32 +0000 [r100418] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_vpb.cc, UPGRADE.txt: Deprecating chan_vpb. It is
- now preferred that users of Voicetronix products use chan_zap in
- combination with their zaptel drivers.
-
-2008-01-25 21:24 +0000 [r100378] Jason Parker <jparker@digium.com>
-
- * channels/chan_sip.c: This would have never been true, since we're
- passing (sizeof(req.data) - 1) as the len to recvfrom().
-
-2008-01-24 21:57 +0000 [r100264] Kevin P. Fleming <kpfleming@digium.com>
-
- * include/asterisk/app.h: make these macros not assume that the
- only other field in the structure is 'argc'... this is true when
- someone uses AST_DECLARE_APP_ARGS, but it's perfectly reasonable
- to define your own structure as long as it has the right fields
-
-2008-01-24 17:22 +0000 [r100164] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c: Update main Asterisk copyright info to 2008
-
-2008-01-24 16:41 +0000 [r100138] Jason Parker <jparker@digium.com>
-
- * main/acl.c: Fix compilation on Solaris. (closes issue #11832)
- Patches: bug-11832.diff uploaded by snuffy (license 35)
-
-2008-01-23 21:07 +0000 [r99977-99978] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Second attempt. Don't change invitestate
- when receiving 18x messages in CANCEL state. (issue #11736)
- Reported by: MVF Patch by oej.
-
- * channels/chan_sip.c: Make sure we don't cancel destruction on
- calls in CANCEL state, even if we get 183 while waiting for
- answer on our CANCEL. (issue #11736) Reported by: MVF Patches:
- bug11736.txt uploaded by oej (license 306) Tested by: MVF
-
-2008-01-23 20:25 +0000 [r99975] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_externalivr.c: Fixing a typo.
-
-2008-01-23 17:46 +0000 [r99923] Russell Bryant <russell@digium.com>
-
- * apps/app_chanspy.c: ChanSpy issues a beep when it starts at the
- beginning of a list of channels to potentially spy on. However,
- if there were no matching channels, it would beep at you over and
- over, which is pretty annoying. Now, it will only beep once in
- the case that there are no channels to spy on, but it will still
- beep again once it reaches the beginning of the channel list
- again. (closes issue #11738, patched by me)
-
-2008-01-23 16:18 +0000 [r99878] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: These flag tests were illogical. They were
- testing sip_peer flags on a sip_pvt. Thanks to Russell for
- helping to get this odd problem figured out.
-
-2008-01-23 04:31 +0000 [r99718-99777] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c: When we reset the password via an external
- command, we should also reset the password stored in the
- in-memory list, too (otherwise it doesn't really take effect).
- (closes issue #11809) Reported by: davetroy Patches:
- fix_externpass.diff uploaded by davetroy (license 384)
-
- * res/res_odbc.c: Oops, should have checked for a NULL obj, here,
- too
-
- * main/acl.c: Just confirmed that all current platforms need this
- header file
-
-2008-01-22 20:56 +0000 [r99652] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Thanks to Russell's education I realize that
- BUFSIZ has changed since I learned the C language over 20 years
- ago... Resetting chan_sip to the size of BUFSIZ that I expected
- in my old head to avoid to heavy memory allocations on some
- systems.
-
-2008-01-22 20:34 +0000 [r99643] Tilghman Lesher <tlesher@digium.com>
-
- * main/acl.c: Fix the defines for OS X (and Solaris, too)
-
-2008-01-22 17:41 +0000 [r99592-99594] Olle Johansson <oej@edvina.net>
-
- * channels/chan_local.c, res/res_features.c, channels/chan_agent.c,
- apps/app_followme.c: Add more dependencies on chan_local and add
- a note to the description of chan_local so that people don't
- disable it in menuselect just to clean up.
-
- * apps/app_dial.c: Add dependency on chan_local to app_dial. Dial
- still runs without chan_local, but will be missing forwarding
- functionality.
-
-2008-01-22 16:54 +0000 [r99540] Tilghman Lesher <tlesher@digium.com>
-
- * main/acl.c: Ensure that we can get an address even when we don't
- have a default route. (closes issue #9225) Reported by: junky
- Patches: 20080122__bug9225.diff.txt uploaded by Corydon76
- (license 14) Tested by: oej, loloski, sergee
-
-2008-01-22 15:08 +0000 [r99501] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Cleaning up some documentation that led to
- confusion in a bug report
-
-2008-01-21 23:55 +0000 [r99426] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_local.c: Fixing an issue wherein monitoring local
- channels was not possible. During a channel masquerade, the
- monitors on the two channels involved are swapped. In 99% of the
- cases this results in the desired effect. However, if monitoring
- a local channel, this caused the monitor which was on the local
- channel to get moved onto a channel which is immediately hung up
- after the masquerade has completed. By swapping the monitors
- prior to the masquerade, we avoid the problem by tricking the
- masquerade into placing the monitor back onto the channel where
- we want it. During the investigation of the issue, the channel's
- monitor was the only thing that was swapped in such a manner
- which did not make sense to have done. All other variable
- swapping made sense.
-
-2008-01-21 18:11 +0000 [r99341] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_odbc.c, configs/res_odbc.conf.sample,
- include/asterisk/res_odbc.h: Permit the user to specify number of
- seconds that a connection may remain idle, which fixes a crash on
- reconnect with the MyODBC driver. (closes issue #11798) Reported
- by: Corydon76 Patches: 20080119__res_odbc__idlecheck.diff.txt
- uploaded by Corydon76 (license 14) Tested by: mvanbaak
-
-2008-01-21 16:01 +0000 [r99301] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Bump the buffer size for Via headers up to
- 512. There are some exceptionally large Via headers out there.
- (closes issue #11783) Reported by: ofirroval
-
-2008-01-19 10:05 +0000 [r99187] Russell Bryant <russell@digium.com>
-
- * main/slinfactory.c: Fix a couple of memory leaks with frame
- handling. Specifically, ast_frame_free() needed to be called on
- the frame that came from the translator to signed linear.
-
-2008-01-18 22:57 +0000 [r99127] Joshua Colp <jcolp@digium.com>
-
- * include/asterisk/channel.h: Remove the __ in front of the unused
- variable. This causes some compilers to freak out.
-
-2008-01-18 21:37 +0000 [r99079-99081] Russell Bryant <russell@digium.com>
-
- * include/asterisk/translate.h, main/frame.c: Revert adding the
- packed attribute, as it really doesn't make sense why that would
- do any good. Fix the real bug, which is to do the check to see if
- the frame came from a translator at the beginning of
- ast_frame_free(), instead of at the end. This ensures that it
- always gets checked, even if none of the parts of the frame are
- malloc'd, and also ensures that we aren't looking at free'd
- memory in the case that it is a malloc'd frame. (closes issue
- #11792, reported by explidous, patched by me)
-
- * include/asterisk/translate.h: Since we're relying on the offset
- between the frame and the beginning of the translator pvt struct,
- set the packed attribute to make sure we get to the right place.
- (potential fix for issue #11792)
-
-2008-01-18 17:13 +0000 [r99032] Terry Wilson <twilson@digium.com>
-
- * res/res_features.c: This should at least temporarily fix a
- problem where the 't' Dial option is incorrectly passed to the
- transferee when built-in attended transfers are used. There is
- still a problem with 'T', but better to fix some problems than no
- problems while we work on it. (closes issue #7904) Reported by:
- k-egg Patches: transfer-fix-b14-r97657.diff uploaded by sergee
- (license 138) Tested by: sergee, otherwiseguy
-
-2008-01-17 23:42 +0000 [r99007-99014] Pari Nannapaneni <paripurnachand@digium.com>
-
- * configs/cdr.conf.sample: doh! revert a revert of a revert
- (changed by mistake in 99010)
-
- * main/manager.c, configs/cdr.conf.sample: missed that one while
- reverting
-
- * main/manager.c: reverting 99001 - We need the Max-Age for
- extending the life of cookie mansession_id
-
-2008-01-17 22:37 +0000 [r99004] Russell Bryant <russell@digium.com>
-
- * main/frame.c, channels/chan_iax2.c, include/asterisk/frame.h:
- Have IAX2 optimize the codec translation path just like chan_sip
- does it. If the caller's codec is in our codec list, move it to
- the top to avoid transcoding. (closes issue #10500) Reported by:
- stevedavies Patches: iax-prefer-current-codec.patch uploaded by
- stevedavies (license 184) iax-prefer-current-codec.1.4.patch
- uploaded by stevedavies (license 184) Tested by: stevedavies, pj,
- sheldonh
-
-2008-01-17 21:31 +0000 [r99001] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/manager.c: we should only send the Set-Cookie header to the
- browser on the first response after creating a manager session,
- not on every response (doing so causes the browser to clear any
- local cookies it may have associated with the session)
-
-2008-01-17 16:19 +0000 [r98991] Jason Parker <jparker@digium.com>
-
- * configs/zapata.conf.sample: Add a clarification about the
- immediate= option of zapata.conf Issue 11784, patch by klaus3000.
-
-2008-01-16 22:36 +0000 [r98982] Russell Bryant <russell@digium.com>
-
- * .cleancount, include/asterisk/channel.h: Add an unused pointer to
- the ast_channel struct. This makes the ast_channel structure
- retain the same size as it had in previous 1.4 releases. Also,
- all of the offsets for members in the structure are still the
- same (except for the two pointers that got replaced for the new
- spy/whisper architecture.)
-
-2008-01-16 20:34 +0000 [r98966-98973] Joshua Colp <jcolp@digium.com>
-
- * .cleancount: Bump up cleancount due to previous commit that
- changed the channel structure.
-
- * apps/app_chanspy.c, apps/app_mixmonitor.c, main/rtp.c,
- main/channel.c, apps/app_meetme.c, include/asterisk/audiohook.h
- (added), main/Makefile, include/asterisk/chanspy.h (removed),
- include/asterisk/channel.h, main/audiohook.c (added): Replace
- current spy architecture with backport of audiohooks. This should
- take care of current known spy issues.
-
- * channels/chan_iax2.c: Add missing NULLs at end of two
- ast_load_realtimes. (closes issue #11769) Reported by: tequ
- Patches: chaniax.patch uploaded by dimas (license 88)
-
-2008-01-16 17:20 +0000 [r98964] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_local.c: Fix a deadlock in chan_local in
- local_hangup. There was contention because the local_pvt was held
- and it was attempting to lock a channel, which is the incorrect
- locking order. (closes issue #11730) Reported by: UDI-Doug
- Patches: 11730.patch uploaded by putnopvut (license 60) Tested
- by: UDI-Doug
-
-2008-01-16 15:08 +0000 [r98951-98960] Joshua Colp <jcolp@digium.com>
-
- * main/dial.c: Introduce a lock into the dialing API that protects
- it when destroying the structure. (closes issue #11687) Reported
- by: callguy Patches: 11687.diff uploaded by file (license 11)
-
- * main/rtp.c: Add two more SDP names for ulaw and alaw. (closes
- issue #11777) Reported by: tootai
-
- * channels/chan_sip.c: Don't drop the old record route information
- when dealing with packets related to a reinvite. (closes issue
- #11545) Reported by: kebl0155 Patches: reinvite-patch.txt
- uploaded by kebl0155 (license 356)
-
- * build_tools/menuselect-deps.in, configure,
- include/asterisk/autoconfig.h.in, codecs/codec_speex.c,
- configure.ac, makeopts.in: Add autoconf logic for speexdsp. Later
- versions use a separate library for some things so we need to use
- it if present in codec_speex. (closes issue #11693) Reported by:
- yzg
-
-2008-01-15 23:50 +0000 [r98943-98946] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Change a buffer in check_auth() to be a
- thread local dynamically allocated buffer, instead of a massive
- buffer on the stack. This fixes a crash reported by Qwell due to
- running out of stack space when building with LOW_MEMORY defined.
- On a very related note, the usage of BUFSIZ in various places in
- chan_sip is arbitrary and careless. BUFSIZ is a system specific
- define. On my machine, it is 8192, but by definition (according
- to google) could be as small as 256. So, this buffer in
- check_auth was 16 kB. We don't even support SIP messages larger
- than 4 kB! Further usage of this define should be avoided, unless
- it is used in the proper context.
-
- * main/rtp.c, include/asterisk/translate.h, main/frame.c,
- main/translate.c, main/abstract_jb.c, channels/chan_iax2.c,
- codecs/codec_zap.c, include/asterisk/frame.h: Commit a fix for
- some memory access errors pointed out by the valgrind2.txt output
- on issue #11698. The issue here is that it is possible for an
- instance of a translator to get destroyed while the frame
- allocated as a part of the translator is still being processed.
- Specifically, this is possible anywhere between a call to
- ast_read() and ast_frame_free(), which is _a lot_ of places in
- the code. The reason this happens is that the channel might get
- masqueraded during this time. During a masquerade, existing
- translation paths get destroyed. So, this patch fixes the issue
- in an API and ABI compatible way. (This one is for you,
- paravoid!) It changes an int in ast_frame to be used as flag
- bits. The 1 bit is still used to indicate that the frame contains
- timing information. Also, a second flag has been added to
- indicate that the frame came from a translator. When a frame with
- this flag gets released and has this flag, a function is called
- in translate.c to let it know that this frame is doing being
- processed. At this point, the flag gets cleared. Also, if the
- translator was requested to be destroyed while its internal frame
- still had this flag set, its destruction has been deffered until
- it finds out that the frame is no longer being processed.
- Admittedly, this feels like a hack. But, it does fix the issue,
- and I was not able to think of a better solution ...
-
-2008-01-15 20:08 +0000 [r98894-98934] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Based on the boundary found move over the
- correct amount. (closes issue #11750) Reported by: tasker
-
- * channels/chan_sip.c: Accept "; boundary=" not just ";boundary="
- in the multipart mixed content type. (closes issue #11750)
- Reported by: tasker
-
-2008-01-14 20:59 +0000 [r98849] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Adding in appropriate unlocks for the locks
- I added. Thanks to joetester on IRC for pointing this out.
-
-2008-01-14 17:38 +0000 [r98774] Russell Bryant <russell@digium.com>
-
- * main/translate.c: Revert a change that introduces an unacceptable
- performance hit and is causing memory leaks ... (from rev 97973)
-
-2008-01-14 16:35 +0000 [r98733-98737] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Fixing another compilation error. I'm a bit off
- today :(
-
- * apps/app_queue.c: Oops. Last commit had compilation error.
-
- * apps/app_queue.c: Adding explicit defaults for missing options to
- init_queue. This is necessary because if a user either removes or
- comments one of these options and reloads their queues, the
- option will not reset to its default, instead maintaining the
- value from prior to the reload. Thanks to John Bigelow for
- pointing this error out to me.
-
-2008-01-12 00:05 +0000 [r98467] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_odbc.c: Add a connection timeout attribute, as that was
- what was intended with the login timeout, but ODBC divides it up
- into 2 different timeouts. (Closes issue #11745)
-
-2008-01-11 22:46 +0000 [r98390] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_dundi.c: Fix up setting the EID on BSD based systems.
- (closes issue #11646) Reported by: caio1982 Patches:
- dundi_osx_eid6.diff.txt uploaded by caio1982 (license 22)
- dundi_osx_eid6-1.4.diff uploaded by caio1982 (license 22) Tested
- by: caio1982, mvanbaak
-
-2008-01-11 21:28 +0000 [r98372] Pari Nannapaneni <paripurnachand@digium.com>
-
- * main/http.c: Comment explaining how to force browser to always
- read some html files from server.
-
-2008-01-11 19:51 +0000 [r98317-98325] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: If the incoming RTP stream changes codec force the
- bridge to break if the other side does not support it. (closes
- issue #11729) Reported by: tsearle Patches:
- new_codec_patch_udiff.patch uploaded by tsearle (license 373)
-
- * res/res_agi.c: If the channel is hungup during RECORD FILE send a
- result code of -1 to be uniform with everything else. (closes
- issue #11743) Reported by: davevg Patches: res_agi.diff uploaded
- by davevg (license 209)
-
-2008-01-11 19:10 +0000 [r98315] Mark Michelson <mmichelson@digium.com>
-
- * main/channel.c: Properly report the hangup cause as no answer
- when someone does not answer (closes issue #10574, reported by
- boch, patched by moy)
-
-2008-01-11 18:25 +0000 [r98266] Tilghman Lesher <tlesher@digium.com>
-
- * codecs/gsm/Makefile: Add another exception (which doesn't work)
- for -march optimization flag. Reported by: thomasmebes Patch by:
- tilghman (Closes issue #11563)
-
-2008-01-11 18:25 +0000 [r98265] Russell Bryant <russell@digium.com>
-
- * doc/security.txt, main/asterisk.c, configure,
- include/asterisk/autoconfig.h.in, main/Makefile, configure.ac,
- makeopts.in: Backport the ability to set the ToS bits on Linux
- when not running as root. Normally, we would not backport
- features into 1.4, but, I was convinced by the justification
- supplied by the supplier of this patch. He pointed out that this
- patch removes a requirement for running as root, thus reducing
- the potential impacts of security issues. (closes issue #11742)
- Reported by: paravoid Patches: libcap.diff uploaded by paravoid
- (license 200)
-
-2008-01-11 17:22 +0000 [r98219] Joshua Colp <jcolp@digium.com>
-
- * apps/app_followme.c: Ensure the return value of ast_bridge_call
- is passed back up as the application return value. This is needed
- for transfers to function so the PBX core knows to continue
- execution. (closes issue #10327) Reported by: kkiely
-
-2008-01-11 15:52 +0000 [r98164] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_sip.c: Back out changes from revision 97077, since
- it wasn't perfect
-
-2008-01-11 03:39 +0000 [r97976-98082] Russell Bryant <russell@digium.com>
-
- * main/frame.c: Fix samples vs. length calculations for g722
-
- * main/translate.c: Simplify this code with a suggestion from Luigi
- on the asterisk-dev list. Instead of using is16kHz(), implement a
- format_rate() function.
-
- * main/translate.c: Fix various timing calculations that made
- assumptions that the audio being processed was at a sample rate
- of 8 kHz.
-
-2008-01-10 23:08 +0000 [r97973] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_sip.c, main/translate.c: 1) When we get a
- translated frame out, clone it, because if the translator pvt is
- freed before we use the frame, bad things happen. 2) Getting a
- failure from ast_sched_delete means that the schedule ID is
- currently running. Don't just ignore it. (Closes issue #11698)
-
-2008-01-10 21:57 +0000 [r97925] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Let us leave a voicemail for ourself if we
- have logged into VoiceMailMain and chosen to leave a message.
- (closes issue #11735, reported and patched by jamessan)
-
-2008-01-10 21:37 +0000 [r97849-97889] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael_lex.c, pbx/Makefile, pbx/ael/ael.flex: Applied the
- same fixes for ael.flex as was done in 97849 for ast_expr2.fl;
- overrode the normally generate yyfree func with our own version
- that checks the pointer for non-null before passing to free().
- Also takes care of a little problem with 2.5.33 and the use of
- the __STDC_VERSION__ macro.
-
- * main/ast_expr2.fl, main/Makefile, main/ast_expr2f.c: This is a
- fix for 2 things: a problem Terry was having in OSX with null
- pointers, which was my fault, as I probably forgot to run the sed
- script last time I made mods. So, I moved the fix into the flex
- input itself. Then, I found when I used flex 2.5.33, that it was
- using __STDC_VERSION__, and that's not real good; so I added back
- in a DIFFERENT sed script to fix that little mess. Tested
- everything, a couple different ways. Hope I did no harm, at the
- least.
-
-2008-01-10 20:12 +0000 [r97847] Jason Parker <jparker@digium.com>
-
- * include/asterisk/frame.h: Fix a comment that is no longer true.
-
-2008-01-10 16:19 +0000 [r97734-97753] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_kdeconsole.h (removed), configs/modules.conf.sample,
- pbx/kdeconsole_main.cc (removed): Remove other remnants of
- pbx_kdeconsole
-
- * pbx/pbx_kdeconsole.cc (removed), build_tools/menuselect-deps.in,
- configure, include/asterisk/autoconfig.h.in, configure.ac,
- makeopts.in: Remove pbx_kdeconsole from the tree. It hasn't
- worked in ages, and nobody has complained. (closes issue #11706,
- reported by caio1982)
-
-2008-01-10 15:07 +0000 [r97697] Joshua Colp <jcolp@digium.com>
-
- * funcs/func_groupcount.c: Don't try to copy the category from the
- group if no category exists. (closes issue #11724) Reported by:
- IgorG Patches: group_count.v1.patch uploaded by IgorG (license
- 20)
-
-2008-01-09 23:01 +0000 [r97640-97645] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_gtkconsole.c: Strip terminal sequences from the verbose
- messages
-
- * pbx/pbx_gtkconsole.c: Make pbx_gtkconsole build ... but doesn't
- actually load on my system still (related to issue #11706)
-
-2008-01-09 20:28 +0000 [r97618-97622] Jason Parker <jparker@digium.com>
-
- * main/cli.c: Correctly display a message if a command could not be
- found. Also fix a comment which may have led to this happening.
- Issue 11718, reported by kshumard.
-
- * main/cli.c: Fix some locking and return value funkiness. We
- really shouldn't be unlocking this lock inside of a function,
- unless we locked it there too.
-
-2008-01-09 18:48 +0000 [r97575] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Part 2 of app_queue doxygen improvements. Some
- smaller functions this time
-
-2008-01-09 18:02 +0000 [r97529] Russell Bryant <russell@digium.com>
-
- * res/res_features.c: Fix saying the parking space number to the
- caller doing the parking ...
-
-2008-01-09 17:21 +0000 [r97491] Kevin P. Fleming <kpfleming@digium.com>
-
- * codecs/codec_zap.c: report the same message whether Zaptel does
- not have transcoder support loaded or no transcoders were found
-
-2008-01-09 16:44 +0000 [r97489] Philippe Sultan <philippe.sultan@gmail.com>
-
- * channels/chan_gtalk.c: Set the caller id within the gtalk_alloc
- function. As underlined in issue #10437 by Josh, we need to
- prevent a possible memory leak. We only set the name part of the
- caller id, the number part is not relevant when dealing with
- JIDs. Closes issue #11549.
-
-2008-01-09 16:11 +0000 [r97450] Joshua Colp <jcolp@digium.com>
-
- * apps/app_meetme.c: Don't do conferencing totally in Zaptel if
- Monitor is running on the channel. (closes issue #11709) Reported
- by: BigJimmy Patches: patch-meetmerec uploaded by BigJimmy
- (license 371)
-
-2008-01-09 15:43 +0000 [r97410-97448] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c: pass the right variable to get an error
- string... oops
-
- * channels/chan_zap.c: add error number output to ioctl failure
- messages to help with debugging
-
-2008-01-09 00:44 +0000 [r97350] Tilghman Lesher <tlesher@digium.com>
-
- * main/cli.c, main/editline/readline.c: Allow filename completion
- on zero-length modules, remove a memory leak, remove a file
- descriptor leak, and make filename completion thread-safe.
- Patched and tested by tilghman. (Closes issue #11681)
-
-2008-01-09 00:17 +0000 [r97206-97308] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: use the \retval doxygen command properly
-
- * apps/app_queue.c: Part 1 of N of adding doxygen comments to
- app_queue. I picked some of the most common functions used (which
- also happen to be some the biggest/ugliest functions too) to
- document first. I'm pretty new to doxygen so criticism is
- welcome.
-
- * apps/app_queue.c: Some coding guidelines-related cleanup
-
-2008-01-08 20:48 +0000 [r97195] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_mgcp.c: Fix various DTMF issues in chan_mgcp.
- (closes issue #11443) Reported by: eferro Patches:
- dtmf_control_hybrid-inband-mode.patch uploaded by eferro (license
- 337)
-
-2008-01-08 20:47 +0000 [r97194] Tilghman Lesher <tlesher@digium.com>
-
- * main/autoservice.c, main/utils.c: Increase constants to where
- we're less likely to hit them while debugging. (Closes issue
- #11694)
-
-2008-01-08 20:42 +0000 [r97192] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Making some changes designed to not allow
- for a corrupted mailstream for a vm_state. 1. Add locking to the
- vm_state retrieval functions so that no linked list corruption
- occurs. 2. Make sure to always grab the persistent vm_state when
- mailstream access is necessary. 3. Correct an incorrect return
- value in the init_mailstream function. (closes issue #11304,
- reported by dwhite)
-
-2008-01-08 19:53 +0000 [r97093-97152] Joshua Colp <jcolp@digium.com>
-
- * funcs/func_groupcount.c: If no group has been provided to the
- GROUP_COUNT dialplan function then use the first one specific to
- the channel. (closes issue #11077) Reported by: m4him
-
- * apps/app_queue.c: Make app_queue calls work with directed pickup.
- (closes issue #11700) Reported by: jbauer
-
-2008-01-08 18:02 +0000 [r97077] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c, channels/chan_sip.c: Apply multiple crash fixes,
- found in issue #11386, but not completely closing that issue.
-
-2008-01-07 20:47 +0000 [r96884-96932] Russell Bryant <russell@digium.com>
-
- * configs/extensions.conf.sample, /: Merged revisions 96931 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r96931 | russell | 2008-01-07 14:46:22 -0600 (Mon, 07 Jan 2008) |
- 2 lines Change misery.digium.com to pbx.digium.com ........
-
- * res/res_smdi.c: Don't crash if something happens when setting up
- an SMDI interface and it gets destroyed before the SMDI port
- handling thread gets created.
-
-2008-01-07 14:34 +0000 [r96797-96815] Philippe Sultan <philippe.sultan@gmail.com>
-
- * res/res_jabber.c: Indentation fix, makes the code easier to read
-
- * res/res_jabber.c: Compute the base64 value over the
- [authzid]\0authcid\0password string, thus excluding the trailing
- NULL byte. This change has already been committed to trunk, see
- #11644.
-
-2008-01-05 02:09 +0000 [r96644] Russell Bryant <russell@digium.com>
-
- * main/devicestate.c: Don't pass an empty string as the device
- name.
-
-2008-01-04 23:03 +0000 [r96575] Tilghman Lesher <tlesher@digium.com>
-
- * main/devicestate.c: Fix the problem of notification of a device
- state change to a device with a '-' in the name. Could probably
- do with a better fix in trunk, but this bug has been open way too
- long without a better solution. Reported by: stevedavies Patch
- by: tilghman (Closes issue #9668)
-
-2008-01-04 22:55 +0000 [r96573] Jason Parker <jparker@digium.com>
-
- * res/res_features.c: Properly continue in the dialplan if using
- PARKINGEXTEN and the slot is full. Issue 11237, patch by me.
-
-2008-01-04 19:27 +0000 [r96525] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_sip.c: If you change the bindaddr in sip.conf to a
- non-bound address and reload, sip goes kablooie. Reported and
- patched by: one47 (Closes issue #11535)
-
-2008-01-04 16:19 +0000 [r96394-96449] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c: Make use of the temporary channel pointer
- while the pvt is unlocked. (closes issue #11675) Reported by:
- flefoll Patches: chan_zap.c.patch-store-owner-before-unlock
- uploaded by flefoll (license 244)
-
- * channels/chan_iax2.c: Don't crash if the iax2 pvt structure has
- been destroyed before we get to this point (closes issue #11672,
- reported by snuffy, patched by me)
-
-2008-01-03 21:37 +0000 [r96318] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_config_pgsql.c: Missed initialization caused crash.
- Reported and fixed by: tiziano (Closes issue #11671)
-
-2008-01-03 12:12 +0000 [r96198-96199] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c: make sure frame is completely clean,
- before we send it to asterisk as DTMF. If we don't make it clean,
- it happens that one way audio occurs..
-
- * channels/chan_misdn.c: when overlapdial was used and no number
- was dialed, the call was dropped, now we just jump into the s
- extension, which makes a lot more sense.
-
-2008-01-02 23:46 +0000 [r96102] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: We need to reset the membername to NULL on each
- iteration of this loop, otherwise the result is that multiple
- members can have the same name, since the variable was not reset
- on each iteration of the loop.
-
-2008-01-02 22:14 +0000 [r96020-96024] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_config.c: Convert locks of the contexts list in
- pbx_config to the appropriate rdlock or wrlock
-
- * pbx/pbx_dundi.c: pbx_dundi only needs a rdlock on the contexts
- list.
-
- * apps/app_macro.c: app_macro only needs a rdlock on the contexts
- list.
-
-2008-01-02 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.17 released.
-
-2008-01-02 20:24 +0000 [r95946] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Allocate a SIP refer structure when
- performing a transfer using BYE with Also so that the transfer
- information is properly stored. (AST-2008-028) (closes issue
- #11637) Reported by: greyvoip
-
-2008-01-02 17:51 +0000 [r95890] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: A change to improve the accuracy of queue
- logging in the case where a member does not answer during the
- specified timeout period. Prior to this change, there was a small
- chance that the member name recorded in this case would be blank.
- Also prior to this change, if using the ringall strategy, if no
- one answered the call during the specified timeout, the member
- name listed in the queue log would randomly be one of the members
- that was rung. (closes issue #11498, reported and tested by
- hloubser, patched by me)
-
-2007-12-31 23:43 +0000 [r95577] Mark Michelson <mmichelson@digium.com>
-
- * main/pbx.c: Avoiding a potentially bad locking situation.
- ast_merge_contexts_and_delete writelocks the conlock, then calls
- ast_hint_extension, which attempts to readlock the same lock.
- Recursion with read-write locks is dangerous, so the inner lock
- needs to be removed. I did this by copying the "guts" of
- ast_hint_extension into ast_merge_contexts_and_delete (sans the
- extra lock). (this change is inspired by the locking problems
- seen in issue #11080, but I have no idea if this is the
- problematic area experienced by the reporters of that issue)
-
-2007-12-31 20:27 +0000 [r95470] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_env.c: Allow the default "0" to be returned if the
- STAT fails (Closes issue #11659)
-
-2007-12-28 18:24 +0000 [r95191] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Remove duplicate increment of the header
- count in the add_header() function. (closes issue #11648)
- Reported by: makoto Patch provided by sergee, committed patch by
- me, inspired by comments from putnopvut
-
-2007-12-28 00:16 +0000 [r95095] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: I found a bug while browsing the queue code and
- managed to reproduce it in a small setup. If a queue uses the
- ringall strategy, it was possible through unfortunate coincidence
- for a single member at a given penalty level to make app_queue
- think that all members at that penalty level were unavailable and
- cause the members at the next penalty level to be rung. With this
- patch, we will only move to the next penalty level if ALL the
- members at a given penalty level are unreachable.
-
-2007-12-27 21:40 +0000 [r95024] Russell Bryant <russell@digium.com>
-
- * main/channel.c: Don't report a syntax error when an empty string
- is passed to ast_get_group. Just return 0. (closes issue #11540)
- Reported by: tzafrir Patches: group_empty.diff uploaded by
- tzafrir (license 46) -- slightly changed by me
-
-2007-12-27 20:09 +0000 [r94977] Mark Michelson <mmichelson@digium.com>
-
- * main/io.c: Fixing a typo in a comment.
-
-2007-12-27 17:32 +0000 [r94905-94924] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_h323.c: Include types.h in chan_h323 as without it
- it can not be compiled on some operating systems like FreeBSD to
- name one. (closes issue #11585) Reported by: sobomax Patches:
- chan_h323.c.diff uploaded by sobomax (license 359)
-
- * channels/chan_sip.c: Use ast_strlen_zero to see if our_contact is
- set or not on the dialog. It is possible for it to be a pointer
- to NULL. (closes issue #11557) Reported by: FuriousGeorge
-
-2007-12-27 15:16 +0000 [r94828-94831] Russell Bryant <russell@digium.com>
-
- * main/pbx.c: Now that the contexts lock is a read/write lock, it
- should not be locked here in ast_hint_state_changed(). This makes
- it get locked recursively which now causes a deadlock. (closes
- issue #11080, thanks to callguy for the access to a deadlocked
- machine)
-
- * include/asterisk/translate.h, main/translate.c: Use the constant
- that I really meant to use here ...
-
- * main/translate.c: Change ast_translator_best_choice() to only pay
- attention to audio formats. This fixes a problem where Asterisk
- claims that a translation path can not be found for channels
- involving video. (closes issue #11638) Reported by: cwhuang
- Tested by: cwhuang Patch suggested by cwhuang, with some
- additional changes by me.
-
-2007-12-27 01:01 +0000 [r94824] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/manager.c: make this comment explain the situation in an
- even more explicit fashion
-
-2007-12-26 20:43 +0000 [r94808] Tilghman Lesher <tlesher@digium.com>
-
- * main/manager.c: Workaround for what is probably a glibc bug (but
- we'll see this crop up again and again, if we don't add the
- workaround). Reported by: rolek Patch by: tilghman (Closes issue
- #11601, closes issue #11426)
-
-2007-12-26 19:04 +0000 [r94789-94801] Russell Bryant <russell@digium.com>
-
- * main/autoservice.c: Just in case the AST_FLAG_END_DTMF_ONLY flag
- was already set before starting autoservice, remember it and
- ensure that the channel has the same setting when autoservice
- gets stopped. (pointed out by d1mas, patched up by me)
-
- * main/autoservice.c: When a channel is in autoservice, mark a flag
- on the channel that says that we only care about the END of a
- digit. That way, no magic digit emulation stuff will happen when
- all we're doing is queueing up END frames.
-
- * res/res_features.c: Don't try to send a parked call back to
- itself. (closes issue #11622, reported by djrodman, patched by
- me)
-
- * main/autoservice.c: Don't store DTMF BEGIN frames while a channel
- is in autoservice. It's just going to make ast_read() do a lot of
- extra work when the channel comes back out of autoservice.
- (closes issue #11628, patched by me)
-
- * Makefile: List include/asterisk/version.h as a .PHONY target
- because we want the commands listed for this target to be
- executed regardless of whether the file exists or not. This fixes
- having the version not up to date when running from svn. (closes
- issue #11619, reported by plack, fixed by me)
-
-2007-12-25 02:27 +0000 [r94769] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: file says... build on the builders.
-
-2007-12-24 19:36 +0000 [r94763-94767] Tilghman Lesher <tlesher@digium.com>
-
- * main/channel.c: Race: we need to wait to queue a NewChannel event
- until after the channel is inserted into the channel list. The
- reason is because some manager users immediately queue requests
- from the channel when they see that event and are confused when
- Asterisk reports no such channel. (Closes issue #11632)
-
- * channels/chan_sip.c: More deadlock avoidance code (this time
- between sip_monitor and sip_hangup) Reported by: apsaras Patch
- by: tilghman (Closes issue #11413)
-
- * channels/chan_sip.c: Another bit of bad logic in realtime_peer
- Reported by: dimas Patch by: dimas (Closes issue #11631)
-
-2007-12-23 01:21 +0000 [r94660] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_sip.c: Argh... I suppose third time's the charm.
-
-2007-12-21 20:21 +0000 [r94468-94543] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Bunch of coding guidelines cleanup
-
- * apps/app_voicemail.c: Better quota support for using IMAP storage
- voicemail (closes issue #11415, reported by jaroth) (closes issue
- #11152, reported by selsky) Patch provided by jaroth
-
- * apps/app_voicemail.c: The mail_copy c-client function does not
- expect a full imap mailbox string, just the name of the mailbox.
- (closes issue #11419, reported and patched by jaroth, with
- additional patchwork from me)
-
- * main/dial.c: Since we are freeing list elements within a list
- traversal, we need to use the safe traversal and remove the item
- from the list before freeing it. (closes issue 11612, reported by
- dtyoo)
-
-2007-12-21 16:37 +0000 [r94466] Russell Bryant <russell@digium.com>
-
- * main/pbx.c, include/asterisk/pbx.h: Convert the contexts lock to
- a read/write lock to resolve a deadlock. This has a nice side
- benefit of improving performance. :) (closes issue #11609)
- (closes issue #11080)
-
-2007-12-21 16:11 +0000 [r94420-94464] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Removing a debug message I accidentally just
- committed
-
- * main/say.c, apps/app_queue.c: Fixing Portuguese syntax for saying
- dates and times. Also some coding guidelines cleanup. (closes
- issue #11599, reported and patched by caio1982, coding guidelines
- cleanup by me)
-
-2007-12-21 15:07 +0000 [r94418] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c: Fix for restart-as-user problem reported via the
- -dev list
-
-2007-12-20 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.16.2 released.
-
-2007-12-20 20:22 +0000 [r94215-94256] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 94255 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r94255 | russell | 2007-12-20 14:21:41 -0600 (Thu, 20 Dec 2007) |
- 5 lines Fix another potential seg fault ... (closes issue #11606)
- Reported by: dimas ........
-
- * channels/chan_zap.c: Fix a deadlock in d-channel handling in
- chan_zap. This deadlock was introduced by the fix to ensure that
- channels are properly locked when handling channel variables.
- There were sections of this code where the channel pvt was locked
- before the channel lock, when in fact it _must_ be the other way
- around. (closes issue #11582) Reported by: bugi
-
-2007-12-19 23:02 +0000 [r94122] Mark Michelson <mmichelson@digium.com>
-
- * res/res_monitor.c: Sox versions 13.0.0 and newer do not have
- "soxmix" and instead use sox -m. res_monitor needs to use this if
- the user does not have soxmix. (closes issue #11589, reported by
- amessina, patch inspired by amessina but with a flourish from me)
-
-2007-12-19 22:48 +0000 [r94077] Russell Bryant <russell@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac: Check
- for the existence of the soxmix application on the target
- platform and have the result available in autoconfig.h. (part of
- issue #11589)
-
-2007-12-19 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.16.1 released.
-
-2007-12-19 17:29 +0000 [r93955] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: Make the 1.4 builders happy, ensure var is
- NULL.
-
-2007-12-19 17:04 +0000 [r93949] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_iax2.c: Avoid segfault in chan_iax when peer isn't
- defined (Closes issue #11602)
-
-2007-12-18 22:42 +0000 [r93764] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: FreeBSD also does not have byte swap
- functions. Issue 11586, patch by sobomax.
-
-2007-12-18 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.16 released.
-
-2007-12-18 18:45 +0000 [r93668-93676] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions
- 93667 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r93667 | tilghman | 2007-12-18 12:23:06 -0600 (Tue, 18 Dec 2007)
- | 2 lines Fixing AST-2007-027 (Closes issue #11119) ........
-
-2007-12-18 17:02 +0000 [r93625] Mark Michelson <mmichelson@digium.com>
-
- * main/channel.c: Rework deadlock avoidance used in ast_write,
- since it meant that agent channels which were being monitored had
- one audio file recorded and one empty audio file saved. (closes
- issue #11529, reported by atis patched by me)
-
-2007-12-17 22:56 +0000 [r93381-93420] Jason Parker <jparker@digium.com>
-
- * main/translate.c: What was I thinking when I wrote this
- masterpiece? -1 + 1 = 0.. who woulda thunk it?.
-
-2007-12-17 22:28 +0000 [r93377] Joshua Colp <jcolp@digium.com>
-
- * main/utils.c: Do not try to access information about a lock when
- printing out a trylock attempt. It is possible for the lock that
- it references to no longer be valid. This would have caused
- segfaults or deadlocks. (issue #BE-263) (closes issue #11080)
- Reported by: callguy (closes issue #11100) Reported by: callguy
-
-2007-12-17 21:12 +0000 [r93336] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/time.h: Today is tomorrow's yesterday, and
- yesterday's tomorrow is today, and tomorrow's tomorrow is the day
- after tomorrow, so who cares if you recycle anyway? If this
- confuses you, that's nothing compared to what this fixes. ;-)
-
-2007-12-17 19:53 +0000 [r93291] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: We need to create the directory for a
- voicemail user even if they are using IMAP storage since
- greetings are stored in the filesystem. (closes issue #11388,
- reported by spditner, patch by me inspired by a patch by
- spditner)
-
-2007-12-17 18:05 +0000 [r93250] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_zap.c: If a call is received with a called number
- IE containing nothing go to the 's' extension. (closes issue
- #9099) Reported by: kb1_kanobe2 Patches: 20070906__9099.diff.txt
- uploaded by Corydon76 (license 14)
-
-2007-12-17 07:21 +0000 [r93183] Kevin P. Fleming <kpfleming@digium.com>
-
- * funcs/Makefile, codecs/Makefile, cdr/Makefile, pbx/Makefile,
- res/Makefile, channels/Makefile, formats/Makefile: fix some
- copy-and-paste leftovers
-
-2007-12-17 07:15 +0000 [r93182] Olle Johansson <oej@edvina.net>
-
- * channels/chan_mgcp.c, channels/chan_zap.c, channels/chan_sip.c,
- apps/app_queue.c, channels/chan_iax2.c: Issue 11574: Add
- dependencies on res_monitor and res_features. I wonder if
- Asterisk can run at all without res_features. My guess is that
- there's propably a lot of more modules and the core that depends
- on it. Reported by: caio1982 (closes issue #11574)
-
-2007-12-17 06:44 +0000 [r93180] Kevin P. Fleming <kpfleming@digium.com>
-
- * formats, Makefile, codecs/Makefile, funcs, apps/Makefile,
- configure, cdr/Makefile, build_tools/prep_tarball, makeopts.in,
- formats/Makefile, pbx, res, channels, funcs/Makefile, codecs,
- include/asterisk/autoconfig.h.in, build_tools/make_version, apps,
- configure.ac, Makefile.moddir_rules, build_tools/prep_moduledeps
- (removed), res/Makefile, pbx/Makefile, cdr, channels/Makefile: In
- http://lists.digium.com/pipermail/asterisk-dev/2007-December/031145.html,
- rizzo brought up some issues related to the way that the metadata
- required for menuselect and the rest of the build system is
- extracted from the source files. Since I had a few hours to kill
- on an airplane today, I decided to improve this situation... so
- now the system caches the extracted metadata and uses it to build
- the menuselect 'tree' as much as it can. The result of this is
- that when a single source file is changed, only the metadata for
- that file needs to be extracted again, and the rest is used from
- the cache files. I also reduced the number of forked processes
- required to do the metadata extraction; it was actually possible
- to do most of what we needed in the Makefiles themselves without
- using any shell scripts at all! On my laptop, these changes
- resulted in an 80% decrease in the time required for the
- 'menuselect.makeopts' automatic check to occur after editing a
- single source file. While doing this work I also cleaned up a few
- minor things in the Makefiles, adding a check for 'awk' to the
- configure script and changed all remaining places we use 'grep'
- or 'awk' to use the ones found by the configure script, and
- changed the 'prep_tarball' script to build the menuselect
- metadata so that tarballs of Asterisk will include it and won't
- require the user to wait while it is extracted after unpacking.
-
-2007-12-14 17:36 +0000 [r93000] Russell Bryant <russell@digium.com>
-
- * main/config.c: There are a lot of existing systems that #include
- non-existent files. So, to make the transition to treating this
- as an error a bit less painless, just issue a huge error message
- for now. Then, later, we can reinstate the code that treats it as
- a failure. (Thanks to philippel for the feedback)
-
-2007-12-14 15:16 +0000 [r92937] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Up the length of the format on the SIP
- channel since it can now be rather long. (closes issue #11552)
- Reported by: francesco_r
-
-2007-12-14 15:05 +0000 [r92934] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c: fixed the sequencing of WAITING_4DIGS
- state setting and overlap_task thread starting.
-
-2007-12-14 15:01 +0000 [r92933] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_agi.c: Change help documentation to match actual behavior
- (FAILURE vs FAILED). Reported by: angeloxx-sir Patch by: tilghman
- (Closes issue #11548)
-
-2007-12-14 01:24 +0000 [r92875] Mark Michelson <mmichelson@digium.com>
-
- * include/asterisk/lock.h: When compiling with DETECT_DEADLOCKS,
- don't spam the CLI with messages about possible deadlocks.
- Instead just print the intended single message every five
- seconds. (closes issue 11537, reported and patched by dimas)
-
-2007-12-13 21:28 +0000 [r92815] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_zap.c: Properly initialize polarity statuses, so
- that they are detected properly. Reported by: julianjm Patch by:
- julianjm (Closes issue #10238)
-
-2007-12-13 20:13 +0000 [r92809] Jason Parker <jparker@digium.com>
-
- * main/pbx.c: Make application help text a little more clear about
- the use of extensions in a filename.
-
-2007-12-13 20:03 +0000 [r92803-92807] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Prevent another potential fd leak
-
- * apps/app_voicemail.c: Prevent a possible fd leak.
-
-2007-12-13 00:11 +0000 [r92696] Jason Parker <jparker@digium.com>
-
- * main/config.c, channels/chan_sip.c, channels/chan_h323.c,
- channels/chan_iax2.c: If a typo is found in a config file, we
- previous continued on with what was already loaded. We do not
- want to do this (see bug below for details). This makes it so
- that if a [ is found without a ], the entire config will fail,
- and nothing in it will be loaded. Isue #10690.
-
-2007-12-12 22:00 +0000 [r92656] Kevin P. Fleming <kpfleming@digium.com>
-
- * codecs/codec_zap.c: emit a warning message when we drop a G.729B
- CNG frame destined for the transcoder
-
-2007-12-12 21:15 +0000 [r92617] Jason Parker <jparker@digium.com>
-
- * apps/app_meetme.c: Don't increment user count until after name
- has been recorded (if enabled). Issue 11048, tested by pep.
-
-2007-12-12 19:40 +0000 [r92556] Russell Bryant <russell@digium.com>
-
- * res/res_features.c: resolve compiler warning
-
-2007-12-12 17:46 +0000 [r92510] Mark Michelson <mmichelson@digium.com>
-
- * res/res_features.c: Correctly detect where a dynamic feature was
- activated. Before this patch, the channel which initiated the
- bridge was always assumed to have been the one which activated
- the dynamic feature. This patch corrects this. (closes issue
- #11529, reported and patched by nic_bellamy)
-
-2007-12-12 16:52 +0000 [r92463] Tilghman Lesher <tlesher@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac: Test
- directly for the API that fixed AST-2007-026, to ensure that
- older versions of PostgreSQL are no longer acceptable. (Closes
- issue #11526)
-
-2007-12-12 16:08 +0000 [r92443] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Removing an unused variable.
-
-2007-12-11 19:51 +0000 [r92363] Joshua Colp <jcolp@digium.com>
-
- * main/global_datastores.c: Fix potential memory leak with the
- dialed interfaces list if another memory allocation fails.
- (closes issue #11507) Reported by: eliel Patches:
- global_datastores.c.patch uploaded by eliel (license 64)
-
-2007-12-11 17:42 +0000 [r92323] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Fixing autofill to be more accurate.
- Specifically, if calls ahead of the current caller were ringing
- members (but not yet bridged) there could be available members
- and waiting callers who would not get matched up. The member
- availability checker was correctly determining the number of
- available members in this scenario, but the queue itself did not
- parallelly reflect this status on the pending calls. This commit
- corrects the issue. (closes issue #11459, reported by
- equissoftware, patched by me)
-
-2007-12-10 16:36 +0000 [r92204] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Add G729A as another possible payload name for G729.
- Some devices use this instead of G729, which is perfectly normal
- since the payload number itself is defined and can't be used by
- anything else so the name doesn't matter that much. (closes issue
- #11483) Reported by: revolution Patches: rtp.diff uploaded by
- revolution (license 346)
-
-2007-12-10 16:29 +0000 [r92202] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: If there are no members in a queue, then the
- loop where the datastore for detecting duplicate dialed numbers
- will be skipped, meaning the datastore isn't created. This means
- that when we try to free it, there's a crash. This stops that
- crash from occurring. (closes issue #11499, reported by slavon,
- patched by eliel)
-
-2007-12-10 16:13 +0000 [r92200] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: It is possible for nativeformats to contain
- more then one codec, so print out multiple ones. (closes issue
- #11366) Reported by: ovi
-
-2007-12-10 14:04 +0000 [r92158] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Avoid reinvite race situations with two
- Asterisks trying to reinvite each other in 1.4 and trunk. This
- patch implements support for the 491 error code that Asterisk 1.4
- generates on situations where we get an incoming INVITE and
- already has one in progress. Thanks to mavetju for reporting and
- to Raj Jain for an excellent explanation of the problem. Patch by
- myself. Tested with 8 Asterisk servers connected to each other in
- a training network. Closes issue #10481
-
-2007-12-07 23:29 +0000 [r91890] Jason Parker <jparker@digium.com>
-
- * main/dsp.c: We need to make sure we free the input frame if we
- return a different frame in ast_dsp_process. Issue 11273, pointed
- out by dimas, with a patch by eliel.
-
-2007-12-07 22:30 +0000 [r91870] Kevin P. Fleming <kpfleming@digium.com>
-
- * codecs/codec_zap.c: even though Asterisk explicitly requests that
- endpoints using G.729 do *not* use Annex B (silence detection and
- comfort noise generation) some do anyway; the transcoder card
- interface does not currently work properly with CNG frames, so
- trim off the CNG before sending the data
-
-2007-12-07 21:24 +0000 [r91777-91830] Russell Bryant <russell@digium.com>
-
- * main/utils.c: Make the lock protecting each thread's list of
- locks it currently holds recursive. I think that this will fix
- the situation where some people have said that "core show locks"
- locks up the CLI. (related to issue #11080)
-
- * include/asterisk/lock.h: Fix another bug in the DEBUG_THREADS
- code. The ast_mutex_init() function had the mutex attribute
- object marked as static. This means that multiple threads
- initializing locks at the same time could step on each other and
- end up with improperly initialized locks. (found when tracking
- down locking issues related to issue #11080)
-
- * include/asterisk/lock.h: I love fixing lock related errors in the
- lock debugging code. That's about as ironic as it gets in
- Asterisk programming land. Anyway, I spotted this bug while
- trying to track down why systems are locking up and acting weird
- in issue #11080. The mutex attribute object was marked as static
- in this function when it should not have been.
-
- * apps/app_dial.c: * Add channel locking around datastore
- operations that expect the channel to be locked. * Document why
- we don't record Local channels in the dialed interfaces list. *
- Remove the dialed variable as it isn't needed. * Restructure some
- code for clarity and coding guidelines stuff
-
- * apps/app_queue.c: * Add channel locking around datastore
- operations that expect the channel to be locked. * Document why
- we don't record Local channels in the dialed interfaces list. *
- Handle memory allocation failure. * Remove the dialed variable,
- as it wasn't actually needed. * Tweak some formatting to conform
- to coding guidelines.
-
- * main/autoservice.c: * Add a bit more of a verbose comment as to
- why a hangup frame needs to be queued up if autoservice gets a
- NULL return from ast_read(). * Make the process of queueing the
- hangup frame more efficient by putting the frame where it is
- going to end up and avoiding some locking and extra memory
- allocations and freeing.
-
-2007-12-07 15:39 +0000 [r91737] Mark Michelson <mmichelson@digium.com>
-
- * main/autoservice.c: Hangups that happen during autoservice were
- not processed appropriately. This is because a hangup actually
- causes a NULL frame to be received, not a hangup frame. Queueing
- a hangup if we receive a NULL frame during autoservice corrects
- this problem (closes issue #11467, reported by jmls, patched by
- me)
-
-2007-12-07 02:51 +0000 [r91675-91693] Russell Bryant <russell@digium.com>
-
- * apps/app_dial.c: Don't unlock the dialed_interfaces list until
- we're done messing with the iterator.
-
- * apps/app_dial.c, apps/app_queue.c: Allow dialing local channels
- from Queue() and Dial() again. There was a slight flaw in the
- code to prevent call forwards from looping that caused this
- problem. (related to issue #11486)
-
- * apps/app_queue.c: Fix in an issue in the call forwarding handling
- code that was causing crashes on every call into a queue. I'm not
- entirely sure about the logic in this part of the code, so I want
- to look at it some more tomorrow. However, this makes it safe and
- keeps it from crashing. (closes issue #11486, reported by adamg,
- patched by me)
-
-2007-12-07 00:52 +0000 [r91637] Tilghman Lesher <tlesher@digium.com>
-
- * main/rtp.c: At the end of a call, when we're reporting, RTCP may
- already be partially torn down, so check for NULL dereference
- Reported by: blitzrage Patch by: tilghman (Closes issue #11450)
-
-2007-12-06 20:25 +0000 [r91541] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: IMAP storage did not honor the maxmsg
- setting in voicemail.conf, and it also had the possibility of
- crashing if a user had more than 256 messages in their voicemail.
- This patch kills two birds with one stone by adding maxmsg
- support and also setting a hard limit on the number of messages
- at 255 so that the crashes cannot happen. (closes issue #11101,
- reported by Skavin, patched by me)
-
-2007-12-06 19:11 +0000 [r91501] Russell Bryant <russell@digium.com>
-
- * main/loader.c, include/asterisk/module.h: Add a new module flag
- to indicate that a build sum is present. Modules built against
- older Asterisk 1.4 headers will now load properly with just a
- warning indicating that they are old and may cause problems.
- (patch by paravoid)
-
-2007-12-06 16:49 +0000 [r91439-91450] Joshua Colp <jcolp@digium.com>
-
- * main/udptl.c: Fix various in the udptl implementation. It could
- return empty modem frames, have an incorrect sequence number on
- packets, and display the wrong sequence number in the debug
- messages. (closes issue #11228) Reported by: Cache Patches:
- udptl-4.patch uploaded by dimas (license 88)
-
- * channels/chan_sip.c: Add support for accepting and sending T.38
- in the initial INVITE. (closes issue #9402) Reported by: thdei
-
-2007-12-06 12:54 +0000 [r91366] Olle Johansson <oej@edvina.net>
-
- * main/loader.c, include/asterisk/logger.h, main/logger.c: Make
- sure logger is reloaded at general reload in the cli. (Discovered
- during Asterisk training in Portugal)
-
-2007-12-05 22:57 +0000 [r91273-91292] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Reverting extra stuff I didn't mean to
- commit
-
- * apps/app_voicemail.c, apps/app_dial.c: The 'G' option for Dial()
- did not properly handle the case where only a label was provided.
- This was due to the fact that the answering channel did not have
- an extension set, so ast_parseable_goto would fail. This fix
- eliminates the call to ast_parseable_goto on the answering
- channel since it is a wasteful call. The answering channel and
- the calling channel are both directed to the same extension and
- context, just different priorities, so we can just copy the
- values from the calling channel to the answering channel and
- increment the answering channel's priority. (closes issue #11382,
- reported by jon, patch by me with correction by jon)
-
-2007-12-05 21:38 +0000 [r91237] Tilghman Lesher <tlesher@digium.com>
-
- * sounds/Makefile: Upgrade to the latest version of extra sounds
-
-2007-12-05 17:31 +0000 [r90967-91192] Russell Bryant <russell@digium.com>
-
- * main/threadstorage.c: Make the lock in the threadstorage
- debugging code untracked to avoid a deadlock on thread
- destruction. (closes issue #11207) Reported by: ys Patches:
- threadstorage.c.diff uploaded by ys (license 281) Also fixes an
- open bug report: (closes issue #11446)
-
- * main/utils.c: When DEBUG_THREADS is enabled, we only have the
- details about who is holding a lock that we are waiting on for a
- mutex, not rwlocks. This should fix the problem where people have
- reported "core show locks" crashing sometimes.
-
- * include/asterisk/lock.h: Fix some crashes in chan_iax2 that were
- reported as happening on Mac systems. It turns out that the
- problem was the Mac version of the ast_atomic_fetchadd_int()
- function. The Mac atomic add function returns the _new_ value,
- while this function is supposed to return the old value. So, the
- crashes happened on unreferencing objects. If the reference count
- was decreased to 1, ao2_ref() thought that it had been decreased
- to zero, and called the destructor. However, there was still an
- outstanding reference around. (closes issue #11176) (closes issue
- #11289)
-
- * include/asterisk/file.h, configure,
- include/asterisk/autoconfig.h.in, configure.ac,
- include/asterisk/compiler.h: Modify file.h to maintain API
- compatibility with earlier versions. If a recent compiler is
- being used, then a warning will show up for any modules still
- using the old name "private" instead of "_private". (patch
- suggested by paravoid)
-
- * main/pbx.c: Make some changes to some additions I made recently
- for doing channel autoservice when looking up extensions. This
- code was added to handle the case where a dialplan switch was in
- use that could block for a long time. However, the way that I
- added it, it did this for all extension lookups. However, lookups
- in the in-memory tree of extensions should _not_ take long enough
- to matter. So, move the autoservice stuff to be only around
- executing a switch.
-
-2007-12-04 17:28 +0000 [r90876] Jason Parker <jparker@digium.com>
-
- * main/channel.c: If we fail to create a channel after allocating a
- timing fd, we need to make sure to close it. Issue 11454, patch
- by eliel.
-
-2007-12-04 05:29 +0000 [r90798] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c: Fix build issue on the build cluster.
-
-2007-12-03 23:50 +0000 [r90736-90753] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/compat.h: Solaris requires the inclusion of
- sys/loadavg.h for getloadavg(). Reported by: snuffy Patch by:
- snuffy,tilghman (Closes issue #11430)
-
- * res/res_config_pgsql.c: If both dbhost and dbsock were not set, a
- NULL deref could result Reported by: xrg Patch by: tilghman
- (Closes issue #11387)
-
-2007-12-03 23:12 +0000 [r90735] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_dial.c, main/channel.c, main/global_datastores.c
- (added), channels/chan_local.c, main/Makefile,
- include/asterisk/channel.h, include/asterisk/global_datastores.h
- (added), apps/app_queue.c: A big one... This is the merge of the
- forward-loop branch. The main change here is that call-forwards
- can no longer loop. This is accomplished by creating a datastore
- on the calling channel which has a linked list of all devices
- dialed. If a forward happens, then the local channel which is
- created inherits the datastore. If, through this progression of
- forwards and datastore inheritance, a device is attempted to be
- dialed a second time, it will simply be skipped and a warning
- message will be printed to the CLI. After the dialing has been
- completed, the datastore is detached from the channel and
- destroyed. This change also introduces some side effects to the
- code which I shall enumerate here: 1. Datastore inheritance has
- been backported from trunk into 1.4 2. A large chunk of code has
- been removed from app_dial. This chunk is the section of code
- which handles the call forward case after the channel has been
- requested but before it has been called. This was removed because
- call-forwarding still works fine without it, it makes the code
- less error-prone should it need changing, and it made this set of
- changes much less painful to just have the forwarding handled in
- one place in each module. 3. Two new files, global_datastores.h
- and .c have been added. These are necessary since the datastore
- which is attached to the channel may be created and attached in
- either app_dial or app_queue, so they need a common place to find
- the datastore info. This approach was taken in case similar
- datastores are needed in the future, there will be a common place
- to add them.
-
-2007-12-03 22:06 +0000 [r90696] Jason Parker <jparker@digium.com>
-
- * apps/app_meetme.c: Make sure we always close the conference fd if
- we have an open one. Issue 11383, reported by markmhy, patch by
- eliel.
-
-2007-12-03 20:59 +0000 [r90639] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_mgcp.c: Changing some bad logic when calculating
- the interdigit timeout. (closes issue #11402, reported and
- patched by eferro)
-
-2007-12-03 20:51 +0000 [r90607] Jason Parker <jparker@digium.com>
-
- * res/res_features.c: Fix crash in ParkAndAnnounce application.
- Issue #11436, reported by lytledd, patch by eliel.
-
-2007-12-03 20:05 +0000 [r90548-90588] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Do not create a smoother for G723.1 frames, they need
- to be left alone to their native 20/24 byte size.
-
- * .cleancount, main/channel.c, include/asterisk/channel.h: Preserve
- the indication currently playing on a channel when a masquerade
- operation happens. (issue #BE-88)
-
-2007-12-03 18:20 +0000 [r90546] Jason Parker <jparker@digium.com>
-
- * channels/chan_iax2.c: Only log debug messages if debug is
- enabled. Closes issue #11416, patch by casper.
-
-2007-12-02 18:18 +0000 [r90470] Russell Bryant <russell@digium.com>
-
- * apps/app_queue.c: The other day when I went through making
- changes as a result of the ao2_link() change, I added some code
- to set pointers to NULL after they were unreferenced. This
- pointed out that in this place, the object was unreferenced
- before the code was done using it. So, move the unref down a
- little bit. (crash reported by jmls on IRC)
-
-2007-12-02 09:34 +0000 [r90432] Tilghman Lesher <tlesher@digium.com>
-
- * main/autoservice.c: Clarify the return value on autoservice.
- Specifically, if you started autoservice and autoservice was
- already on, it would erroneously return an error. Reported by:
- adiemus Patch by: dimas (Closes issue #11433)
-
-2007-11-30 19:26 +0000 [r90310-90348] Russell Bryant <russell@digium.com>
-
- * main/astobj2.c, main/manager.c, include/asterisk/astobj2.h,
- apps/app_queue.c, channels/chan_iax2.c: Change the behavior of
- ao2_link(). Previously, in inherited a reference. Now, it
- automatically increases the reference count to reflect the
- reference that is now held by the container. This was done to be
- more consistent with ao2_unlink(), which automatically releases
- the reference held by the container. It also makes it so it is no
- longer possible for a pointer to be invalid after ao2_link()
- returns.
-
- * include/asterisk/astobj2.h: Add some notes on the behavior of
- ao2_unlink() after a discussion with Tilghman
-
-2007-11-30 14:43 +0000 [r90269] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Fix locking issues under one legged replaces
- scenarios. (closes issue #11420) Reported by: irroot Patches:
- chan_sip_oneleg.patch uploaded by irroot (license 52)
-
-2007-11-30 00:16 +0000 [r90231] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_mgcp.c: Clear the DTMF buffer if the call times
- out. (closes issue #11418, reported and patched by eferro)
-
-2007-11-29 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.15 released.
-
-2007-11-29 19:48 +0000 [r90166] Tilghman Lesher <tlesher@digium.com>
-
- * cdr/cdr_pgsql.c: Properly escape cdr->src and cdr->dst and ensure
- we use thread-safe escaping (Fixes AST-2007-026)
-
-2007-11-29 19:38 +0000 [r90163] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: This patch handles the case where a queue
- member with a negative penalty is added via the manager. If a
- negative value is submitted for a member penalty, we set it to 0.
- (closes issue #11411, reported and patched by Laureano)
-
-2007-11-29 19:24 +0000 [r90154-90160] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_config_pgsql.c: Properly escape input buffers (Fixes
- AST-2007-025)
-
- * formats/format_g726.c, include/asterisk/file.h,
- formats/format_wav.c, formats/format_pcm.c,
- formats/format_ogg_vorbis.c, main/file.c, formats/format_h263.c,
- formats/format_h264.c, formats/format_wav_gsm.c: Use of "private"
- as a field name in a header file messes with C++ projects
- Reported by: chewbacca Patch by: casper (Closes issue #11401)
-
- * sounds/Makefile: Upgrade the core sounds release version
-
-2007-11-29 00:36 +0000 [r90142-90147] Russell Bryant <russell@digium.com>
-
- * funcs/func_callerid.c: fix some formatting i accidentally changed
-
- * funcs/func_callerid.c, main/channel.c,
- include/asterisk/channel.h: This set of changes is to make some
- callerID handling thread-safe. The ast_set_callerid() function
- needed to lock the channel. Also, the handlers for the CALLERID()
- dialplan function needed to lock the channel when reading or
- writing callerid values directly on the channel structure.
-
- * include/asterisk/file.h, main/file.c: Merge a change from
- team/russell/chan_refcount ... This makes ast_stopstream()
- thread-safe.
-
-2007-11-28 22:59 +0000 [r90101] Joshua Colp <jcolp@digium.com>
-
- * apps/app_queue.c: Fix a few memory leaks. (closes issue #11405)
- Reported by: eliel Patches: load_realtime.patch uploaded by eliel
- (license 64)
-
-2007-11-28 22:30 +0000 [r90098] Kevin P. Fleming <kpfleming@digium.com>
-
- * configs/users.conf.sample, main/manager.c: it is impossible to
- set permissions for manager accounts created by users.conf
- (reported internally, patched by me)
-
-2007-11-28 22:08 +0000 [r89999-90059] Mark Michelson <mmichelson@digium.com>
-
- * main/pbx.c: Removing some seemingly pointless code. This sets a
- channel variable for every priority executed in the dialplan if
- you have debug set to anything non-zero. This seems pointless due
- to the fact that these channel variables are not referenced
- anywhere else in the code and their names are esoteric enough
- that they would not be practical to reference in the dialplan.
- Plus the fact that this behavior isn't documented anywhere means
- that the change is not likely to cause any disruption. If
- anything, this may actually cause a slight performance increase
- if running with debug on. The motivating influence for this code
- change is the eventwhencalled option for queues. If set to vars,
- all channel variables will be output to the manager. These
- unnecessary channel variables make the output a lot more
- difficult to deal with.
-
- * apps/app_voicemail.c: Recording greetings when using IMAP storage
- was causing zero-length files to be stored. Since greetings are
- not retrieved from IMAP anyway, it is pointless to attempt
- storing them there. (closes issue #11359, reported by spditner,
- patched by me)
-
-2007-11-28 00:20 +0000 [r89839-89893] Russell Bryant <russell@digium.com>
-
- * main/pbx.c, include/asterisk/pbx.h: - update documentation for
- some of the goto functions to note that they handle locking the
- channel as needed - update ast_explicit_goto() to lock the
- channel as needed
-
- * main/autoservice.c: Don't do frame processing if ast_read()
- returned NULL.
-
- * apps/app_queue.c: Instead of depending on the return value of
- ast_true(), explicitly set the eventwhencalled variable to 1.
-
- * main/pbx.c: Don't start/stop autoservice in
- pbx_extension_helper() unless a channel exists
-
-2007-11-27 23:10 +0000 [r89837] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Two changes with regards to the
- 'eventwhencalled' option of queues.conf 1) Due to some signed vs.
- unsigned silliness, setting 'eventwhencalled' to 'vars' or 'yes'
- did exactly the same thing. Thus the sign change of the ast_true
- call. 2) The vars2manager function overwrote a \n for every
- channel variable it parsed, resulting in bizarre output for the
- channel variables. This patch remedies this. (related to issue
- #11385, however I'm not sure if this will actually be enough to
- close it)
-
-2007-11-27 21:45 +0000 [r89790] Russell Bryant <russell@digium.com>
-
- * main/autoservice.c, main/pbx.c: Merge changes from
- team/russell/autoservice_1.4 This set of changes fixes an issue
- that was reported to me on IRC yesterday. The user, d1mas, was
- using chan_zap for incoming calls and was having DTMF recognition
- issues in some situations. Specifically, he noticed that the
- problem occurred when using DISA or WaitExten. He also noticed
- that when using Read, the problem did not occur. His system also
- used DUNDi for dialplan lookups. So, he theorized that if the
- DUNDi lookups blocked for some period of time, that audio from
- the zap channel could get lost. If the audio got lost, then it
- wouldn't be run through the DTMF detector, and digits could get
- lost. He was correct, and the following set of changes fixes the
- problem. However, the changes go a little bit further than what
- was necessary to fix this exact problem. 1) I updated
- pbx_extension_helper() to autoservice the associated channel to
- handle cases where extension lookups may take a long time. This
- would normally be a dialplan switch that does some lookup over
- the network, such as the DUNDi or IAX2 switches. This ensures
- that even while a DUNDi lookup is blocking, the channel will be
- continuously serviced. 2) I made a change to the autoservice
- code. This is actually something that has bothered me for a long
- time. When a channel is in autoservice, _all_ frames get thrown
- away. However, some frames really shouldn't be thrown away. The
- most notable examples are signalling (CONTROL) frames, and DTMF.
- So, this patch queues up important frames while a channel is in
- autoservice. When autoservice is stopped on the channel, the
- queued up frames get stuck back on the channel so that they can
- get processed instead of thrown away. 3) I made another change to
- the autoservice code to handle the case where autoservice is
- started on channels recursively. Previously, you could call
- ast_autoservice_start() multiple times on a channel, and it would
- stop the first time ast_autoservice_stop() gets called. Now, it
- will ensure that autoservice doesn't actually stop until the
- final call to ast_autoservice_stop().
-
-2007-11-27 20:22 +0000 [r89727] Mark Michelson <mmichelson@digium.com>
-
- * res/res_config_pgsql.c: Changing some calls from free() to
- ast_free() since they were allocated with ast_calloc(). (closes
- issue #11390, reported and patched by Laureano)
-
-2007-11-27 20:16 +0000 [r89701-89709] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/app.c: on second thought... revert all the other changes
- i've made in app options parsing leaving only one: if an empty
- argument is supplied for an option, set that argument pointer to
- point to an empty string rather than NULL, so that the
- application can do normal checks on it without worrying about it
- being NULL
-
- * main/app.c: generate a warning when an application option that
- requires an argument is ignored due to lack of an argument
-
-2007-11-27 16:12 +0000 [r89634] Russell Bryant <russell@digium.com>
-
- * configs/voicemail.conf.sample: Add a note to the sample voicemail
- config noting that when using IMAP storage, only the first format
- specified will be attached to the message.
-
-2007-11-27 15:38 +0000 [r89631] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_env.c: Default result of STAT should be "0" not "".
- Reported via the -users mailing list, fixed by me.
-
-2007-11-27 15:23 +0000 [r89624-89630] Olle Johansson <oej@edvina.net>
-
- * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: If we
- get a codec offer using a well-known payload type, but using it
- for another codec that we don't know, Asterisk did not remove
- that codec from the list. With this patch, we remove the codec
- from audio and video rtp objects and deny it ever existed. Thanks
- to lasse for testing. (closes issue #11376) Reported by: lasse
- Patches: bug11376.txt uploaded by oej (license 306) Tested by:
- lasse
-
- * configs/sip.conf.sample: Clarify limitonpeers=yes (closes issue
- #11304) Reported by: pj
-
-2007-11-27 06:24 +0000 [r89622] Steve Murphy <murf@digium.com>
-
- * apps/app_dial.c, main/cdr.c, configs/cdr.conf.sample,
- include/asterisk/cdr.h: closes issue #11379; OK, this is an
- attempt to make both sides happy. To the cdr.conf file, I added
- the option 'unanswered', which defaults to 'no'. In this mode,
- you will see a cdr for a call, whether it was answered or not.
- The disposition will be NO ANSWER or ANSWERED, as appropriate.
- The src is as you'd expect, the destination channel will be one
- of the channels from the Dial() call, usually the last in the
- list if more than one chan was specified. With unanswered set to
- 'yes', you will still see this cdr entry in both cases. But in
- the case where the dial timed out, you will also see a cdr for
- each line attempted, marked NO ANSWER, with no destination
- channel name. The new option defaults to 'no', so you don't see
- the pesky extra cdr's by default, and you will not see the
- irritating 'not posted' messages.
-
-2007-11-26 23:10 +0000 [r89616-89618] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_playback.c: After issuing a "say load new", if a caller
- hangs up during the middle of playback of a number, app_playback
- will continue to try to play the remaining files. With this
- change, no more files will be played back upon hangup. (closes
- issue #11345, reported and patched by IgorG)
-
- * apps/app_playback.c: After issuing a "say load new" tons of
- warning messages are printed out to the CLI every time do_say in
- app_playback is called. Removing these warnings
-
-2007-11-26 21:10 +0000 [r89599-89610] Joshua Colp <jcolp@digium.com>
-
- * main/dial.c: Fix issues with async dialing with an application
- executing. The application has to be terminated and control
- returned to the thread before hanging things up. (issue #BE-252)
-
- * res/res_features.c: Add module counting removal for error
- conditions. (closes issue #11333) Reported by: Laureano Patches:
- res_features_v2.c.patch uploaded by Laureano (license 265)
-
-2007-11-26 17:41 +0000 [r89594] Russell Bryant <russell@digium.com>
-
- * main/pbx.c: Add channel locking to a function that needed to be
- doing it. This is just a little something I noticed while working
- on a completely unrelated issue.
-
-2007-11-26 17:36 +0000 [r89587-89592] Joshua Colp <jcolp@digium.com>
-
- * pbx/pbx_config.c: Use ast_free to free memory, or else we shall
- implode if MALLOC_DEBUG is enabled. (closes issue #11347)
- Reported by: ys Patches: pbx.pbx_config.c.diff uploaded by ys
- (license 281)
-
- * apps/app_mixmonitor.c: Close the audio file before sending it to
- the post processing application. (closes issue #11357) Reported
- by: reformed Patches: mixmonitor.patch uploaded by reformed
- (license 330)
-
-2007-11-26 17:20 +0000 [r89586] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/app.c: when parsing application options that take arguments,
- don't indicate that the option was supplied unless a
- non-zero-length argument was found for it
-
-2007-11-26 15:48 +0000 [r89580] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Revert vmu->email back to an empty string
- if it was empty when imap_store_file was called. This prevents
- sending a duplicate e-mail. (closes issue #11204, reported by
- spditner, patched by me)
-
-2007-11-26 15:34 +0000 [r89571-89577] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: If channel allocation fails because the alert
- pipe could not be created also free the scheduler context.
- (closes issue #11355) Reported by: eliel Patches:
- main.channel.c.patch uploaded by eliel (license 64)
-
- * apps/app_meetme.c: When unloading app_meetme destroy any auto
- created contexts created by SLA. (closes issue #11367) Reported
- by: eliel
-
-2007-11-25 17:17 +0000 [r89559] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_odbc.c, configs/res_odbc.conf.sample,
- include/asterisk/res_odbc.h, res/res_config_odbc.c: We previously
- attempted to use the ESCAPE clause to set the escape delimiter to
- a backslash. Unfortunately, this does not universally work on all
- databases, since on databases which natively use the backslash as
- a delimiter, the backslash itself needs to be delimited, but on
- other databases that have no delimiter, backslashing the
- backslash causes an error. So the only solution that I can come
- up with is to create an option in res_odbc that explicitly
- specifies whether or not backslash is a native delimiter. If it
- is, we use it natively; if not, we use the ESCAPE clause to make
- it one. Reported by: elguero Patch by: tilghman (Closes issue
- #11364)
-
-2007-11-24 16:59 +0000 [r89534-89545] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_adsi.c: Free some frames that would otherwise leak on
- error. Reported by: Laureano Patch by: Laureano,tilghman (Closes
- issue #11351)
-
- * apps/app_voicemail.c, main/app.c: Currently, zero-length
- voicemail messages cause a hangup in VoicemailMain. This change
- fixes the problem, with a multi-faceted approach. First, we do
- our best to avoid these messages from being created in the first
- place, and second, if that fails, we detect when the voicemail
- message is zero-length and avoid exiting at that point. Reported
- by: dtyoo Patch by: gkloepfer,tilghman (Closes issue #11083)
-
- * main/manager.c: Up until this point, the XML output of the
- manager has been technically invalid, due to the repetition of
- certain parameters in a single event. This caused various issues
- for XML parsers, some of which refused to parse at all, given the
- invalidity of the rendered XML. So this commit fixes the XML
- output, ensuring that each entity parameter has a unique name,
- thus ensuring valid XML. Reported by: msetim Patch by: tilghman
- (Closes issue #10220)
-
- * res/res_config_odbc.c: Use ESCAPE clause for the first parameter,
- not just 2nd-Nth parameters. Reported by: apsaras Patch by:
- tilghman (Closes issue #11353)
-
-2007-11-22 17:29 +0000 [r89527] Russell Bryant <russell@digium.com>
-
- * configs/agents.conf.sample: mvanbaak pointed out a spelling error
- in this sample configuration file. While I was at it, I went
- ahead and tweaked it a little bit more.
-
-2007-11-21 19:27 +0000 [r89493-89495] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Fix a small error I made in my previous commit
-
- * apps/app_queue.c: Changing an inaccurate debug message to be less
- inaccurate. Under the circumstances, this message would always
- report that there were 0 members available, even though that may
- not be true.
-
-2007-11-21 18:59 +0000 [r89491] Terry Wilson <twilson@digium.com>
-
- * res/res_features.c: If a channel gets masqueraded in the middle
- of a park, don't play the announcement to the masqueraded
- channel, and dial back to the original channel on timeout.
-
-2007-11-20 19:16 +0000 [r89461-89462] Kevin P. Fleming <kpfleming@digium.com>
-
- * include/asterisk/module.h: re-doxygen some comments
-
- * main/loader.c, include/asterisk/module.h,
- build_tools/make_buildopts_h: bring back compile-option checking
- when loading modules, only this time use a string-based storage
- and comparison mechanism because it is easier to support on other
- platforms
-
-2007-11-20 17:50 +0000 [r89457] Mark Michelson <mmichelson@digium.com>
-
- * main/pbx.c: According to comments in main/pbx.c, it is essential
- that if we are going to lock the conlock as well as the hints
- lock, it must be locked in that respective order. In order to
- prevent a potential deadlock, we need to lock the conlock prior
- to locking the hints lock in ast_hint_state_changed (see the call
- stack example on issue #11323 for how this can happen). (closes
- issue #11323, reported by eelcob, suggestion for patch by eelcob,
- patch by me)
-
-2007-11-20 15:22 +0000 [r89450] Steve Murphy <murf@digium.com>
-
- * doc/queues-with-callback-members.txt: closes issue #11324; break
- statements missing in switch cases.
-
-2007-11-20 13:40 +0000 [r89445] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c: added RR patch from iroot #10908, thanks.
-
-2007-11-19 15:53 +0000 [r89416-89419] Joshua Colp <jcolp@digium.com>
-
- * res/res_features.c: Print out the correct filename
- (features.conf) in the log message when parkpos options are
- incorrect. (closes issue #11295) Reported by: Laureano Patches:
- res_features.c.patch uploaded by Laureano (license 265)
-
- * doc/localchannel.txt: Clarify documentation a bit, include that a
- frame has to pass through the core in order for the Local channel
- optimization to happen. (closes issue #11246) Reported by: jon
-
-2007-11-16 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.14 released.
-
-2007-11-16 22:26 +0000 [r89339] Russell Bryant <russell@digium.com>
-
- * main/loader.c, include/asterisk/module.h,
- build_tools/make_buildopts_h: Temporarily revert revision 89325,
- which added md5 magic for keeping track of what build options
- were used. We agreed that we should remove this before making a
- 1.4 release, and then we can put it back in. Then, we can take a
- month or so to play around with it to get it how we want it.
-
-2007-11-16 16:47 +0000 [r89325] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/loader.c, include/asterisk/module.h,
- build_tools/make_buildopts_h: To help combat problems where
- people build external modules (asterisk-addons or others) and
- then change the build options of the Asterisk build in a way that
- makes the incompatible without warning, this commit introduces an
- MD5 signature of the important build-time options and includes
- that signature into modules when they are built. When the loader
- loads one of these modules and notices the problem, it will emit
- a warning to console and refuse to initialize the module, as
- doing so could cause the system to be unstable or even crash. If
- you upgrade to this version of Asterisk, you must rebuild *all*
- of your modules that came from other sources before trying to run
- this version. If you are using Digium's G.729 binary codec
- module, you will need v33 or newer.
-
-2007-11-16 15:28 +0000 [r89323] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Make realtime queues accessible from the
- QUEUE_MEMBER_COUNT function. (closes issue #11271, reported and
- patched by atis, with small modifications from me)
-
-2007-11-15 18:37 +0000 [r89298-89302] Tilghman Lesher <tlesher@digium.com>
-
- * Makefile: Start Asterisk in Debian at a more reasonable time
- (since zaptel is at level 20)
-
- * channels/misdn/isdn_lib.c: Fix an uninitialized memory read found
- by valgrind
-
- * channels/chan_iax2.c: Yet another memory corruption issue.
- Reported by: atis Patch by: tilghman Fixes issue #10923
-
-2007-11-15 17:19 +0000 [r89296] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c: Update the SLAStation application to account
- for the case where the SLA thread has a call out to the station,
- but the user has pressed a line button to answer the call instead
- of picking up the handset. If they do, the phone sends out a new
- INVITE. So, the SLAStation app must check to see if it is picking
- up a ringing trunk, and ensure that the other stations stop
- ringing. (reported internally, patched by me, tested by mogorman)
-
-2007-11-15 14:57 +0000 [r89286-89288] Mark Michelson <mmichelson@digium.com>
-
- * main/manager.c: Undoing previous commit since I realize it was
- wrong
-
- * main/manager.c: Adding a missing mutex unlock. (closes issue
- 11256, reported and patched by ys)
-
-2007-11-15 11:26 +0000 [r89280-89281] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Don't send re-invites during pending INVITE
- transactions. Patch by one47 - thanks! Closes issue #9305
-
- * channels/chan_sip.c: Improve support for multipart messages. Code
- by gasparz, changes by me (mostly formatting). Thanks, gasparz!
- Closes issue #10947
-
-2007-11-14 23:23 +0000 [r89275] Tilghman Lesher <tlesher@digium.com>
-
- * main/app.c: When a recording ends with '#', we are improperly
- trimming an extra 200ms from the recording. Reported by: sim
- Patch by: tilghman Closes issue #11247
-
-2007-11-14 01:15 +0000 [r89260] Joshua Colp <jcolp@digium.com>
-
- * main/srv.c: Return the proper value when the srv_callback
- function executes properly. (closes issue #11240) Reported by:
- jtodd
-
-2007-11-13 21:07 +0000 [r89248-89254] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c, channels/chan_iax2.c: Fix building on newer
- systems which require a third arg to open() when using O_CREAT.
- Issue 11238, reported by puzzled.
-
- * res/res_features.c: Revert change from revision 67064. It is
- documented behavior that if a parking extension already exists
- while using PARKINGEXTEN, dialplan execution will continue. If
- blind transferring to a Park with PARKINGEXTEN, you must keep
- this in mind, and handle the failure yourself. Issue 11237,
- reported by jon.
-
-2007-11-13 17:34 +0000 [r89246] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_sip.c: If we set a value for qualify, we should
- actually pay attention to it, instead of overriding the value
-
-2007-11-13 16:02 +0000 [r89241] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_mixmonitor.c: Reverting commit made in revision 89205
- since it is unnecessary. Thanks to Kevin for pointing this out
-
-2007-11-13 13:51 +0000 [r89239] Tilghman Lesher <tlesher@digium.com>
-
- * main/utils.c: Debugging is running into the 16-lock limit.
- Increase to avoid. (This define is only effective when debugging
- is turned on, so there's no effect for most installations.)
-
-2007-11-13 00:56 +0000 [r89205] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_mixmonitor.c: Some sanity checking for MixMonitor. If
- only 1 argument is given, then the args.options and
- args.post_process strings are uninitialized and could contain
- garbage. This change handles this situation properly by only
- using arguments that we have parsed.
-
-2007-11-12 20:46 +0000 [r89194] Jason Parker <jparker@digium.com>
-
- * main/pbx.c: Fix a typo pointed out by De_Mon on #asterisk-dev
-
-2007-11-12 20:16 +0000 [r89184-89191] Tilghman Lesher <tlesher@digium.com>
-
- * main/config.c: If two config writes collide, file corruption
- could result. Use a mkstemp() file, instead. Reported by:
- paravoid Patch by: tilghman Closes issue #10781
-
- * main/channel.c, channels/chan_sip.c: Fix two cases of memory
- corruption caused by background threads. Reported by: atis Patch
- by: tilghman Fixes issue #10923
-
-2007-11-12 11:26 +0000 [r89169-89173] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, configs/misdn.conf.sample: if we're NT and
- no number was dialed and overlapdial is set, we wait for the ISDN
- timeout instead of starting our own timer. added a comment for
- the misdn.conf.sample for the overlapdial config option.
-
- * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h,
- channels/chan_misdn.c, channels/misdn/isdn_msg_parser.c: added
- restart all interfaces Restart_Indicator, to automatically send a
- RESTART after the L2 of a PTP Port comes up. Also fixed some
- places where we have send a RELEASE without need for it.
-
- * channels/misdn/isdn_lib.c, channels/chan_misdn.c: fixed a
- state/event issue with overlapdial=yes when no extension matched.
- removed the general sending of a RELEASE_COMPLETE when we receive
- a RELEASE, this is done by mISDNuser/mISDN. This makes it
- possible to use asterisk-1.4 with mISDN trunk, but requires users
- of mISDN/mISDNuser-1.1.X to upgrade to at least mISDNuser-1.1.6
- (when using the NT mode at all)
-
- * channels/misdn/isdn_lib.c: fixed the support for CW and therefore
- for the reject_cause option.
-
- * channels/misdn/isdn_lib.c, channels/misdn_config.c,
- channels/misdn/isdn_lib.h, channels/chan_misdn.c,
- channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample:
- aded ntkeepcalls option, to avoid droÃpping calls when the L2
- goes down on a PTP link. There are some pbx which do turn off the
- L1 for a very short while and restart it immediately. normally
- T310 should be started and after 10 seconds or so the calls
- should be dropped, this is a simple fix wihtout this timer.
-
-2007-11-08 23:52 +0000 [r89125] Jason Parker <jparker@digium.com>
-
- * main/say.c: Properly say the seconds here.. Issue 11203, fix
- described by vma.
-
-2007-11-08 21:00 +0000 [r89119] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: Rework of the commit I made yesterday to use
- the already built-in ast_uri_decode function as opposed to my
- home-rolled one. Also added comments. Thanks to oej for pointing
- me in the right direction
-
-2007-11-08 18:45 +0000 [r89115] Jason Parker <jparker@digium.com>
-
- * configs/res_odbc.conf.sample: Avoid warnings on load when using
- sample configuration files. Issue 11195, patch by eliel.
-
-2007-11-08 16:47 +0000 [r89111] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: I made this same adjustment in trunk to fix
- a bug, and it makes sense to do it in 1.4 as well. If an
- imapfolder is specified in voicemail.conf, don't ever explicitly
- connect to INBOX since it may not exist.
-
-2007-11-08 05:26 +0000 [r89105] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/srv.c: fix a glaring bug in the new SRV record handling that
- would cause incorrect weight sorting
-
-2007-11-08 04:55 +0000 [r89103] Tilghman Lesher <tlesher@digium.com>
-
- * doc/valgrind.txt: Typo
-
-2007-11-08 02:26 +0000 [r89095-89101] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Do not add a sip: to the beginning of the To
- URI unless needed. (closes issue #10756) Reported by: goestelecom
-
- * channels/chan_sip.c: Improve the devicestate logic for multiple
- devices. If any are available then the extension is considered
- available. (closes issue #10164) Reported by: nic_bellamy
- Patches: sip-hinting-svn-branch-1.4.patch uploaded by nic
- (license 299)
-
- * channels/chan_sip.c: Add support for allowing one outgoing
- transaction. This means if a response comes back out of order
- chan_sip will still handle it. I dream of a chan_sip with real
- transaction support. (closes issue #10946) Reported by: flefoll
- (closes issue #10915) Reported by: ramonpeek (closes issue #9567)
- Reported by: atca_pres
-
- * channels/chan_sip.c: If callerid is configured in sip.conf use
- that for checking the presence of an extension in the dialplan.
- (closes issue #11185) Reported by: spditner
-
-2007-11-07 23:39 +0000 [r89093] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_queue.c: The member refcount must be incremented, to
- avoid using it after deallocation. A huge thanks go to lvl- for
- patiently providing the necessary valgrind output that was
- necessary to finding this problem of memory corruption. Reported
- by: lvl- Patch by: tilghman Closes issue #11174
-
-2007-11-07 22:40 +0000 [r89090] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: This patch makes it possible for SIP phones
- to dial extensions defined with '#' characters in extensions.conf
- AND maintain their escaped characters when forming URI's (closes
- issue #10681, reported by cahen, patched by me, code review by
- file)
-
-2007-11-07 21:40 +0000 [r89088] Steve Murphy <murf@digium.com>
-
- * cdr/cdr_tds.c, pbx/pbx_ael.c, res/res_jabber.c: In response to
- 10578, I just ran 1.4 thru valgrind; some of the config leakage
- I've already fixed, but it doesn't hurt to double check. I found
- and fixed leaks in res_jabber, cdr_tds, pbx_ael. Nothing major,
- tho.
-
-2007-11-07 15:56 +0000 [r89085] Mark Michelson <mmichelson@digium.com>
-
- * main/manager.c: Fixing a segfault in the manager "core show
- channels concise" command. (closes issue #11183, reported by arnd
- and patched by ys)
-
-2007-11-07 04:07 +0000 [r89079] Tilghman Lesher <tlesher@digium.com>
-
- * configs/extensions.ael.sample: Suppress AEL warnings on load.
- Reported by: eliel Patch by: eliel Closes issue #11178
-
-2007-11-06 20:18 +0000 [r89053] Russell Bryant <russell@digium.com>
-
- * res/res_musiconhold.c: Fix init_classes() so that classes that
- actually do have files loaded aren't treated as empty, and
- immediately destroyed ...
-
-2007-11-06 19:09 +0000 [r89046] Jason Parker <jparker@digium.com>
-
- * codecs/codec_zap.c: Correctly set the total number of channels
- from a zaptel transcoder board. SPD-49, patch by Matthew
- Nicholson.
-
-2007-11-06 19:09 +0000 [r89045] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/lock.h: We went to the trouble of creating a
- method of tracking failed trylocks, then never turned it on
- (oops).
-
-2007-11-06 18:53 +0000 [r89042] Olle Johansson <oej@edvina.net>
-
- * main/tdd.c: Bug fixes to tdd support in zaptel.
-
-2007-11-06 18:20 +0000 [r89037] Russell Bryant <russell@digium.com>
-
- * res/res_musiconhold.c: If someone were to delete the files used
- by an existing MOH class, and then issue a reload, further use of
- that class could result in a crash due to dividing by zero. This
- set of changes fixes up some places to prevent this from
- happening. (closes issue #10948) Reported by: jcomellas Patches:
- res_musiconhold_division_by_zero.patch uploaded by jcomellas
- (license 282) Additional changes added by me.
-
-2007-11-06 17:52 +0000 [r89036] Steve Murphy <murf@digium.com>
-
- * main/config.c: closes issue #8786 - where the [catname](!) and
- [catname](othercat1,othercat2,...) notation gets dropped across a
- ConfigUpdate (or any other thing that would cause a config file
- to be written). While I was at it, I also cleaned up some of the
- destroy routines to free up comments, which was not being done.
- Made sure the new struct I introduced is also cleaned up properly
- at destruction time. My code handles multiple template
- inclusions. Many thanks to ssokol for his patch, which, while not
- literally used in the final merge, served as a foundation for the
- fix.
-
-2007-11-06 17:08 +0000 [r88994-89032] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Make it so that if a peer is determined to
- be unreachable using qualify their devicestate will report back
- unavailable. (closes issue #11006) Reported by: pj
-
- * channels/chan_zap.c: Fix improbable but possible memory leaks in
- chan_zap. (closes issue #11166) Reported by: eliel Patches:
- chan_zap.c.patch uploaded by eliel (license 64)
-
-2007-11-06 13:50 +0000 [r88931] Russell Bryant <russell@digium.com>
-
- * include/asterisk/lock.h: Remove some checks to see if locks are
- initialized from the non-DEBUG_THREADS versions of the lock
- routines. These are incorrect for a number of reasons: - It
- breaks the build on mac. - If there is a problem with locks not
- getting initialized, then the proper fix is to find that place
- and fix the code so that it does get initialized. - If additional
- debug code is needed to help find the problem areas, then this
- type of things should _only_ be put in the DEBUG_THREADS
- wrappers.
-
-2007-11-06 02:52 +0000 [r88862] Kevin P. Fleming <kpfleming@digium.com>
-
- * include/asterisk/srv.h: update comment to match the state of the
- code
-
-2007-11-05 23:29 +0000 [r88826] Mark Michelson <mmichelson@digium.com>
-
- * main/channel.c: Reworked deadlock avoidance in __ast_read.
- Restored audio to callback agents. (closes issue #11071, reported
- by callguy, patched by me, tested by callguy and Ted Brown)
-
-2007-11-05 22:07 +0000 [r88709-88805] Russell Bryant <russell@digium.com>
-
- * main/pbx.c, include/asterisk/pbx.h: After seeing crashes related
- to channel variables, I went looking around at the ways that
- channel variables are handled. In general, they were not handled
- in a thread-safe way. The channel _must_ be locked when reading
- or writing from/to the channel variable list. What I have done to
- improve this situation is to make pbx_builtin_setvar_helper() and
- friends lock the channel when doing their thing. Asterisk API
- calls almost all lock the channel for you as necessary, but this
- family of functions did not. (closes issue #10923, reported by
- atis) (closes issue #11159, reported by 850t)
-
- * channels/chan_sip.c: When traversing the list of channel
- variables here in transmit_invite(), the asterisk channel must be
- locked, as this data may change at any time. (I have seen
- numerous reports of crashes related to the handling of channel
- variables. There are a couple of issues on the bug tracker
- related to it, but it has also been noted on IRC and mailing
- lists. So, I am finding and fixing some places where channel
- variables are handled improperly.)
-
- * channels/chan_sip.c: Fix up some indentation.
-
- * main/srv.c, include/asterisk/srv.h: Merge changes from
- asterisk/team/kpfleming/SRV-priority-handling Previously, the SRV
- record support in Asterisk was broken. There was no guarantee on
- what record Asterisk would choose to actually use. This set of
- changes improves the situation by ensuring that Asterisk will
- choose the highest priority record.
-
- * main/channel.c: Merge the last bit of changes from
- asterisk/team/russell/readq-1.4 The issue here is that the
- channel frame readq handling got broken when the code was
- converted to use the linked list macros. It caused corruption of
- the list head and tail pointers. So, I fixed up the usage of the
- linked list macros and in passing, simplified the code. I also
- documented what the code is doing, as it was a bit difficult to
- figure out at first. This bug showed itself with crashes showing
- messed up head/tail pointers for the readq. However, there are a
- couple of crashes that aren't quite as obvious, but I think may
- be related. So, if your bug gets closed by this commit, but you
- still have a problem, please reopen or create a new bug report.
- (closes issue #10936) (closes issue #10595) (closes issue #10368)
- (closes issue #11084) (closes issue #10040) (closes issue #10840)
-
-2007-11-05 18:47 +0000 [r88671] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: If a SIP channel is put on hold multiple
- times do not keep incrementing the onHold value. (closes issue
- #11085) Reported by: francesco_r Tested by: blitzrage (closes
- issue #10474) Reported by: acennami
-
-2007-11-05 17:46 +0000 [r88624] Russell Bryant <russell@digium.com>
-
- * main/channel.c: Fix up datastore handling in ast_do_masquerade().
- The code is intended to move any channel datastores from the old
- channel to the new one. However, it did not use the linked list
- macros properly to accomplish the task. The existing code would
- only work if there was only a single datastore on the old
- channel.
-
-2007-11-05 17:19 +0000 [r88585] Jason Parker <jparker@digium.com>
-
- * channels/chan_sip.c: Make sure we destroy the config structure on
- configuration failure. Issue 11163, patch by eliel.
-
-2007-11-05 16:20 +0000 [r88539] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_odbc.c: Don't check used pooled connections for
- connection status, as it will cause issues for prepared queries.
- Reported by: Nick Gorham (via -dev list) Patch by: tilghman
-
-2007-11-04 22:38 +0000 [r88471] Luigi Rizzo <rizzo@icir.org>
-
- * include/asterisk/stringfields.h, main/channel.c,
- apps/app_meetme.c, channels/chan_sip.c, channels/chan_iax2.c:
- Rename ast_string_field_free_pool to
- ast_string_field_free_memory, and ast_string_field_free_all to
- ast_string_field_reset_all to avoid misuse (due to too similar
- names and an error in documentation). Fix two related memory
- leaks in app_meetme. No need to merge to trunk, different fix
- already applied there. Not applicable to 1.2
-
-2007-11-02 20:49 +0000 [r88328-88366] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Make subscribecontext behave as advertised.
- It will now look for the presence of a hint in the given context
- (be it subscribecontext or context). (closes issue #10702)
- Reported by: slavon
-
- * channels/chan_sip.c: If an INFO request within a dialog is
- received with a content length of 0 simply send back a 200 OK. It
- is valid to do this and the remote side is probably using it to
- make sure the signalling is still alive. (closes issue #5747)
- Reported by: chandi Patches: infofix-81430-1.patch uploaded by
- IgorG (license 20)
-
-2007-11-02 16:51 +0000 [r88283] Jason Parker <jparker@digium.com>
-
- * main/say.c: We need to make sure to specify a language to
- ast_fileexists, otherwise it may fail for anything besides en
- Issue 11147, fix discovered by both citats and myself
- (independently), with input from Corydon76
-
-2007-11-02 13:03 +0000 [r88116-88210] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/lock.h: Fix build on Solaris Reported by: snuffy
- Patch by: ys Closes issue #11143
-
- * doc/valgrind.txt (added): Add some notes on using valgrind
-
-2007-11-01 16:21 +0000 [r88078] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c: Make sure we set the poll fds to NULL after
- free()ing it. Part of issue 11017, patch by tzafrir.
-
-2007-11-01 13:27 +0000 [r87970-88026] Joshua Colp <jcolp@digium.com>
-
- * apps/app_meetme.c: Fix up commit for my Zap channel with spies in
- Meetme fix. (thanks Tony Mountifield!)
-
- * apps/app_meetme.c: If a Zap channel contains a spy or a spy is
- added take it out of the conference in kernel space and make it
- go through Asterisk so the spy gets audio from both sides.
- (closes issue #10060) Reported by: mparker
-
-2007-10-31 21:23 +0000 [r87906-87908] Jason Parker <jparker@digium.com>
-
- * res/res_jabber.c: Make sure we free some allocated memory before
- returning. Issue 11131, patch by eliel.
-
- * channels/chan_gtalk.c: Don't try to allocate memory that we're
- just going to re-allocate later anyways. Issue 11130, patch by
- eliel.
-
-2007-10-31 18:03 +0000 [r87852] Tilghman Lesher <tlesher@digium.com>
-
- * Makefile: Create samples for ALL of the available options in
- asterisk.conf
-
-2007-10-31 17:49 +0000 [r87775-87849] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_config.c: closes issue #11108 -- where the 'dialplan
- save' cli command saves a file where the semicolon is not
- escaped. Fixed this; User also wanted comments to be preserved
- across dialplan save, but this is impossible at this point in
- time, because comments are not stored in the dialplan. They are
- 'compiled' out of extensions.conf. The only way to preserve those
- comments is to use the config file reader/writer that the GUI
- uses to allow online user edits. extensions.conf is first and
- foremost, a config file, and is read in by the normal config-file
- reading routines. Then, it is processed into a dialplan
- (context/exten structs).
-
- * pbx/pbx_ael.c: Included some verbage in the check_includes func,
- to inform the user that included contexts that have no match in
- the AEL, might be OK, as AEL cannot check in the extensions.conf
- or the in-memory contexts, as they may not be there at the time
- of the check.
-
-2007-10-30 23:02 +0000 [r87739] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/lock.h: Fix for uninitialized mutexes on *BSD
- Reported by: ys Fixed by: ys Closes issue #11116
-
-2007-10-30 21:19 +0000 [r87686] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Merge the changes from
- team/russell/iax2_poke_fix and iax2-poke-fix-trunk There was a
- race condition related to the handling of POKEing peers.
- Essentially, a reference to a peer is held by the scheduler when
- there are pending callbacks, but the reference count didn't
- reflect it. So, it was possible for a peer to hit a reference
- count of zero and have its destructor begin to be called at the
- same time that the scheduler thread ran a POKE related callback.
- If that happened, a crash would likely occur. (closes issue
- #11082, closes issue #11094)
-
-2007-10-30 20:29 +0000 [r87650] Jason Parker <jparker@digium.com>
-
- * channels/Makefile: Only try to clean out h323/ if the
- h323/Makefile exists.
-
-2007-10-30 16:13 +0000 [r87571] Joshua Colp <jcolp@digium.com>
-
- * res/res_features.c: Add two more checks before printing out a
- warning message about bridging. If either channel has hungup of
- course the bridge will have failed. (closes issue #10009)
- Reported by: dimas
-
-2007-10-30 15:45 +0000 [r87567] Jason Parker <jparker@digium.com>
-
- * main/editline/np/vis.c: Fix build of editline on Solaris. Issue
- 11113, patch by snuffy.
-
-2007-10-30 15:10 +0000 [r87534] Joshua Colp <jcolp@digium.com>
-
- * apps/app_followme.c: Return 1.4 to a state where it builds.
- Changing the arguments to a function and not changing where they
- are used is bad, mmmk?
-
-2007-10-30 14:31 +0000 [r87514] BJ Weschke <bweschke@btwtech.com>
-
- * apps/app_followme.c: Fix issue where the recorded name wasn't
- getting removed correctly. (closes issue #11115) Reported by:
- davevg Patches: followme-v3.diff
-
-2007-10-29 22:13 +0000 [r87460-87465] Kevin P. Fleming <kpfleming@digium.com>
-
- * codecs/gsm: missed one directory
-
- * codecs/ilbc, formats, utils/Makefile, agi/Makefile, funcs,
- codecs/lpc10, main/db1-ast, main/editline, main,
- codecs/ilbc/Makefile, pbx, res, channels, main/db1-ast/Makefile,
- codecs/lpc10/Makefile, utils, codecs, agi,
- main/editline/Makefile.in, apps, Makefile.moddir_rules, cdr:
- clean up (and ignore) assembler and preprocessor intermediate
- files if any are created during the build
-
- * Makefile: don't put '-pipe' into ASTCFLAGS if '-save-temps' is
- already there (used when debugging preprocessor issues) because
- the compiler will whine about each compile command
-
-2007-10-29 21:06 +0000 [r87427] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Removing a completely unnecessary quota
- check from IMAP code.
-
-2007-10-29 20:22 +0000 [r87373-87396] Russell Bryant <russell@digium.com>
-
- * main/utils.c, include/asterisk/lock.h: Add some more details to
- the output of "core show locks". When a thread is waiting for a
- lock, this will now show the details about who currently has it
- locked. (inspired by issue #11100)
-
- * main/astmm.c: Remove a lock that doesn't make any sense. The
- regions lock needs to be held when traversing the list of
- allocated chunks so that they can be printed out to the CLI.
- (Thanks to eliel on #asterisk-dev for pointing this out!)
-
-2007-10-29 17:20 +0000 [r87342] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Fix issue where if both sides of the dialog
- cancelled the dialog at the same time chan_sip could kepe
- retransmitting a response for no reason. (closes issue #9566)
- Reported by: atca_pres Patches: bug9566.patch uploaded by oej
-
-2007-10-29 17:13 +0000 [r87340] Jason Parker <jparker@digium.com>
-
- * funcs/func_realtime.c, funcs/func_cut.c: Allow some function
- modules to compile under dev mode. Issue 11104, patch by andrew.
-
-2007-10-29 14:23 +0000 [r87294] Joshua Colp <jcolp@digium.com>
-
- * main/utils.c: Fix issue with ast_unescape_semicolon going into an
- endless loop. (closes issue #10550) Reported by: ramonpeek
- Patches: unescape-85177-1.patch uploaded by IgorG (license 20)
-
-2007-10-28 13:46 +0000 [r87262] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_realtime.c, funcs/func_odbc.c, funcs/func_strings.c,
- funcs/func_cut.c: Add autoservice to several more functions which
- might delay in their responses. Also, make sure that func_odbc
- functions have a channel on which to set variables. Reported by
- russell Fixed by tilghman Closes issue #11099
-
-2007-10-26 16:34 +0000 [r87168] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael-test/ref.ael-test19, pbx/ael/ael.tab.c,
- pbx/ael/ael.y, pbx/ael/ael_lex.c, pbx/pbx_ael.c,
- include/asterisk/ael_structs.h, pbx/ael/ael.tab.h,
- utils/ael_main.c, pbx/ael/ael-test/ref.ael-test16,
- pbx/ael/ael.flex: closes issue #11086 where a user complains that
- references to following contexts report a problem; The problem
- was REALLy that he was referring to empty contexts, which were
- being ignored. Reporter stated that empty contexts should be OK.
- I checked it out against extensions.conf, and sure enough, empty
- contexts ARE ok. So, I removed the restriction from AEL. This,
- though, highlighted a problem with multiple contexts of the same
- name. This should be OK, also. So, I added the extend keyword to
- AEL, and it can preceed the 'context' keyword (mixed with
- 'abstract', if nec.). This will turn off the warnings in AEL if
- the same context name is used 2 or more times. Also, I now call
- ast_context_find_or_create for contexts now, instead of just
- ast_context_create; I did this because pbx_config does this. The
- 'extend' keyword thus becomes a statement of intent. AEL can now
- duplicate the behavior of pbx_config,
-
-2007-10-26 13:54 +0000 [r87120] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_curl.c: The addition of autoservice to func_curl
- additionally made func_curl dependent on the existence of a
- channel, with no real reason. This should make func_curl once
- again work without a channel. Reported by jmls. Fixed by
- tilghman. Closes issue #11090
-
-2007-10-25 23:03 +0000 [r87069] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/channel.c, include/asterisk/linkedlists.h: appending one
- list to another should leave the first list empty, and not
- require the user to do that
-
-2007-10-25 22:53 +0000 [r87067] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_cut.c: Backport alternate encoding of newline
- delimiters from trunk to 1.4, as approved by Russell Reported by
- blitzrage Closes issue #10903
-
-2007-10-24 20:56 +0000 [r86982] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c: Correctly respect hidecalleridname
- configuration option. Simplify code slightly in the process.
- Issue 11079, reported by ddv2005
-
-2007-10-24 04:14 +0000 [r86880-86936] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael.tab.c, pbx/ael/ael.y: closes issue #11037 -- unable
- to specify app:spec in hint arguments
-
- * funcs/func_logic.c: closes issue #11052 -- where nothing after
- the ? will allow un-initialized variable values to corrupt and
- crash asterisk on 64-bit platforms
-
- * main/Makefile: this update to Makefile corrects how ast_expr2f.c
- should be generated
-
- * main/ast_expr2f.c: This should get rid of a really, really
- irritating warning generated by some 64-bit platforms from libc,
- where free(0) is frowned upon
-
-2007-10-22 21:36 +0000 [r86836] Russell Bryant <russell@digium.com>
-
- * include/asterisk/lock.h: If lock tracking is not enabled, then we
- can not attempt to log any mutex failures. If so, we could end up
- in infinite recursion. The only lock that is affected by this is
- a mutex in astmm.c used when MALLOC_DEBUG is enabled. (closes
- issue #11044) Reported by: ys Patches: lock.h.diff uploaded by ys
- (license 281)
-
-2007-10-22 17:38 +0000 [r86787] Tilghman Lesher <tlesher@digium.com>
-
- * main/astmm.c: Minor FreeBSD build fix
-
-2007-10-22 16:35 +0000 [r86754-86756] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: After reading online I have confirmed that
- Record-Route headers should be copied to 1xx responses as well.
- (closes issue #10113) Reported by: makoto
-
- * apps/app_controlplayback.c: Make sure res is a positive value
- before performing the check to determine whether the user stopped
- it or not. (closes issue #11023) Reported by: cfc
-
-2007-10-22 15:52 +0000 [r86726-86750] Russell Bryant <russell@digium.com>
-
- * main/channel.c: Don't leak a frame in the case that an END frame
- is received and the time since the BEGIN is less than that of the
- defined minimum DTMF duration. (closes issue #11051) Reported by:
- casper Patches: channel.c.86664.diff uploaded by casper (license
- 55)
-
- * include/asterisk/lock.h: Update the static mutex initializer to
- include the initialization of the internal mutex used to protect
- the lock debugging data. (closes issue #11044, patch suggested by
- Ivan)
-
-2007-10-22 14:48 +0000 [r86694] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Account for the fact that sometimes headers
- may be terminated with \r\n instead of just \n (closes issue
- #11043, reported by yehavi)
-
-2007-10-22 14:27 +0000 [r86630-86663] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: Move log message to before the frame it
- references is freed. (closes issue #11050) Reported by: slavon
- Patches: channel.c.86662.diff uploaded by casper (license 55)
-
- * pbx/pbx_dundi.c: Fix tab completion for dundi show peer. (closes
- issue #11041) Reported by: jsmith Patches:
- asterisk-dundicomplete.diff.txt uploaded by jamesgolovich
- (license 176)
-
- * main/loader.c: Fixes for building under OpenSolaris. (closes
- issue #11047) Reported by: snuffy Patches: 11047-fixes.diff
- uploaded by snuffy (license 35)
-
-2007-10-22 09:21 +0000 [r86598] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/chan_misdn.c: we send
- DISCONNECT instead of RELEASE/RELEASE_COMPLETE if the dialplan
- does not match after an overlap call. Also added out_cause=1
-
-2007-10-19 16:38 +0000 [r86469-86502] Joshua Colp <jcolp@digium.com>
-
- * main/app.c: When returning a DTMF digit from
- ast_control_streamfile cast it as a char so that 0 does not
- overlap with the success return code. (closes issue #11023)
- Reported by: cfc
-
- * channels/chan_sip.c: Fix two issues with domains and transfers.
- If a port was given in the hostname it was treated as part of the
- hostname. If domains were configured but external domains were
- not enabled all transfers would be considered remote. (closes
- issue #11027) Reported by: ramonpeek Patches: 11027-1.diff
- uploaded by ramonpeek (license 266)
-
- * channels/chan_sip.c: Set port number in received as information
- for registrations as well. (closes issue #11028) Reported by:
- brad-x
-
-2007-10-19 01:45 +0000 [r86438] TransNexus OSP Development <support@transnexus.com>
-
- * apps/app_osplookup.c: Fixed OSP module did not report
- source/devinfo IP in correct format.
-
-2007-10-18 22:01 +0000 [r86405-86406] Jason Parker <jparker@digium.com>
-
- * Makefile: Correct documentation. I removed the wrong line..
-
- * Makefile: Add documentation for options in asterisk.conf Issue
- 11029, patch by eserra
-
-2007-10-18 21:16 +0000 [r86330-86372] Russell Bryant <russell@digium.com>
-
- * configs/iax.conf.sample, channels/chan_iax2.c: Revert erroneous
- commit.
-
- * configs/iax.conf.sample, channels/chan_iax2.c: Add support for
- setting the maximum trunk size for IAX2 trunking
-
- * main/channel.c, include/asterisk/channel.h: The channel needs to
- stay locked while running timer callbacks, as they access and
- modify channel data that may change elsewhere. I went through
- every timer callback in the source tree to make sure that none of
- them did any additional locking that could introduce deadlocks,
- and all is well. (closes issue #10765) Reported by: Ivan Patches:
- ast_1_4_11_svn_patch_channel_rc.diff uploaded by Ivan (license
- 229)
-
-2007-10-18 17:38 +0000 [r86328] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: If a non-existent file is specified to be
- played either as a periodic announcement or as a hold/position
- announcement, the caller would be kicked out of the queue. No
- longer does this happen.
-
-2007-10-18 15:45 +0000 [r86237-86296] Russell Bryant <russell@digium.com>
-
- * codecs/codec_zap.c: Execute the RELEASE operation on transcoder
- channels in the destroy callback. (patch from jsloan)
-
- * main/utils.c: Revert a change that I made for issue #10979 which,
- as has been pointed out to me in issue #11018, doesn't really
- make sense. There is no reason to have the base64 decode function
- force a '\0' terminated buffer, when the result is almost always
- binary, anyway. In fact, this caused some breakage, as some code
- in res_crypto passed in a buffer exactly the right size to get
- its binary result, which got stomped on by this patch. (closes
- issue #11018, reported by dimas)
-
-2007-10-17 21:39 +0000 [r86202] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Changing the strategy field of the call_queue
- struct to be signed instead of unsigned, since the code attempts
- to set the strategy to -1 if you specify a bogus strategy. While
- this isn't a huge issue in 1.4, it could be a problem for someone
- who, say, tries to use the roundrobin strategy in trunk (despite
- all the deprecation warnings in 1.4).
-
-2007-10-17 17:57 +0000 [r86149] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: If Asterisk is in the middle of shutting
- down, respond to OPTIONS with 503 Unavailable. (closes issue
- #10994) Reported by: eserra Patches: sip-options-503.patch
- uploaded by eserra (license 45)
-
-2007-10-17 16:58 +0000 [r86117] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Whoops, forgot to remove the original
- sip_scheddestroy. (closes issue #11010) Reported by: vadim
-
-2007-10-17 15:23 +0000 [r86066] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c: When runuser/rungroup is specified, a remote
- console could only be attained by root (Closes issue #9999)
-
-2007-10-17 15:06 +0000 [r86063] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Don't schedule dialog destruction if a
- MESSAGE is received using an existing dialog. (closes issue
- #11010) Reported by: vadim
-
-2007-10-16 23:35 +0000 [r86028-86032] Mark Michelson <mmichelson@digium.com>
-
- * configs/queues.conf.sample: Since monitor-join is deprecated now,
- remove the example from the sample queues.conf file
-
- * UPGRADE.txt: Updating UPGRADE.txt to reflect the deprecation of
- the monitor-join queue option
-
- * apps/app_queue.c: Adding deprecated warning to monitor-join
- option, since the plan is to no longer support this in favor of
- monitor-type = mixmonitor (related to issue #10885)
-
-2007-10-16 22:36 +0000 [r85994-85997] Russell Bryant <russell@digium.com>
-
- * include/asterisk/lock.h: really picky formatting tweak ...
-
- * include/asterisk/lock.h: Some locking errors exposed the fact
- that the lock debugging code itself was not thread safe. How
- ironic! Anyway, these changes ensure that the code that is
- accessing the lock debugging data is thread-safe. Many thanks to
- Ivan for finding and fixing the core issue here, and also thanks
- to those that tested the patch and provided test results. (closes
- issue #10571) (closes issue #10886) (closes issue #10875) (might
- close some others, as well ...) Patches: (from issue #10571)
- ivan_ast_1_4_12_rel_patch_lock.h.diff uploaded by Ivan (license
- 229) - a few small changes by me
-
-2007-10-16 21:14 +0000 [r85958] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Trying to remove a non-dynamic queue member via
- dynamic means can lead to some interesting (read nasty)
- situations. This patch clears up the issue by making only dynamic
- queue members removable via dynamic methods.
-
-2007-10-16 19:41 +0000 [r85921] Tilghman Lesher <tlesher@digium.com>
-
- * main/stdtime/localtime.c: Also set up gmtoff (this is used in the
- %z gnu extension to strftime) Reported and fixed by jcmoore
- Closes issue #11002
-
-2007-10-16 19:10 +0000 [r85896] Russell Bryant <russell@digium.com>
-
- * apps/app_voicemail.c: Remove a pointless lock.
-
-2007-10-16 15:21 +0000 [r85852] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Fixing a double free which happens in the
- statechange thread. (closes issue #10987, reported by andrew)
-
-2007-10-16 14:52 +0000 [r85818-85850] Joshua Colp <jcolp@digium.com>
-
- * apps/app_hasnewvoicemail.c: Check to make sure a value has been
- given to the VMCOUNT dialplan function. (closes issue #10996)
- Reported by: marsosa
-
- * main/threadstorage.c: Fix memory allocation issue in
- threadstorage. (closes issue #10995) Reported by: snuffy Patches:
- new-patch.diff uploaded by snuffy (license 35)
-
-2007-10-16 10:46 +0000 [r85800] Philippe Sultan <philippe.sultan@gmail.com>
-
- * channels/chan_gtalk.c: Fix the output for this channel help CLI
- command
-
-2007-10-15 21:10 +0000 [r85717-85720] Russell Bryant <russell@digium.com>
-
- * apps/app_queue.c: Ensure that no pending state changes are leaked
- when the device state change thread gets stopped on module
- unload.
-
- * apps/app_queue.c: Previously, app_queue created a thread to
- handle every single device state change. I changed this a while
- ago in trunk for performance reasons. However, bug 8407 points
- out that it is actually a race condition, causing device state
- changes to get processed in random order. So, I backported my
- changes from trunk to 1.4. (closes issue #8407, patch provided by
- tim_ringenbach, committed patch by me)
-
-2007-10-15 20:29 +0000 [r85687] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_stack.c: Don't execute a gosub if the arguments is
- zero-len (not just NULL) Reported by davevg Fixed by me Closes
- issue #10985
-
-2007-10-15 20:21 +0000 [r85686] Russell Bryant <russell@digium.com>
-
- * main/say.c: Add a small fix for the tw version of saying dates.
- (closes issue #7827) Reported by: sharkey Patches: say.nits.patch
- uploaded by sharkey (license 172)
-
-2007-10-15 20:15 +0000 [r85684] Jason Parker <jparker@digium.com>
-
- * Makefile: Properly use DESTDIR in 'config' target. Do not try to
- run chkconfig or similar if using DESTDIR. Issue 10938, patch by
- cabal95.
-
-2007-10-15 19:22 +0000 [r85604-85649] Russell Bryant <russell@digium.com>
-
- * main/utils.c: Be pedantic about handling memory allocation
- failure.
-
- * main/utils.c: The loop in the handler for the "core show locks"
- could potentially block for some amount of time. Be a little bit
- more careful and prepare all of the output in an intermediary
- buffer while holding a global resource. Then, after releasing it,
- send the output to ast_cli().
-
- * channels/chan_sip.c: Make the default for the srvlookup option to
- be yes. It doesn't really make sense for it to default to off.
- The default configuration file has it on, and proper RFC
- behavior, as indicated by a comment in the code, is for it to be
- on. So, let's have it on by default to make lives easier. (closes
- issue #10954, suggested by jtodd)
-
-2007-10-15 16:39 +0000 [r85571] Joshua Colp <jcolp@digium.com>
-
- * configs/features.conf.sample: Document that DTMF based features
- only work when two channels are bridged together. (closes issue
- #10773) Reported by: pbayley
-
-2007-10-15 16:34 +0000 [r85561] Russell Bryant <russell@digium.com>
-
- * include/asterisk/strings.h: Make a few changes so that characters
- in the upper half of the ISO-8859-1 character set don't get
- stripped when reading configuration. (closes issue #10982,
- dandre)
-
-2007-10-15 16:22 +0000 [r85559] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Bring both DTMF begin and end frames up through to
- the core for DTMF feature handling. (closes issue #10826)
- Reported by: dimas
-
-2007-10-15 15:40 +0000 [r85556] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_dundi.c: Ensure the buffer passed to
- ast_canmatch_extension() is properly initialized so that it is
- null terminated. (issue #10977) Reported by: dimas Patches:
- pbxdundi.patch uploaded by dimas (license 88) - small mods by me
-
-2007-10-15 14:55 +0000 [r85552] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: If Monitor or a spy was added to a P2P or native
- bridged channel bring the channel back to the generic bridging
- core so the monitor or spy operations work. (closes issue #10943)
- Reported by: julianjm
-
-2007-10-15 13:16 +0000 [r85540-85548] Russell Bryant <russell@digium.com>
-
- * main/db.c: Suppress a LOG_DEBUG message if debug is not enabled.
- (closes issue #10980) Reported by: casper Patches:
- db.c.84633.diff uploaded by casper (license 55)
-
- * main/asterisk.c: Make sure remote consoles unmute themselves
- again after reconnecting. (closes issue #10847) Reported by: atis
- Patches: console_unmute_on_reconnect.patch uploaded by atis
- (license 242)
-
- * main/utils.c: Make sure that the base64 decoder returns a
- terminated string. (closes issue #10979) Reported by: ys Patches:
- util.c.diff uploaded by ys (license 281) - small mods by me
-
- * pbx/pbx_config.c: Don't create the context for users in
- users.conf until we know at least one user exists. (closes issue
- #10971) Reported by: dimas Patches: pbxconfig.patch uploaded by
- dimas (license 88)
-
-2007-10-13 15:26 +0000 [r85536] Tilghman Lesher <tlesher@digium.com>
-
- * configs/extensions.ael.sample: Remove deprecated syntax from
- sample ael file Reported and patched by: dimas Closes issue
- #10967
-
-2007-10-13 05:48 +0000 [r85532-85533] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c, main/cli.c, include/asterisk/logger.h: Fix an
- issue with console verbosity when running asterisk -rx to execute
- a command and retrieve its output. The issue was that there was
- no way for the main Asterisk process to know that the remote
- console was connecting in the -rx mode. The way that James has
- fixed this is to have all remote consoles muted by default. Then,
- regular remote consoles automatically execute a CLI command to
- unmute themselves when they first start up. (closes issue #10847)
- Reported by: atis Patches: asterisk-consolemute.diff.txt uploaded
- by jamesgolovich (license 176)
-
- * main/asterisk.c, main/cli.c, include/asterisk/cli.h: Properly
- handle the case where read() may return the text for more than
- one CLI command at once for a remote console. (closes issue
- #10888) Reported by: jamesgolovich Patches:
- asterisk-climultiple.diff.txt uploaded by jamesgolovich (license
- 176)
-
-2007-10-12 18:30 +0000 [r85523] Tilghman Lesher <tlesher@digium.com>
-
- * doc/asterisk-mib.txt, doc/PEERING, LICENSE: Change Digium address
-
-2007-10-12 15:45 +0000 [r85515-85517] Russell Bryant <russell@digium.com>
-
- * res/res_smdi.c: Fix a spelling error in a log message. SMDI, not
- SDMI. (closes issue #10959)
-
- * pbx/pbx_realtime.c: Fix the potential use of an uninitialized
- buffer in a log message. (closes issue #10958) Reported by: dimas
- Patches: realtime.patch uploaded by dimas (license 88)
-
-2007-10-11 15:26 +0000 [r85397] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: When creating a new packet don't try to stop
- retransmission of it. It was just allocated/created so it's
- impossible for it to have already been scheduled. (closes issue
- #10945) Reported by: flefoll Patches:
- chan_sip.c.br14.85280.xmit_reliable-patch uploaded by flefoll
- (license 244)
-
-2007-10-11 04:35 +0000 [r85356] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c: A dollar sign by itself, not indicating a start of a
- variable or expression prematurely ends substitution (closes
- issue #10939)
-
-2007-10-10 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.13 released.
-
-2007-10-10 15:56 +0000 [r85316] Russell Bryant <russell@digium.com>
-
- * include/asterisk/file.h: I introduced a new member to the
- ast_filestream struct in 1.4.12, but put it in the middle of the
- struct, instead of at the end. One of the Debian folks, paravoid,
- pointed out that this breaks binary compatability with modules
- compiled against older headers. So, I'm moving the new member to
- the end of the struct to resolve the situation.
-
-2007-10-10 15:51 +0000 [r85315] Mark Michelson <mmichelson@digium.com>
-
- * main/utils.c: The thread ID should be unsigned.
-
-2007-10-10 14:42 +0000 [r85277-85280] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: If devicestate is passed a port number strip
- it out. (closes issue #10930) Reported by: ibc
-
- * channels/chan_sip.c: Add support for handling a 182 Queued
- response. (closes issue #10924) Reported by: ramonpeek Patches:
- queued-182.diff uploaded by ramonpeek (license 266)
-
-2007-10-10 14:26 +0000 [r85276] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: A bunch of changes from sprintf to
- snprintf. See security advisory AST-2002-022
-
-2007-10-10 14:14 +0000 [r85242] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c: Close voicemail message description file if
- duration did not meet the minimum, or else we will eventually run
- out of file descriptors. (closes issue #10918) Reported by:
- brak2718 Patches: vm1.4.12.1.patch uploaded by brak2718 (license
- 279)
-
-2007-10-10 06:24 +0000 [r85195] Kevin P. Fleming <kpfleming@digium.com>
-
- * include/asterisk/frame.h: use a macro instead of an inline
- function, so that backtraces will report the caller of
- ast_frame_free() properly
-
-2007-10-09 21:55 +0000 [r85158] Tilghman Lesher <tlesher@digium.com>
-
- * main/channel.c, main/utils.c, include/asterisk/lock.h: This
- commit fixes the following issues: - Deadlock in ast_write (issue
- #10406) - Deadlock in ast_read (issue #10406) - Possible mutex
- initialization error in lock.h (issue #10571)
-
-2007-10-09 14:30 +0000 [r84990-85093] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Don't perform a reinvite if a transfer is in
- progress. (issue #10915) Reported by: ramonpeek
-
- * main/rtp.c: Only update codec information if the channel has a
- technology private structure. (issue #10915) Reported by:
- ramonpeek
-
- * main/rtp.c: Update codec information as well as address when
- doing hold reinvites. (issue #10868) Reported by: mavince
-
- * main/channel.c: Don't keep trying to native bridge if either of
- the channels are involved in a masquerade operation to be done.
- (closes issue #10696) Reported by: tbelder
-
-2007-10-08 03:28 +0000 [r84957] Russell Bryant <russell@digium.com>
-
- * Makefile.rules: Enable file dependency tracking for _all_ builds,
- and not just for builds with dev-mode enabled. I have seen enough
- problems caused by this that I don't think it's worth keeping. I
- want to continue to encourage anybody that is interested to
- continue to run Asterisk from svn. Furthermore, I do not want
- their systems to break when we change a structure definition in a
- header file. :)
-
-2007-10-07 16:15 +0000 [r84890-84902] Philippe Sultan <philippe.sultan@gmail.com>
-
- * res/res_jabber.c: Presence packets from a client who's connected
- with our Jabber ID are valid, therefore, those clients must be
- considered as buddies. The resource string helps us make the
- distinction between clients. Closes issue #10707, reported by
- yusufmotiwala.
-
- * res/res_jabber.c: Prevent Asterisk from crashing when receiving a
- presence packet without resource from a buddy that is known to
- have a resource list. Revert a change I previously made, where
- Asterisk could point to a freed memory location.
-
-2007-10-05 19:42 +0000 [r84851] Tilghman Lesher <tlesher@digium.com>
-
- * main/db.c: Log exactly why we can't open the database, if we fail
- (closes issue #10887)
-
-2007-10-05 18:55 +0000 [r84818] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Update the remembered RTP peer information when
- putting an endpoint on hold or taking it off hold so that the RTP
- stack does not initiate a needless reinvite. (closes issue
- #10868) Reported by: mavince
-
-2007-10-05 16:44 +0000 [r84783] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c: Do deadlock avoidance in a couple more
- places. You can't lock two channels at the same time without
- doing extra work to make sure it succeeds. (closes issue #10895,
- patch by me)
-
-2007-10-05 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.12.1 released. (This is mainly to include the
- app_queue fix for a memory leak on reload, but includes a couple
- of other bug fixes, as well.)
-
-2007-10-05 01:39 +0000 [r84742] Russell Bryant <russell@digium.com>
-
- * main/manager.c: Fix a copy/paste error in the description of
- UpdateConfig that was pointed out by JerJer on #asterisk-dev
-
-2007-10-04 21:57 +0000 [r84692] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Don't allocate space for queue members unless
- it's needed. You end up deleting dynamic members on a reload. Not
- good. closes issue (#10879, reported by dazza76, patched by me)
-
-2007-10-04 21:36 +0000 [r84690] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c: callers of sig2str already add the word
- 'signalling' in the appropriate place, so don't duplicate it
-
-2007-10-04 14:51 +0000 [r84637] Joshua Colp <jcolp@digium.com>
-
- * apps/app_queue.c: Create a duplicate of the channel's member name
- as the tab completion stuff will free it. (closes issue #10884)
- Reported by: adamg
-
-2007-10-03 22:59 +0000 [r84581] Tilghman Lesher <tlesher@digium.com>
-
- * main/rtp.c: When an RFC 2833 event is sent that we don't
- recognize, ignore it, don't queue a NULL digit (closes issue
- #10877)
-
-2007-10-03 18:20 +0000 [r84511-84544] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_ael.c: closes issue #10870 ; where a CUT() function call
- in a switch expr doesn't execute correctly, because the commas in
- the function args are not converted to vertbars before the func
- is called. I modified just the switch code to convert the commas
- to vertbars if there, but if more of these sort of probs are
- found, I may have to resort to something a little more
- fundamental. We'll see, I guess.
-
- * pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18,
- pbx/ael/ael-test/ref.ael-vtest13,
- pbx/ael/ael-test/ref.ael-vtest17,
- pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
- pbx/ael/ael-test/ref.ael-test3, pbx/pbx_ael.c,
- pbx/ael/ael-test/ref.ael-test5: closes issue #10834 ; where a
- null input to a switch statement results in a hangup; since
- switch is implemented with extensions, and the default case is
- implemented with a '.', and the '.' matches 1 or more remaining
- characters, the case where 0 characters exist isn't matched, and
- the extension isn't matched, and the goto fails, and a hangup
- occurs. Now, when a default case is generated, it also generates
- a single fixed extension that will match a null input. That
- extension just does a goto to the default extension for that
- switch. I played with an alternate solution, where I just tack an
- extra char onto all the patterns and the goto, but not the
- default case's pattern. Then even a null input will still have at
- least one char in it. But it made me nervous, having that extra
- char in , even if that's a pretty secret and low-level issue.
-
-2007-10-02 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.12 released.
-
-2007-10-02 20:06 +0000 [r84474] Russell Bryant <russell@digium.com>
-
- * Makefile, build_tools/prep_tarball: * Don't build the
- menuselect-tree for the tarball, as it requires running the
- configure script first * Change the Makefile to note that
- menuselect-tree depends on the configure script.
-
-2007-10-02 19:01 +0000 [r84410-84437] Jason Parker <jparker@digium.com>
-
- * res/res_features.c: Fix some odd formatting I missed..
-
- * res/res_features.c: Finish up on transferee channel before return
- on failure. Issue 10821, patch by Ivan
-
-2007-10-02 14:12 +0000 [r84370] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Use snprintf instead of sprintf in one
- place. There is no vulnerability here due to various buffer sizes
- around the code, but I still didn't like seeing a non
- length-limited copy of data coming off of the wire into a stack
- buffer, as this would be a problem in the future if buffer sizes
- elsewhere got changed or size limitations removed ...
-
-2007-10-02 09:48 +0000 [r84345] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c: terminate USERUSER String with 0
-
-2007-10-01 21:52 +0000 [r84291] Jason Parker <jparker@digium.com>
-
- * Makefile, Makefile.rules, channels/Makefile: Add dist-clean
- support for subdirs. Change h323 to only remove the Makefile on a
- dist-clean, rather than a clean. This fixes a bug I found with
- trying to run make after a make clean
-
-2007-10-01 21:25 +0000 [r84274] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * main/channel.c, main/manager.c, channels/chan_agent.c: moved
- get_base_channel() code from action_redirect to
- ast_channel_masquerade() for issue 7706 and BE-160
-
-2007-10-01 21:18 +0000 [r84273] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_ael.c: Anything to keep gcc 4.2 happy...
-
-2007-10-01 21:07 +0000 [r84271] Russell Bryant <russell@digium.com>
-
- * main/utils.c, include/asterisk/lock.h: Fulfull a feature request
- from Qwell on the "core show locks" output. It will now note the
- lock type for each lock that a thread holds. (mutex, rdlock, or
- wrlock)
-
-2007-10-01 20:27 +0000 [r84239] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/pbx_ael.c: closes issue
- #10777 -- by returning a null for the parse tree when there's
- really nothing there, and making sure we don't try to do checking
- on a null tree.
-
-2007-10-01 19:56 +0000 [r84166-84236] Russell Bryant <russell@digium.com>
-
- * res/res_agi.c: Add another sanity check in the AGI read loop. We
- really don't care about EAGAIN unless we didn't read an entire
- line. If there is a newline at the end if the read buffer, break,
- because we got the whole thing. (reported and patched by bmd)
-
- * include/asterisk/lock.h: Show rwlocks in the "core show locks"
- output. Before, it only showed mutexes.
-
- * channels/Makefile: Remove another file in "make clean". (closes
- issue #10814, paravoid)
-
- * apps/app_dial.c: Simplify the CAN_EARLY_BRIDGE macro a bit.
-
-2007-10-01 14:10 +0000 [r84158-84163] Joshua Colp <jcolp@digium.com>
-
- * configs/usbradio.conf.sample (removed): Remove chan_usbradio
- config file from tree, it is not present in here. (closes issue
- #10839) Reported by: casper
-
- * res/res_musiconhold.c: Fix randomness. save_pos was being set to
- 0 initially instead of -1, causing it to jump to position 0 when
- moh started. (closes issue #10859) Reported by: jamesgolovich
- Patches: asterisk-mohpos2.diff.txt uploaded by jamesgolovich
- (license 176)
-
- * apps/app_dial.c: Only attempt early bridging if the options given
- to Dial() permit it. (closes issue #10861) Reported by: peekyb
-
-2007-09-30 20:02 +0000 [r84146] Russell Bryant <russell@digium.com>
-
- * include/asterisk/module.h: Fix the AST_MODULE_INFO macro for C++
- modules. The load and reload parameters were in the wrong place.
- (closes issue #10846, alebm)
-
-2007-09-29 23:00 +0000 [r84133-84135] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael-test/ael-ntest22/t1/a.ael (added),
- pbx/ael/ael-test/ael-ntest22/t1/b.ael (added),
- pbx/ael/ael-test/ael-ntest22/t1/c.ael (added),
- pbx/ael/ael-test/ael-ntest22/t2/d.ael (added),
- pbx/ael/ael-test/ael-ntest22/t2/e.ael (added),
- pbx/ael/ael-test/ael-ntest22/t2/f.ael (added),
- pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-ntest22
- (added), pbx/ael/ael-test/ael-ntest22/t3/g.ael (added),
- pbx/ael/ael-test/ref.ael-test3,
- pbx/ael/ael-test/ael-ntest22/t3/h.ael (added),
- pbx/ael/ael-test/ref.ael-test4,
- pbx/ael/ael-test/ael-ntest22/t3/i.ael (added),
- pbx/ael/ael-test/ael-ntest22/t3/j.ael (added),
- pbx/ael/ael-test/ael-ntest22/qq.ael (added),
- pbx/ael/ael-test/ael-ntest22/t1 (added),
- pbx/ael/ael-test/ael-ntest22/t2 (added),
- pbx/ael/ael-test/ael-ntest22/t3 (added),
- pbx/ael/ael-test/ael-ntest22/extensions.ael (added),
- pbx/ael/ael-test/ael-ntest22 (added): This is a regression update
- that matches what I did in 84134 for AEL regressions.
-
- * pbx/ael/ael_lex.c, pbx/ael/ael.flex: This issue sort of closes
- 10786; All config files support #include with globbing (you know,
- *,[chars],?,{list,list},etc), so I've updated the AEL system to
- support this also.
-
-2007-09-28 14:13 +0000 [r84049-84078] Tilghman Lesher <tlesher@digium.com>
-
- * main/say.c: Correct pronunciations of numbers for .nl (Closes
- issue #10837)
-
- * main/channel.c: Avoid a deadlock with ALL of the locks in the
- masquerade function, not just the pairs of channels. (Closes
- issue #10406)
-
-2007-09-27 23:12 +0000 [r84018] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * main/manager.c, channels/chan_agent.c,
- include/asterisk/channel.h: if an Agent is redirected, the base
- channel should actually be redirected. This was causing multiple
- issues, especially issue 7706 and BE-160
-
-2007-09-27 00:01 +0000 [r83976] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_dundi.c: remove a todo item that has been completed
-
-2007-09-26 23:53 +0000 [r83974] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_alsa.c: avoid the weird usage of assert() in the
- ALSA header files that gcc 4.2 wants to complain about
-
-2007-09-26 21:35 +0000 [r83910-83943] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: I changed my mind ... I think this should be
- a LOG_NOTICE.
-
- * channels/chan_sip.c: Add a log message that was requested by the
- masses in the developer tutorial session at Astricon. chan_sip
- did not output any message when a call was rejected because the
- extension was not found. This adds a verbose message (at verbose
- level 3) to note when this happens.
-
- * channels/chan_misdn.c: Fix building chan_misdn under dev-mode.
- (please run the configure script with --enable-dev-mode so this
- doesn't happen again ...)
-
-2007-09-26 18:35 +0000 [r83879] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_zap.c: Remove unused 4k of memory on the program
- stack (closes issue #10827)
-
-2007-09-25 14:13 +0000 [r83637-83773] Tilghman Lesher <tlesher@digium.com>
-
- * main/app.c: jmls pointed out that unsetting the group and setting
- the group to the blank string aren't quite the same.
-
- * build_tools/make_defaults_h: In the source, keys are relative to
- the datadir, not varlib (which is the same in most cases, but
- it's good to be accurate). Closes issue #10811
-
- * doc/realtime.txt: Oops. Removed the unworkable workaround. This
- note should never have been in the release.
-
- * main/app.c: Making change to group splitting, as discussed on the
- -dev list. The main effect of this will be to permit
- Set(GROUP([cat])=), i.e. unsetting a group.
-
-2007-09-24 07:54 +0000 [r83620] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c: fixed round_robin group dial method, this
- never worked well on BRI Ports (2 channels)
-
-2007-09-22 19:39 +0000 [r83558-83589] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_ael.c: This closes issue #10788 -- The exact same fixes
- are made here for the first arg in the for(arg1; arg2; arg3) {}
- statement, as were done for the 3rd arg. It can now be an
- assignment that will embedded in a Set() app, or a macro call, or
- an app call.
-
- * pbx/pbx_ael.c: This closes issue #10788 -- the 3rd arg in the for
- statement is now wrapped in Set() only if there's an '=' in that
- string. Otherwise, if it begins with '&', then a Macro call is
- generated; otherwise it is made into an app call. A bit more
- accomodating, keeps the new guys happy, and the guys with ael-1
- code should be happy, too
-
-2007-09-21 14:37 +0000 [r83432] Russell Bryant <russell@digium.com>
-
- * main/rtp.c, channels/misdn_config.c, main/cdr.c, main/channel.c,
- channels/chan_misdn.c, pbx/ael/ael.tab.c, main/ast_expr2f.c,
- main/file.c, include/asterisk/sched.h, channels/chan_h323.c,
- pbx/pbx_dundi.c, utils/ael_main.c, main/ast_expr2.fl,
- channels/chan_mgcp.c, main/sched.c, res/res_config_pgsql.c,
- main/dnsmgr.c, channels/chan_sip.c, pbx/ael/ael.y,
- main/db1-ast/hash/hash.c, include/asterisk/channel.h,
- channels/chan_iax2.c: gcc 4.2 has a new set of warnings dealing
- with cosnt pointers. This set of changes gets all of Asterisk
- (minus chan_alsa for now) to compile with gcc 4.2. (closes issue
- #10774, patch from qwell)
-
-2007-09-21 13:34 +0000 [r83400] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Fix video under certain circumstances. It
- would have been possible for the formats on the channel to not
- contain the video format. (closes issue #10782) Reported by:
- cwhuang
-
-2007-09-20 21:16 +0000 [r83316-83348] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c: When daemonizing, don't change working directory
- to "/". It makes it not be able to do a core dump when not
- running as uid=root. (closes issue #10766, xrg)
-
- * contrib/scripts/safe_asterisk: Change safe_asterisk to explicitly
- ask for /bin/bash, as it uses bashisms. (closes issue #10772,
- reported by culrich)
-
-2007-09-20 17:09 +0000 [r83246] Jason Parker <jparker@digium.com>
-
- * apps/app_disa.c: If # is pressed after dialing an extension in
- DISA, stop trying to collect more digits. (issue #10754) Reported
- by: atis Patches: app_disa.c.branch.patch uploaded by atis
- (license 242) app_disa.c.trunk.patch uploaded by atis (license
- 242)
-
-2007-09-20 16:25 +0000 [r83230-83232] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Make sure the minimum T1 timer value is
- obeyed in all cases. (closes issue #10768) Reported by: flefoll
- Patches: chan_sip.c.trunk.83071.retrans-patch uploaded by flefoll
- (license 244) chan_sip.c.br14.83070.retrans-patch uploaded by
- flefoll (license 244)
-
- * channels/chan_sip.c: Fix a minor spelling error. (closes issue
- #10769) Reported by: flefoll Patches:
- chan_sip.c.trunk.83071.inita-patch uploaded by flefoll (license
- 244) chan_sip.c.br14.83070.inita-patch uploaded by flefoll
- (license 244)
-
-2007-09-19 19:50 +0000 [r83121-83179] Russell Bryant <russell@digium.com>
-
- * apps/app_system.c: The System() and TrySystem() applications can
- take a substantial amount of time to execute while not servicing
- the channel. So, put the channel in autoservice while the command
- is being executed. (closes issue #10726, reported by mnicholson)
-
- * funcs/func_curl.c: Using curl can take a substantial amount of
- time, so the channel should be autoserviced while waiting for it
- to complete. (closes issue #10725, reported by mnicholson)
-
- * channels/chan_iax2.c: When handling a reload of chan_iax2, don't
- use an ao2_callback() to POKE all peers. Instead, use an
- iterator. By using an iterator, the peers container is not locked
- while the POKE is being done. It can cause a deadlock if the
- peers container is locked because poking a peer will try to lock
- pvt structs, while there is a lot of other code that will hold a
- pvt lock when trying to go lock the peers container. (reported to
- me directly by Loic Didelot. Thank you for the debug info!)
-
- * main/manager.c: Fix up another potential race condition. Do the
- loop decrementing use count on events with the eventq protected
- from being changed. (reported on IRC by Ivan)
-
-2007-09-19 13:47 +0000 [r83070-83074] Joshua Colp <jcolp@digium.com>
-
- * apps/app_queue.c: Protect the CDR record from modification by
- pbx_exec so that the application data contains the Queue data.
- (closes issue #10761) Reported by: snar Patches:
- app-queue-mixmonitor.patch uploaded by snar (license 245)
-
- * channels/chan_sip.c: (closes issue #10760) Reported by: dimas
- Patches: chan_sip.patch uploaded by dimas (license 88) Read in
- subscribecontext option in general to be the default.
-
-2007-09-19 09:32 +0000 [r83023-83024] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c: removed comment which violates the coding
- guidelines.
-
- * channels/misdn_config.c, channels/chan_misdn.c,
- channels/misdn/chan_misdn_config.h: added 'astdtmf' option to
- allow configuring the asterisk dtmf detector instead of the
- mISDN_dsp ones. also added the patch from irroot #10190, so that
- dtmf tones detected by the asterisk detector are passed outofband
- to asterisk, to make any use of dtmf tones at all.
-
-2007-09-19 00:19 +0000 [r82992] Russell Bryant <russell@digium.com>
-
- * apps/app_flash.c: Change the description of app_flash to note how
- it can be a useful tool instead of just saying that it is
- generally a worthless feature. (Thanks to Jim Van Meggelen for
- pointing it out and providing the proposed text)
-
-2007-09-18 23:41 +0000 [r82961] Joshua Colp <jcolp@digium.com>
-
- * apps/app_queue.c: Initialize a variable to NULL to make the world
- happy.
-
-2007-09-18 22:42 +0000 [r82929] Russell Bryant <russell@digium.com>
-
- * include/asterisk/agi.h, res/res_agi.c: Add a new patch to handle
- interrupting the fgets() call when using FastAGI. This version of
- the patch maintains the original behavior of the code when not
- using FastAGI. (closes issue #10553) Reported by: juggie Patches:
- res_agi_fgets-4.patch uploaded by juggie (license 24)
- res_agi_fgets_1.4svn.patch uploaded by juggie (license 24) Slight
- mods by me Tested by: juggie, festr
-
-2007-09-18 21:49 +0000 [r82887-82913] Doug Bailey <dbailey@digium.com>
-
- * main/manager.c: Corrected patch applied in revision r82887.
-
- * main/manager.c: Fixed a bug where http manager sessions prevented
- the eventq from being cleaned out because http manager sessions
- do not have a valid file descriptor.
-
-2007-09-18 20:56 +0000 [r82867] Russell Bryant <russell@digium.com>
-
- * main/manager.c: Fix a memory leak that can occur on systems under
- higher load. The issue is that when events are appended to the
- master event queue, they use the number of active sessions as a
- use count so it will know when all active sessions at the time
- the event happened have consumed it. However, the handling of the
- number of sessions was not properly synchronized, so the use
- count was not always correct, causing an event to disappear
- early, or get stuck in the event queue for forever. (closes issue
- #9238, reported by bweschke, patch from Ivan, modified by me)
-
-2007-09-18 20:09 +0000 [r82865] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Moving the logic for handling an empty
- membername to the create_member function so that there is a
- common place where this occurs instead of being spread out to
- several different places.
-
-2007-09-18 18:59 +0000 [r82834] Kevin P. Fleming <kpfleming@digium.com>
-
- * apps/app_queue.c: there is no need for conditional logic to
- select ->interface or ->membername, snince ->membername will
- always be populated
-
-2007-09-18 16:31 +0000 [r82802] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_dundi.c: When copying the contents from the wildcard
- peer, do a deep copy instead of shallow copy so that it doesn't
- crash when beging destroyed. (closes issue #10546, patch by me)
-
-2007-09-18 15:28 +0000 [r82751] Jason Parker <jparker@digium.com>
-
- * configs/sip.conf.sample: Correct the allowexternaldomains option
- in SIP sample config. Issue 10753
-
-2007-09-17 20:16 +0000 [r82594-82676] Russell Bryant <russell@digium.com>
-
- * apps/app_voicemail.c, main/stdtime/localtime.c: Put a memset in
- ast_localtime() instead of a couple places in app_voicemail to
- prevent the problem everywhere instead of just a couple of
- places. (related to issue #10746)
-
- * apps/app_voicemail.c: Initialize some memory to fix crashes when
- leaving voicemail. This problem was fixed by running Asterisk
- under valgrind. (closes issue #10746, reported by arcivanov,
- patched by me) *** IMPORTANT NOTE: We need to check to see if
- this same bug exists elsewhere.
-
- * res/res_features.c: Handle the case where there are multiple
- dynamic features with the same digit mapping, but won't always
- match the activated on/by access controls. In that case, the code
- needs to keep trying features for a match. (reported by Atis on
- the asterisk-dev list, patched by me)
-
-2007-09-17 16:40 +0000 [r82590-82592] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_iax2.c: revert a change that wasn't supposed to be
- committed... doh!
-
- * apps/app_queue.c, channels/chan_iax2.c: fix a couple of places
- where a logical member name (if specified) was not used, but
- instead the direct interface was listed
-
-2007-09-17 02:00 +0000 [r82514] Joshua Colp <jcolp@digium.com>
-
- * main/pbx.c: (closes issue #10734) Reported by: asgaroth Instead
- of passing a NULL pointer into snprintf pass "". It makes Solaris
- much happier.
-
-2007-09-14 21:19 +0000 [r82444] Steve Murphy <murf@digium.com>
-
- * main/cdr.c: closes issue #10668; thanks to arkadia for his patch;
- had to leave out the bit about ending the previous cdr in the
- fork; it would destroy current implementations.
-
-2007-09-14 21:17 +0000 [r82435] Russell Bryant <russell@digium.com>
-
- * configs/zapata.conf.sample: Add a note to help clarify the value
- set with the echocancel option. (inspired by Malcolm's blog post
- on blogs.digium.com about HPEC)
-
-2007-09-14 18:35 +0000 [r82396-82398] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Crap, I broke the build. Fixed.
-
- * apps/app_queue.c: Adding member name field to manager events
- where they were missing before (closes issue #10721, reported by
- snar)
-
-2007-09-14 17:48 +0000 [r82394] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c: If a channel does not have an owner, do not
- try to set a channel variable. This will end up making the
- channel variable global, which is not right. Closes issue #10720,
- patch by flefoll.
-
-2007-09-14 15:50 +0000 [r82382-82385] Russell Bryant <russell@digium.com>
-
- * build_tools/menuselect-deps.in, configure,
- include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add
- checking for libusb here, so nobody has to deal with conflicts in
- the chan_usbradio-1.4 branch every time the configure script gets
- changed
-
- * channels/chan_usbradio.c (removed), channels/xpmr (removed),
- channels/Makefile: Remove chan_usbradio from the main 1.4 branch.
- It can't live here because we have a strict policy to not include
- new features in release branches. However, I'm going to merge it
- into trunk, and I also have a special 1.4 based branch that
- includes this module. svn co
- http://svn.digium.com/svn/asterisk/team/jdixon/chan_usbradio-1.4
-
-2007-09-14 14:42 +0000 [r82376] Mark Michelson <mmichelson@digium.com>
-
- * doc/CODING-GUIDELINES: Fixing a typo in the coding guidelines
- (closes issue #10717, reported and patched by leedm777)
-
-2007-09-14 01:24 +0000 [r82368] Jim Dixon <telesistant@hotmail.com>
-
- * apps/app_rpt.c: Fixed problem with changes made to cdr
- functionality
-
-2007-09-14 00:52 +0000 [r82367] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_usbradio.c: this new driver may not live in this
- branch for long (since it is a new feature), but it definitely
- should not be built by default
-
-2007-09-14 00:34 +0000 [r82366] Jim Dixon <telesistant@hotmail.com>
-
- * apps/app_rpt.c, channels/xpmr/xpmr_coef.h (added),
- channels/chan_usbradio.c (added), channels/xpmr/xpmr.h (added),
- channels/xpmr (added), channels/xpmr/LICENSE (added),
- channels/xpmr/sinetabx.h (added), configs/usbradio.conf.sample
- (added), channels/Makefile, channels/xpmr/xpmr.c (added): Added
- channel driver for USB Radio device and support thereof.
-
-2007-09-13 23:11 +0000 [r82358] Jason Parker <jparker@digium.com>
-
- * pbx/pbx_spool.c: Fix a small typo. retrytime > waittime
-
-2007-09-13 20:16 +0000 [r82346] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Preemptively fixing a possible segfault. It is
- possible that queuename is NULL (meaning pause ALL queues), so
- use q->name instead.
-
-2007-09-13 20:11 +0000 [r82344] Jason Parker <jparker@digium.com>
-
- * cdr/cdr_csv.c: Fix a crash that could occur in cdr_csv when
- mutliple threads tried to close the same file. Do we actually
- need the locking here? What happens if you open the same file
- twice, and two threads try to write to it at the same time? Is
- fputs() going to write out the entire line at once? I suspect
- that it could be possible for the second fopen to run during the
- first fputs, so the position could be in the middle of the
- previously written line... Issue 10347, initial patch by
- explidous (but I removed all of the paranoia stuff..)
-
-2007-09-13 18:57 +0000 [r82337-82339] Russell Bryant <russell@digium.com>
-
- * main/astobj2.c: resolve a warning when not building under dev
- mode
-
- * main/astobj2.c, main/asterisk.c, include/asterisk.h: Only compile
- in tracking astobj2 statistics if dev-mode is enabled. Also, when
- dev mode is enabled, register the CLI command that can be used to
- run the astobj2 test and print out statistics.
-
-2007-09-13 18:12 +0000 [r82335] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, LICENSE: Merged revisions 82334 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r82334 | kpfleming | 2007-09-13 11:10:12 -0700 (Thu, 13 Sep 2007)
- | 2 lines clarify the OpenSSL and OpenH323 license exceptions
- ........
-
-2007-09-13 16:25 +0000 [r82326] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Added logic to handle the unlikely case that
- someone has two queues with the same name. Asterisk will log a
- warning message letting the user know that one was already
- defined with that name and is it skipping all further instances.
- This also will work for realtime queues but in order for that to
- happen, the user would have to trigger a perfectly timed reload
- as a realtime queue is being looked up, which is highly unlikely
- (but taken care of nonetheless).
-
-2007-09-13 11:47 +0000 [r82309] Philippe Sultan <philippe.sultan@gmail.com>
-
- * channels/chan_gtalk.c: Closes issue #9401, reported and patched
- by irrot, with slight modifications by me. Handle DTMF sent by
- Asterisk properly.
-
-2007-09-12 21:56 +0000 [r82296] Russell Bryant <russell@digium.com>
-
- * res/res_agi.c: Fix a check of the wrong pointer, as pointed out
- by an XXX comment left in the code. The problem was harmless,
- however.
-
-2007-09-12 21:28 +0000 [r82291] Tilghman Lesher <tlesher@digium.com>
-
- * main/stdtime/tzfile.h: Oops, wrong location for FreeBSD zone
- files
-
-2007-09-12 20:24 +0000 [r82286] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * apps/app_meetme.c: remove a race condition for the creation of
- recordthread's, and fix a small memory leak. This closes issue#
- 10636
-
-2007-09-12 20:12 +0000 [r82285] Tilghman Lesher <tlesher@digium.com>
-
- * main/stdtime/private.h, main/stdtime/tzfile.h,
- include/asterisk/localtime.h, main/stdtime/localtime.c: Working
- on issue #10531 exposed a rather nasty 64-bit issue on
- ast_mktime, so we updated the localtime.c file from source. Next
- we'll have to write ast_strptime to match.
-
-2007-09-12 15:16 +0000 [r82278-82280] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c: Clean up the output of "asterisk -h". This
- tweaks the wording and wraps lines at 80 characters. (closes
- issue #10699, seanbright)
-
- * res/res_agi.c: revert patch from issue #10553, as someone not
- using fastagi reported that this broke their system.
-
-2007-09-12 14:30 +0000 [r82274-82276] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Accidentally committed changes to
- app_voicemail which do NOT need to be in the 1.4 branch yet.
- reverting...
-
- * apps/app_voicemail.c, apps/app_queue.c: We should only initialize
- a realtime queue when it is allocated, not every time we access
- it. This prevents the members ao2_container from being
- reallocated every time the queue is accessed. I also removed a
- debug message I had accidentally left in on a previous commit.
-
-2007-09-11 22:37 +0000 [r82267] Russell Bryant <russell@digium.com>
-
- * apps/app_queue.c: Fix incorrect uses of ao2_find(). Every one of
- these calls was reading bogus memory ...
-
-2007-09-11 21:41 +0000 [r82265] Joshua Colp <jcolp@digium.com>
-
- * codecs/gsm/src/lpc.c, codecs/gsm/src/long_term.c: (closes issue
- #10679) Reported by: andrew Build under dev mode when K6OPTS is
- enabled.
-
-2007-09-11 20:49 +0000 [r82263] Russell Bryant <russell@digium.com>
-
- * apps/app_queue.c: Fix another missing unref of member objects.
- This one was pointed out by Marta. When building the outgoing
- list in try_calling(), a member reference is stored in each
- outgoing entry. However, when this list got destroyed, the
- reference was not released.
-
-2007-09-11 20:36 +0000 [r82261] Steve Murphy <murf@digium.com>
-
- * main/cdr.c: this change should fix issue # 10659 -- what I worry
- about is how many other bug reports it may generate. Hopefully,
- we can please the/a majority. Hopefully. We shall see. Calls not
- marked ANSWERED and with only one channel name will not be
- posted. This should eliminate the double CDR's.
-
-2007-09-11 16:05 +0000 [r82252] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: All instances of ao2_iterators which were just
- named 'i' have been renamed to 'mem_iter' so that when refcounted
- queues are merged into trunk, there will be little confusion
- regarding iterator names, especially when a queue and member
- iterator are used in the same function.
-
-2007-09-11 16:03 +0000 [r82250] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_dundi.c: The sample dundi.conf claims support for a
- wildcard peer entry - [*], but the code did not support it. This
- patch makes it work. (closes issue #10546, patch by dds, with
- some changes by me)
-
-2007-09-11 16:01 +0000 [r82249] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/chan_misdn.c: fixed a
- hold/retrieve issue.
-
-2007-09-11 15:26 +0000 [r82245] Russell Bryant <russell@digium.com>
-
- * res/res_agi.c: (closes issue #10553) Reported by: juggie Patches:
- res_agi_fgets-2.patch uploaded by juggie (license 24) Tested by:
- juggie When using fastagi, fgets() can return before a full line
- is read. Add explicit handling for the case where it gets
- interrupted.
-
-2007-09-11 14:56 +0000 [r82243] Joshua Colp <jcolp@digium.com>
-
- * pbx/pbx_dundi.c: (closes issue #10577) Reported by: jamesgolovich
- Patches: asterisk-dundifree.diff.txt uploaded by jamesgolovich
- (license 176) Don't leak memory when unloading DUNDi.
-
-2007-09-11 14:34 +0000 [r82198-82240] Russell Bryant <russell@digium.com>
-
- * apps/app_queue.c: Add a couple more missing unrefs of queue
- member objects
-
- * apps/app_queue.c: Add a missing unref of a queue member in an
- error handling block
-
- * apps/app_queue.c: Document why membercount can not simply be
- replaced by ao2_container_count()
-
- * main/astobj2.c: backport astobj2 race condition fix. This
- function is the exact same as trunk so it applies here as well.
-
-2007-09-10 18:02 +0000 [r82155] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_queue.c: Convert struct member to use refcounts (closes
- issue #10199)
-
-2007-09-10 15:02 +0000 [r82091] Mark Michelson <mmichelson@digium.com>
-
- * configs/misdn.conf.sample: Removing non-existent options from
- misdn configuration sample. (closes issue #10678, reported and
- patched by IgorG)
-
-2007-09-09 02:35 +0000 [r82028] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/lock.h: Fix inline compiles on really old
- compilers (who uses gcc 2.7 anymore, really?)
-
-2007-09-08 18:41 +0000 [r81952-81997] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c: Fix a small memory leak. ast_unregister_atexit()
- did not free the entry it removed.
-
- * .cleancount: (closes issue #10672) Bump the cleancount so that a
- "make clean" will be forced. This is needed because my fix in
- revision 81599 made a change to a data structure in file.h, and
- since file dependency tracking is only on with dev-mode enabled,
- file format modules that don't get rebuilt may crash, as is the
- case with this issue. This makes me wonder - how much faster does
- the code build without the file dependency tracking enabled? If
- it doesn't make much of a difference, then it may be worth just
- keeping it on all of the time, or perhaps just not in release
- tarballs, so that this type of issue is avoided.
-
-2007-09-07 19:48 +0000 [r81923] Jason Parker <jparker@digium.com>
-
- * apps/app_queue.c: Allow the MEMBERINTERFACE variable to be used
- as the mixmonitor filename. This moves the setting of the
- MEMBERINTERFACE variable to before mixmonitor. Issue 10671, patch
- by sim.
-
-2007-09-07 15:25 +0000 [r81886] Mark Michelson <mmichelson@digium.com>
-
- * configs/queues.conf.sample: Moving the explanation for joinempty
- to a more appropriate place
-
-2007-09-06 22:28 +0000 [r81832] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: (closes issue #9724, closes issue #10374)
- Reported by: kenw Patches: 9724.txt uploaded by russell (license
- 2) Tested by: kenw, russell Resolve a deadlock that occurs when
- doing a SIP transfer to parking. I come across this type of
- deadlock fairly often it seems. It is very important to mind the
- boundary between the channel driver and the core in respect to
- the channel lock and the channel-pvt lock. Channel drivers lock
- to lock the pvt and then the channel once it calls into the core,
- while the core will do it in the opposite order. The way this is
- avoided is by having channel drivers either release their pvt
- lock while calling into the core, or such as in this case,
- unlocking the pvt just long enough to acquire the channel lock.
-
-2007-09-06 22:05 +0000 [r81778-81826] Jason Parker <jparker@digium.com>
-
- * Makefile: We added COPTS for ASTCFLAGS additions, but not LDOPTS
- for ASTLDFLAGS. This adds LDOPTS
-
- * include/asterisk/astobj2.h: This should fix a build issue that
- people building against uClibc were seeing with the addition of
- astobj2
-
-2007-09-06 19:40 +0000 [r81776] Joshua Colp <jcolp@digium.com>
-
- * apps/app_meetme.c: (closes issue #10122) Reported by:
- stevefeinstein Patches: meetme-unmute-manager.diff uploaded by
- qwell (license 4) Tested by: stevefeinstein After looking over
- the code I agree with Qwell. Setting the file descriptor to
- conference each time just causes a fight back and forth.
-
-2007-09-06 16:56 +0000 [r81743] Philippe Sultan <philippe.sultan@gmail.com>
-
- * include/asterisk/jabber.h, channels/chan_gtalk.c: Various string
- length fixes. Removed an unused variable in aji_client structure
- (context)
-
-2007-09-06 16:25 +0000 [r81682-81713] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Fixes an issue where valid DTMF had to be
- pressed twice to exit a queue if a member's phone was ringing.
- (closes issue #10655, reported by strider2k, patched by me)
-
- * res/res_features.c: Fixes a memory leak (closes issue #10658,
- reported and patched by Ivan)
-
-2007-09-06 14:20 +0000 [r81650] Philippe Sultan <philippe.sultan@gmail.com>
-
- * res/res_jabber.c: According to both RFC 3920 - section 9.1.2 -
- and Google's XMPP server complaint, if set, the 'from' attribute
- must be set to the user's full JID.
-
-2007-09-05 20:53 +0000 [r81599] Russell Bryant <russell@digium.com>
-
- * include/asterisk/file.h, main/say.c, res/res_features.c,
- main/file.c, include/asterisk/channel.h: Fix an issue that can
- occur when you do an attended transfer to parking. If you
- complete the transfer before the announcement of the parking spot
- finishes, then the channel being parked will hear the remainder
- of the announcement. These changes make it so that will not
- happen anymore. Basically, res_features sets a flag on the
- channel is playing the announcement to so that the file streaming
- core knows that it needs to watch out for a channel masquerade,
- and if it occurs, to abort the announcement. (closes BE-182)
-
-2007-09-05 17:18 +0000 [r81569] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/lock.h: Solaris x86 compatibility fix
-
-2007-09-05 15:19 +0000 [r81525] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Fixing the build...
-
-2007-09-05 15:14 +0000 [r81523] Jason Parker <jparker@digium.com>
-
- * channels/chan_phone.c: Do not try to unregister a NULL channel
- tech. Also changed load_module function to use defines rather
- than numbers for return values. Issue 10651, patch by
- rbraun_proformatique, with additions by me.
-
-2007-09-05 15:03 +0000 [r81520] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Reverting behavior of QUEUE_MEMBER_COUNT to
- only count members who are logged in and available. (related to
- issue #10652, reported by wuwu)
-
-2007-09-05 13:11 +0000 [r81492] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: (closes issue #10650) Reported by: tacvbo Only
- print out that the spy was removed while holding the spy lock.
-
-2007-09-04 20:54 +0000 [r81453-81455] Jason Parker <jparker@digium.com>
-
- * apps/app_followme.c: Rather than attempt to play a file, we can
- just check whether it exists. Issue 10634, patch by me, testing
- by pabelanger, sanity checked by bweschke
-
- * configs/followme.conf.sample: Change default followme config file
- to point to the correct files. Issue 10644, patch by pabelanger
-
-2007-09-04 18:37 +0000 [r81448] Russell Bryant <russell@digium.com>
-
- * main/astobj2.c, include/asterisk/astobj2.h, channels/chan_iax2.c:
- Remove the typedefs on ao2_container and ao2_iterator. This is
- simply because we don't typedef objects anywhere else in
- Asterisk, so we might as well make this follow the same
- convention.
-
-2007-09-04 16:40 +0000 [r81442] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_sip.c: there is no point in sending 401
- Unauthorized to a UAS that sent us a properly-formatted
- Authentication header with the expected username and nonce but an
- incorrect response (which indicates the shared secret does not
- match)... instead, let's send 403 Forbidden so that the UAS
- doesn't retry with the same authentication credentials repeatedly
-
-2007-09-04 14:23 +0000 [r81435-81439] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: (closes issue #10632) Reported by:
- jamesgolovich Patches: asterisk-iaxfirmwareleak.diff.txt uploaded
- by jamesgolovich (license 176) Fix memory leak when unloading
- chan_iax2. The firmware files were not being freed.
-
- * main/channel.c: (closes issue #10476) Reported by: mdu113 Only
- look for the end of a digit when waiting for a digit. This in
- turn disables emulation in the core.
-
- * main/dns.c: (closes issue #10610) Reported by: john Patches:
- dns.c.patch uploaded by john (license 218) Tested by: mvanbaak
- Don't return a match if no SRV record actually exists.
-
-2007-09-03 18:57 +0000 [r81433] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Remove a couple of calls to
- ast_string_field_free_pools() on peers in error handling blocks
- in the code for building peers. The peer object destructor does
- this and doing it twice will cause a crash. (closes issue #10625,
- reported by and patched by pnlarsson)
-
-2007-09-01 15:57 +0000 [r81426-81428] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Changed a comment to be more accurate. (really
- this is just a test to make sure I can commit properly from home)
-
- * main/astobj2.c, include/asterisk/astobj2.h: Making match_by_addr
- into ao2_match_by_addr and making it available everywhere since
- it could be a handy callback to have
-
-2007-08-31 21:27 +0000 [r81418] Russell Bryant <russell@digium.com>
-
- * include/asterisk/astobj2.h: Remove references to a debugging
- parameter that does not exist
-
-2007-08-31 19:48 +0000 [r81416] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Fixed broken behavior of a reload on realtime
- queues. Prior to this patch, if a reload was issued and a
- realtime queue had callers waiting in it, then the queue would be
- removed from the queue list, but it would not actually be freed
- (in fact, a debug message warning about a memory leak would come
- up). With this patch, reloads do not touch realtime queues at
- all.
-
-2007-08-31 19:16 +0000 [r81415] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_logic.c: The IF() function was not allowing true
- values that had embedded colons (closes issue #10613)
-
-2007-08-31 18:44 +0000 [r81412] Jason Parker <jparker@digium.com>
-
- * apps/app_dial.c: Re-order dial options to be in line with the
- existing alpha order. Issue 10621, initial patch by junky
-
-2007-08-31 17:38 +0000 [r81410] Philippe Sultan <philippe.sultan@gmail.com>
-
- * channels/chan_gtalk.c: Make the 'gtalk show channels' CLI command
- available. Closes issue 10548, reported by keepitcool.
-
-2007-08-31 15:53 +0000 [r81406] Joshua Colp <jcolp@digium.com>
-
- * res/res_speech.c: Make it the engine's responsible to check for
- the presence of results.
-
-2007-08-31 15:51 +0000 [r81405] Kevin P. Fleming <kpfleming@digium.com>
-
- * codecs/codec_zap.c: add missing "transcoder show" (and deprecated
- "show transcoder") CLI commands that were in 1.2 but never added
- to 1.4
-
-2007-08-31 14:38 +0000 [r81401-81403] Joshua Colp <jcolp@digium.com>
-
- * res/res_features.c: (closes issue #10618) Reported by: dimas
- Don't pass through the stopped sounds frame.... just drop it.
-
- * res/res_features.c: (closes issue #10009) Reported by: dimas
- Don't output a bridge failed warning message if it failed because
- one of the channels was part of the masquerade process. That is
- perfectly normal.
-
-2007-08-30 22:05 +0000 [r81397] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Removing an extraneous (and possibly
- misleading) log message. Firstly, if the announce file isn't
- found, the streaming functions will report it. Secondly, not all
- non-zero returns from play_file mean that the announce file
- wasn't found. Positive return values simply mean that a digit was
- pressed (most likely to skip through the announcement). (closes
- issue #10612, reported and patched by dimas)
-
-2007-08-30 21:23 +0000 [r81395] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: (closes issue #10514) Reported by: casper
- Patches: chan_sip.c.80129.diff uploaded by casper (license 55)
- Remove needless check for AUTH_UNKNOWN_DOMAIN. It was impossible
- for it to ever be that value.
-
-2007-08-30 21:11 +0000 [r81392] Steve Murphy <murf@digium.com>
-
- * main/cdr.c: via issue 10599, where 'CDR already initialized'
- messages are being generated. Since all channels will have an
- init'd CDR attached at creation time, this message is now
- particularly useless. Removed.
-
-2007-08-30 15:38 +0000 [r81383] Russell Bryant <russell@digium.com>
-
- * channels/h323/ast_h323.cxx: Add missing checks for the PTRACING
- define. (closes issue #10559, paravoid)
-
-2007-08-30 15:35 +0000 [r81381] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Changed some manager event messages to reflect
- whether a queue member is a realtime member or not
-
-2007-08-30 15:33 +0000 [r81379] Russell Bryant <russell@digium.com>
-
- * configs/modem.conf.sample (removed), configs/enum.conf.sample,
- configs/extensions.ael.sample: Fix a typo, update a reload
- command, and remove an unused configuration file. (closes issue
- #10606, casper)
-
-2007-08-30 14:53 +0000 [r81375] Joshua Colp <jcolp@digium.com>
-
- * main/pbx.c: (closes issue #10603) Reported by: jmls Patches:
- pbx.diff uploaded by jmls (license 141) Backport changes from
- 81372. Add REASON dialplan variable for when an originated call
- fails and the failed extension is executed.
-
-2007-08-30 14:43 +0000 [r81373] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c: Fixed some warnings.
-
-2007-08-30 14:23 +0000 [r81369] Joshua Colp <jcolp@digium.com>
-
- * res/res_features.c: (issue #10599) Reported by: dimas Handle the
- -1 control subclass during feature dialing (it indicates to stop
- sounds).
-
-2007-08-30 08:31 +0000 [r81367] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/chan_misdn.c: Fixed a severe
- issue where a misdn_read would lock the channel, but read would
- not return because it blocks. later chan_misdn would try to queue
- a frame like a AST_CONTROL_ANSWER which could result in a
- deadlock situation. misdn_read will now not block forever
- anymore, and we don't queue the ANSWER frame at all when we
- already was called with misdn_answer -> answer would be called
- twice. Also we don't explicitly send a RELEASE_COMPLETE on
- receiption of a RELEASE anymore, because mISDN does that for us,
- this resulted in a problem on some switches, which would block
- our port after some calls for a short while.
-
-2007-08-29 16:35 +0000 [r81346-81349] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: This patch, in essence, will correctly pause a
- realtime queue member and reflect those changes in the realtime
- engine. (issue #10424, reported by irroot, patch by me) This
- patch creates a new function called update_realtime_member_field,
- which is a generic function which will allow any one field of a
- realtime queue member to be updated. This patch only uses this
- function to update the paused status of a queue member, but it
- lays the foundation for persisting the state of a realtime member
- the same way that static members' state is maintained when using
- the persistentmembers setting
-
- * apps/app_queue.c: Changed some tabs to spaces
-
-2007-08-29 15:57 +0000 [r81342] Russell Bryant <russell@digium.com>
-
- * main/Makefile: If chan_h323 is not being built, don't use g++ to
- do the final link of Asterisk. (in response to a question on the
- asterisk-dev list)
-
-2007-08-29 15:52 +0000 [r81340] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: This fix creates a more accurate way of
- detecting whether realtime members were deleted. (closes issue
- 10541, reported by Alric, patched by me) The REALLY nice things
- about this patch is that queue members now have a "realtime"
- field which will be true if the member is a realtime member. This
- means we can check this value prior to certain processing if it
- should ONLY be done for realtime members.
-
-2007-08-29 14:13 +0000 [r81331] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: (closes issue #9690) Reported by: mattv Make
- rtp timeouts work even if two RTP streams are directly bridged in
- the RTP stack.
-
-2007-08-28 21:38 +0000 [r81226-81291] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Change the message about receiving a
- mini-frame before the first full voice frame to a DEBUG message.
-
- * pbx/pbx_dundi.c: revert unintentional changes in rev 81226
-
- * configs/indications.conf.sample, pbx/pbx_dundi.c: Add Russian
- tones. (closes issue #7953, hanabana)
-
-2007-08-28 14:12 +0000 [r81120-81189] Mark Michelson <mmichelson@digium.com>
-
- * contrib/scripts/vmail.cgi: Fixes a forwarding problem when using
- res_config_mysql (closes issue #10573, reported by chrisvaughan,
- patch suggested by chrisvaughan as well)
-
- * apps/app_queue.c: Resolve a potential deadlock. In this case, a
- single queue is locked, then the queue list. In changethread(),
- the queue list is locked, and then each individual queue is
- locked. Under the right circumstances, this could deadlock. As
- such, I have unlocked the individual queue before locking the
- queue list, and then locked the queue back after the queue list
- is unlocked.
-
- * channels/chan_agent.c: DTMF begin frames should be ignored so
- that when an agent acks a call with the '#' key, he doesn't cause
- a queue's announce file to be interrupted. Also went ahead and
- did the same for the '*' key and for ending a call. (closes issue
- #10528, reported by deskhack, patched by me)
-
-2007-08-27 17:27 +0000 [r81042-81074] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_dundi.c: Add a \todo to note that this module leaks most
- of the memory it allocates on unload and should be fixed (when
- I'm not in the middle of something else ...).
-
- * pbx/pbx_dundi.c: explicity define a variable as a boolean
-
- * res/res_musiconhold.c: (closes issue #10419) Reported by:
- mustardman Patches: asterisk-mohposition.diff.txt uploaded by
- jamesgolovich (license 176) This patch fixes a few problems with
- music on hold. * Fix issues with starting at the beginning of a
- file when it shouldn't. * Fix the inuse counter to be decremented
- even if the class had not been set to be deleted when not in use
- anymore * Don't arbitrarily limit the number of MOH files to 255
-
-2007-08-27 15:01 +0000 [r81012] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: (closes issue #10561) Reported by: jesselang
- Patches: chan_sip-ChannelReload-20080825.patch uploaded by
- jesselang (license 202) Remove an extra \r\n to make the
- ChannelReload event conform with every other event.
-
-2007-08-27 14:55 +0000 [r81010] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Found a case where the queue's membercount is
- off. It does not take into account dynamic members on a reload.
-
-2007-08-27 13:20 +0000 [r80974] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: (closes issue #10562) Reported by: idkpmiller Correct
- jitter value output in the CLI to be as expected.
-
-2007-08-26 18:11 +0000 [r80932] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Remove an extra signal_condition() for the
- scheduler thread. (closes issue #10564, patch from casper)
-
-2007-08-25 17:37 +0000 [r80895] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix some issues with the handling of the
- scheduler in chan_iax2. Most of the places that scheduled items
- to be executed by the scheduler thread did not signal the
- scheduler thread to wake up so that it could recalculate the time
- until the next action. These changes will make the scheduler
- thread more responsive and ensure that actions get executed as
- close to when intended as possible instead of it being possible
- for very long delays.
-
-2007-08-24 22:59 +0000 [r80878] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * apps/app_zapateller.c: An empty string is an empty callerid ...
- so zap it. This closes issue #10502, which was pointed out by
- dswartz. Thank you, and may the swartz be with you
-
-2007-08-24 21:22 +0000 [r80820-80849] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: If dnsmgr is in use, and no DNS servers are
- available when Asterisk first starts, then don't give up on
- poking peers. Allow the poke to get rescheduled so that it will
- work once the dnsmgr is able to resolve the host. (closes issue
- #10521, patch by jamesgolovich)
-
- * main/dsp.c: Improve the debouncing logic in the DTMF detector to
- fix some reliability issues. Previously, this code used a shift
- register of hits and non-hits. However, if the start of the digit
- isn't clean, it is possible for the leading edge detector to miss
- the digit. These changes replace the flawed shift register logic
- and also does the debouncing on the trailing edge as well.
- (closes issue #10535, many thanks to softins for the patch)
-
-2007-08-24 19:52 +0000 [r80818] BJ Weschke <bweschke@btwtech.com>
-
- * apps/app_queue.c: A minor correction to the available logic of
- autofill. If a queue member is paused, they're not really
- "available" so don't count them as such. Somewhat related to
- issue #10155
-
-2007-08-24 18:52 +0000 [r80789] Steve Murphy <murf@digium.com>
-
- * main/cdr.c: From a complaint by jmls, I realize that the message
- in cdr_disposition is unnecessary. To get failure disposition,
- just return -1; no use having more than one case do that.
-
-2007-08-24 15:51 +0000 [r80750] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Fix a possible crash in IMAP voicemail.
-
-2007-08-24 15:41 +0000 [r80747] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c, UPGRADE.txt: Make the deprecation warning inline with
- the code, instead of only in documentation (closes issue #10549)
-
-2007-08-24 15:28 +0000 [r80722] Russell Bryant <russell@digium.com>
-
- * utils/ael_main.c: Tweak the formatting of this MODULEINFO block.
- I think this would have caused a "*" to get in the
- menuselect-tree file.
-
-2007-08-24 14:48 +0000 [r80689-80717] Steve Murphy <murf@digium.com>
-
- * utils/ael_main.c: This change addresses JerJer's complaint that
- aelparse builds and installs even if pbx_ael is unchecked in the
- menuselect stuff.
-
- * pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/ael/ael-test/ref.ael-test6:
- backport of 80649, a fix to an unreported problem in the ael
- parser, that results in a crash on a 64bit machine
-
-2007-08-24 11:42 +0000 [r80661] Philippe Sultan <philippe.sultan@gmail.com>
-
- * channels/chan_gtalk.c: Closes issue #10509 Googletalk calls are
- answered too early, which results in CDRs wrongly stating that a
- call was ANSWERED when the calling party cancelled a call before
- before being established. We must not answer the call upon
- reception of a 'transport-accept' iq packet, but this packet
- still needs to be acknowledged, otherwise the remote peer would
- close the call (like in #8970).
-
-2007-08-23 21:34 +0000 [r80601-80617] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * channels/misdn/isdn_lib.c: make misdn/isdn_lib compile without
- warnings
-
- * channels/chan_misdn.c: make chan_misdn compile without warnings
-
-2007-08-23 20:16 +0000 [r80539-80573] Russell Bryant <russell@digium.com>
-
- * include/asterisk/features.h, res/res_features.c: When executing a
- dynamic feature, don't look it up a second time by digit pattern
- after we already looked it up by name. This causes broken
- behavior if there is more than one feature defined with the same
- digit pattern. (closes issue #10539, reported by bungalow, patch
- by me)
-
- * funcs/func_timeout.c: Revert very broken fix for issue #10540 ...
- none of these values take ms so I don't know what I was thinking
-
- * funcs/func_timeout.c: Fix func_timeout to take values in floating
- point so 1.5 actually means 1.5 seconds instead of being rounded.
- (closes issue #10540, reported by spendergrass, patch by me)
-
-2007-08-23 17:14 +0000 [r80505-80507] Jason Parker <jparker@digium.com>
-
- * /: *sigh*
-
- * /: use autotagged externals
-
-2007-08-23 17:08 +0000 [r80501] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c: report the actual channel number that was
- unregistered, instead of assuming that the interface list
- consists of channels 1 through <x> with no gaps in the sequence
-
-2007-08-23 17:02 +0000 [r80360-80499] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix some code where it was possible for a
- reference to a peer to not get released when it should. Thank you
- to Marta Carbone for pointing this out!
-
- * main/astobj2.c, include/asterisk/astobj2.h, channels/chan_iax2.c:
- This is a hack to maintain old behavior of chan_iax2. This
- ensures that if the peers and users are being stored in a linked
- list, that they go in the list in the same order that the older
- code used. This is necessary to maintain the behavior of which
- peers and users get matched when traversing the container.
-
- * res/res_agi.c: Revert res_agi fix that didn't quite work until we
- get it right ...
-
- * include/asterisk/astobj2.h: Add some more documentation on
- iterating ao2 containers. The documentation implies that is
- possible to miss an object or see an object twice while
- iterating. After looking through the code and talking with
- mmichelson, I have documented the exact conditions under which
- this can happen (which are rare and harmless in most cases).
-
- * main/astobj2.c: When converting this code to use the list macros,
- I changed it so objects are added to the head of a bucket instead
- of the tail. However, while looking over code with mmichelson, we
- noticed that the algorithm used in ao2_iterator_next requires
- that items are added to the tail. This wouldn't have caused any
- huge problem, but it wasn't correct. It meant that if an object
- was added to a container while you were iterating it, and it was
- added to the same bucket that the current element is in, then the
- new object would be returned by ao2_iterator_next, and any other
- objects in the bucket would be bypassed in the traversal.
-
- * channels/chan_sip.c: Don't crash when using realtime in chan_sip
- without an insecure setting in the database. (closes issue
- #10348, reported by link55, fixed by me)
-
- * main/astobj2.c (added), main/Makefile, include/asterisk/astobj2.h
- (added), doc/iax.txt, UPGRADE.txt, include/asterisk/strings.h,
- channels/chan_iax2.c: Merge changes from
- team/russell/iax_refcount. This set of changes fixes problems
- with the handling of iax2_user and iax2_peer objects. It was very
- possible for a thread to still hold a reference to one of these
- objects while a reload operation tries to delete them. The fix
- here is to ensure that all references to these objects are
- tracked so that they can't go away while still in use. To
- accomplish this, I used the astobj2 reference counted object
- model. This code has been in one of Luigi Rizzo's branches for a
- long time and was primarily developed by one of his students,
- Marta Carbone. I wanted to go ahead and bring this in to 1.4
- because there are other problems similar to the ones fixed by
- these changes, so we might as well go ahead and use the new
- astobj if we're going to go through all of the work necessary to
- fix the problems. As a nice side benefit of these changes, peer
- and user handling got more efficient. Using astobj2 lets us not
- hold the container lock for peers or users nearly as long while
- iterating. Also, by changing a define at the top of chan_iax2.c,
- the objects will be distributed in a hash table, drastically
- increasing lookup speed in these containers, which will have a
- very big impact on systems that have a large number of users or
- peers. The use of the hash table will be made the default in
- trunk. It is not the default in 1.4 because it changes the
- behavior slightly. Previously, since peers and users were stored
- in memory in the same order they were specified in the
- configuration file, you could influence peer and user matching
- order based on the order they are specified in the configuration.
- The hash table does not guarantee any order in the container, so
- this behavior will be going away. It just means that you have to
- be a little more careful ensuring that peers and users are
- matched explicitly and not forcing chan_iax2 to have to guess
- which user is the right one based on secret, host, and access
- list settings, instead of simply using the username. If you have
- any questions, feel free to ask on the asterisk-dev list.
-
- * res/res_agi.c: Juggie in #asterisk-dev was reporting problems
- where fgets would return without reading the whole line when
- using fastagi. When this happens, errno was set to EINTR or
- EAGAIN. This patch accounts for the possibility and lets fgets
- continue in that case.
-
-2007-08-22 18:53 +0000 [r80302-80330] Jason Parker <jparker@digium.com>
-
- * Makefile, build_tools/mkpkgconfig, build_tools/make_build_h,
- build_tools/strip_nonapi, build_tools/prep_moduledeps,
- build_tools/make_buildopts_h: Fix a few build issues in Solaris
- (and likely others). Use GREP and ID variables from autoconf.
- Reported to me in #asterisk-dev I forgot who reported this -
- sorry. :(
-
- * Makefile: Change a syntax that the GNU make in Solaris dislikes.
-
- * build_tools/make_version: Fix a bashism (we explicitly request
- /bin/sh). Remove some oddly placed quotes I found in passing.
-
-2007-08-22 16:21 +0000 [r80257] Russell Bryant <russell@digium.com>
-
- * Makefile: Honor the contents of the COPTS variable as custom
- target CFLAGS. Apparently this is what openwrt does. (reported by
- Brian Capouch on the asterisk-dev list, patch by me)
-
-2007-08-22 16:14 +0000 [r80255] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: (closes issue #10526) Reported by: sinistermidget
- Revert commit from issue #10355 and return timestamp skew to 640.
-
-2007-08-21 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.11 released.
-
-2007-08-21 18:42 +0000 [r80183] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Don't record SIP dialog history if it's not
- turned on. Also, put an upper limit on how many history entires
- will be stored for each SIP dialog. It is currently set to 50,
- but can be increased if deemed necessary. (closes issue #10421,
- closes issue #10418, patches suggested by jmoldenhauer, patches
- updated by me) (Security implications documented in AST-2007-020)
-
-2007-08-21 16:39 +0000 [r80166-80167] Steve Murphy <murf@digium.com>
-
- * include/asterisk/alaw.h, include/asterisk/ulaw.h: ugh. removing
- the diffs from ulaw.h and alaw.h for now; accidentally added them
- in 80166
-
- * main/alaw.c, include/asterisk/alaw.h, include/asterisk/ulaw.h:
- This patch solves problem 1 in 8126; it should not slow down the
- alaw codec, but should prevent signal degradation via multiple
- trips thru the codec. Fossil estimates the twice thru this codec
- will prevent fax from working. 4-6 times thru would result
- hearable, noticeable, voice degradation.
-
-2007-08-21 15:22 +0000 [r80132] Russell Bryant <russell@digium.com>
-
- * channels/chan_mgcp.c: Don't try to dereference the owner channel
- when it may not exist (issue #10507, maxper)
-
-2007-08-21 15:03 +0000 [r80130] Jason Parker <jparker@digium.com>
-
- * configs/cdr.conf.sample: (issue #10510) Reported by: casper
- Patches: cdr.conf.diff uploaded by casper (license 55) Fix a few
- errors in sample cdr config file.
-
-2007-08-20 21:57 +0000 [r80088] Russell Bryant <russell@digium.com>
-
- * apps/app_queue.c: Fix the build of app_queue
-
-2007-08-20 21:39 +0000 [r80049-80086] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: After a discussion on #asterisk-dev, it was
- decided that this should be in 1.4 as well. (issue #10424,
- reported and patched by irroot)
-
- * apps/app_queue.c: Found a pointless ternary if. member->dynamic
- was set to 1 and has no opportunity to change between then and
- this line, so "dynamic" will ALWAYS be output.
-
-2007-08-20 16:08 +0000 [r80047] Jason Parker <jparker@digium.com>
-
- * configs/extensions.conf.sample: (issue #10499) Reported by:
- casper Patches: extensions.conf.sample.diff uploaded by casper
- (license 55) Update CLI examples in extensions.conf.sample to
- reflect command changes.
-
-2007-08-20 15:34 +0000 [r80044] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Ukrainian language voicemail support.
- (closes issue #10458, reported and patched by Oleh)
-
-2007-08-20 02:42 +0000 [r79998] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_voicemail.c: Missing curly braces. Oops. (Reported by
- snuffy via IRC)
-
-2007-08-18 14:30 +0000 [r79947] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_voicemail.c: Don't allocate vmu for messagecount when we
- could just use the stack instead (closes issue #10490) Also,
- remove a useless (and leaky) SQLAllocHandle (closes issue #10480)
-
-2007-08-17 21:01 +0000 [r79912] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c: Avoid a crash in the handling of DTMF based
- Caller ID. It is valid for ast_read to return NULL in the case
- that the channel has been hung up. (crash reported by
- anonymouz666 on IRC in #asterisk-dev)
-
-2007-08-17 19:14 +0000 [r79906] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Patch allows for more seamless transition
- from file storage voicemail to ODBC storage voicemail. If a
- retrieval of a greeting from the database fails, but the file is
- found on the file system, then we go ahead an insert the greeting
- into the database. The result of this is that people who switch
- from file storage to ODBC storage do not need to rerecord their
- voicemail greetings.
-
-2007-08-17 19:12 +0000 [r79902-79904] Jason Parker <jparker@digium.com>
-
- * channels/chan_sip.c, main/utils.c, include/asterisk/strings.h:
- Don't send a semicolon over the wire in sip notify messages.
- Caused by fix for issue 9938. I basically took the code that
- existed before 9938 was fixed, and copied it into a new function
- - ast_unescape_semicolon There should be very few places this
- will be needed (pbx_config does NOT need this (see issue 9938 for
- details)) Issue 10430, patch by me, with help/ideas from murf
- (thanks murf).
-
- * channels/chan_local.c: Re-add the setting of callerid name and
- number. Issue 10485, reported by and fix explained by paradise.
-
-2007-08-17 13:37 +0000 [r79857] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Fix some crashes in chan_sip. This patch
- changes various places that add items to the scheduler to ensure
- that they don't overwrite the ID of a previously scheduled item.
- If there is one, it should be removed. (closes issue #10391,
- closes issue #10256, probably others, patch by me)
-
-2007-08-17 08:22 +0000 [r79833] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c: sometimes we don't need to signal dtmf
- tones to asterisk, we just want them to go through as inband.
- Otherwise they might be generated by the other channel partner
- and then there is a double tone.
-
-2007-08-16 22:32 +0000 [r79756-79792] Russell Bryant <russell@digium.com>
-
- * res/res_musiconhold.c: Fix a little race condition that could
- cause a crash if two channels had MOH stopped at the same time
- that were using a class that had been marked for deletion when
- its use count hits zero.
-
- * res/res_musiconhold.c: This patch fixes a bug where reloading the
- module with "module reload" did not delete classes from memory
- that were no longer in the config. This patch fixes that problem
- as well as another one. Previously, if you reloaded MOH using the
- "moh reload" CLI command, which behaved differently than "module
- reload ...", MOH had to be stopped on every channel and started
- again immediately. However, there was no way to tell what class
- was being used, so they would all fall back to the default class.
- (closes issue #10139) Reported by: blitzrage Patches:
- asterisk-10139-advanced.diff.txt uploaded by jamesgolovich
- (license 176) Tested by: jamesgolovich
-
- * channels/chan_iax2.c: Fix more deadlocks in chan_iax2 that were
- introduced by making frame handling and scheduling
- multi-threaded. Unfortunately, we have to do some expensive
- deadlock avoidance when queueing frames on to the ast_channel
- owner of the IAX2 pvt struct. This was already handled for
- regular frames, but ast_queue_hangup and ast_queue_control were
- still used directly. Making these changes introduced even more
- places where the IAX2 pvt struct can disappear in the context of
- a function holding its lock due to calling a function that has to
- unlock/lock it to avoid deadlocks. I went through and fixed all
- of these places to account for this possibility. (issue #10362,
- patch by me)
-
-2007-08-16 21:16 +0000 [r79690-79748] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_agent.c: Fixes a problem where agents would get
- stuck busy due to their wrapuptime being longer than the queue's
- wrapuptime and ringinuse=no for the queue. (closes issue #10215,
- reported by Doug, repaired by me) Special thanks to fkasumovic
- for pointing out the source of the problem and to bweschke for
- helping to come up with a solution!
-
- * apps/app_voicemail.c: base_encode is not trying to open a log
- file, so we should not call it a log file in the warning.
- (related to issue #10452, reported by bcnit)
-
-2007-08-16 09:37 +0000 [r79665] Philippe Sultan <philippe.sultan@gmail.com>
-
- * res/res_jabber.c: A fix for two critical problems detected while
- working with Daniel McKeehan in issue #10184. Upon priority
- change, the resource list is not NULL terminated when moving an
- item to the end of the list. This makes Asterisk endlessy loop
- whenever it needs to read the list. Jids with different resource
- and priority values, like in Gmail's and GoogleTalk's jabber
- clients put that problem in evidence. Upon reception of a 'from'
- attribute with an empty resource string, Asterisk crashes when
- trying to access the found->cap pointer if the resource list for
- the given buddy is not empty. This situation is perfectly valid
- and must be handled. The Gizmoproject's jabber client put that
- problem in evidence. Also added a few comments in the code as
- well as a handle for the capabilities from Gmail's jabber client,
- which are stored in a caps:c tag rather than the usual c tag.
- Closes issue #10184.
-
-2007-08-16 08:21 +0000 [r79642] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/ie.c: 0x80 + protocol is wrong for USERUSER when
- we want to send IA5 Chars.
-
-2007-08-15 14:40 +0000 [r79553] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: (closes issue #10440) Reported by: irroot (closes
- issue #10454) Reported by: flo_turc Increase maximum timestamp
- skew to 120. 20 was apparently far too low.
-
-2007-08-15 14:26 +0000 [r79527] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Fixed an error in the Russian language
- voicemail intro. (issue #10458, reported and patched by Oleh)
-
-2007-08-15 14:18 +0000 [r79523] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: (closes issue #10456) Reported by: irroot
- Patches: sip_timeout.patch uploaded by irroot (license 52) Change
- hardcoded timer value to defined value. I'm doing this in 1.4 as
- well so if it needs to be changed in the future this place would
- not have been forgotten.
-
-2007-08-14 18:49 +0000 [r79436-79470] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix another spot where an iax2_peer would
- be leaked if realtime was in use.
-
- * channels/chan_iax2.c: Fix some memory leaks throughout chan_iax2
- related to the use of realtime. I found these while working on
- iax2_peer object reference tracking.
-
-2007-08-14 15:27 +0000 [r79397] Joshua Colp <jcolp@digium.com>
-
- * res/res_features.c: (closes issue #10415) Reported by: atis
- Revert fix for #10327 as it causes more issues then it solves.
-
-2007-08-13 22:40 +0000 [r79363] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_ael.c: memset really, really needs to be used here.
-
-2007-08-13 21:57 +0000 [r79334] Joshua Colp <jcolp@digium.com>
-
- * res/res_speech.c, apps/app_speech_utils.c,
- include/asterisk/speech.h: Instead of accepting a single DTMF
- character accept a full string.
-
-2007-08-13 20:37 +0000 [r79272-79301] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Don't call find_peer in
- registry_authrequest with the pvt lock held to avoid a deadlock.
-
- * channels/chan_iax2.c: Release the pvt lock before calling
- find_peer in register_verify to avoid a deadlock. Also, remove
- some unnecessary locking in auth_fail that was only done
- recursively.
-
- * channels/chan_iax2.c: Don't call find_peer within update_registry
- with a pvt lock held. This can cause a deadlock as the code will
- eventually call find_callno.
-
- * channels/chan_iax2.c: I am fighting deadlocks in chan_iax2. I
- have tracked them down to a single core issue. You can not call
- find_callno() while holding a pvt lock as this function has to
- lock another (every) other pvt lock. Doing so can lead to a
- classic deadlock. So, I am tracking down all of the code paths
- where this can happen and fixing them. The fix I committed
- earlier today was along the same theme. This patch fixes some
- code down the path of authenticate_reply.
-
-2007-08-13 17:49 +0000 [r79255] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael-test/ref.ael-vtest21 (added),
- pbx/ael/ael-test/ref.ael-test19,
- pbx/ael/ael-test/ael-vtest21/extensions.ael (added),
- pbx/ael/ael-test/ael-vtest21 (added),
- pbx/ael/ael-test/ref.ael-vtest17,
- pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
- pbx/ael/ael-test/ref.ael-test11, pbx/pbx_ael.c,
- pbx/ael/ael-test/ref.ael-test14, utils/ael_main.c: This patch
- fixes bug 10411. I added a new regression test, some regression
- test cleanups
-
-2007-08-13 15:28 +0000 [r79214] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix a potential deadlock in socket_process.
- check_provisioning can eventually call find_callno. You can't
- hold a pvt lock while calling find_callno because it goes through
- and locks every single one looking for a match.
-
-2007-08-13 14:51 +0000 [r79174-79207] Joshua Colp <jcolp@digium.com>
-
- * res/res_speech.c, apps/app_speech_utils.c,
- include/asterisk/speech.h: Add an API call to allow the engine to
- know that DTMF was received.
-
- * channels/chan_oss.c, channels/chan_mgcp.c, channels/chan_phone.c,
- channels/chan_local.c, channels/chan_misdn.c,
- channels/chan_zap.c, channels/chan_sip.c, channels/chan_skinny.c,
- channels/chan_h323.c, channels/chan_gtalk.c,
- channels/chan_iax2.c: (closes issue #10437) Reported by: haklin
- Don't set the callerid name and number a second time on a newly
- created channel. ast_channel_alloc itself already sets it and
- setting it twice would cause a memory leak.
-
-2007-08-11 05:23 +0000 [r79142] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * res/res_odbc.c: Ensure the connection gets marked as used at
- allocation time (closes issue #10429, report and fix by
- mnicholson)
-
-2007-08-10 20:53 +0000 [r79044-79099] Steve Murphy <murf@digium.com>
-
- * main/channel.c, pbx/pbx_spool.c, include/asterisk/channel.h: From
- a user complaint on #asterisk, I have forced pbx_spool to explain
- what reason codes mean, when they are logged
-
- * main/cdr.c: Re bug behavior mentioned in #asterisk, made this
- tweak to code, to prevent hundreds of log messages from being
- generated
-
- * main/cdr.c: This will help debug; from a question asked on
- #asterisk
-
-2007-08-10 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.10.1 released.
-
-2007-08-10 15:20 +0000 [r78995] Russell Bryant <russell@digium.com>
-
- * include/asterisk/lock.h: The last set of changes that I made to
- "core show locks" made it not able to track mutexes unless they
- were declared using AST_MUTEX_DEFINE_STATIC. Locks initialized
- with ast_mutex_init() were not tracked. It should work now.
-
-2007-08-10 14:15 +0000 [r78951-78955] Joshua Colp <jcolp@digium.com>
-
- * main/file.c: Don't bother having the core pass through or emulate
- begin DTMF frames when in an ast_waitstream. It only cares about
- the end of DTMF.
-
- * configs/queues.conf.sample: (closes issue #10422) Reported by:
- bhowell Add note to sample configuration about module load order
- and how it can cause perfectly good queue members to be marked as
- invalid.
-
-2007-08-10 13:24 +0000 [r78936] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, channels/misdn/ie.c,
- channels/misdn/isdn_msg_parser.c: fixed a bug with the useruser
- information element. We send them now also in the disconnect
- message.
-
-2007-08-09 23:47 +0000 [r78907] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Improved a bit of logic regarding
- comma-separated mailboxes in has_voicemail. Also added some
- braces to some compound if statements since unbraced if
- statements scare me in general.
-
-2007-08-09 23:10 +0000 [r78891] Steve Murphy <murf@digium.com>
-
- * Makefile: This fixes bug 10416; thanks to mvanbaak for the pretty
- output
-
-2007-08-09 22:03 +0000 [r78826-78860] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Removing some extra debug code I left in my
- last commit
-
- * apps/app_voicemail.c: Quite a few changes regarding IMAP storage.
- 1. instead of using inboxcount as the core message counting
- function, we use messagecount instead. This makes it possible to
- count messages in folders besides just INBOX and Old. 2.
- inboxcount and hasvoicemail now use messagecount as their means
- of determining return values. 3. Added a copy_message function
- for IMAP storage. Unfortunately I don't have the means to test
- it, but it seems like a pretty straightforward function. 4.
- Removed a #ifndef IMAP_STORAGE and matching #endif from
- leave_voicemail for a couple of reasons. One, we want to support
- copying mail to multiple IMAP boxes, and two, IMAP was broken
- because a STORE macro had been moved into this section of code.
-
- * channels/chan_sip.c: I broke canreinvite...Now I'm fixing it. I
- put some new code in the wrong place and so I've reverted the
- canreinvite section to how it was and put my new code where it
- should be.
-
-2007-08-09 17:58 +0000 [r78717-78778] Russell Bryant <russell@digium.com>
-
- * apps/app_voicemail.c: add a comment to indicate that inboxcount
- for ODBC_STORAGE needs to be fixed to support multiple mailboxes
-
- * apps/app_voicemail.c: Fix subscriptions to multiple mailboxes for
- ODBC_STORAGE. Also, leave a comment for this to be fixed for
- IMAP_STORAGE, as well. I left IMAP alone since I know MarkM was
- working on this code right now for another reason. This is broken
- even worse in trunk, but for a different reason. The fact that
- the mailbox option supported multiple mailboxes is completely not
- obvious from the code in the channel drivers. Anyway, I will fix
- that in another commit ...
-
- * apps/app_meetme.c: Fix a problem with the combination of the 'F'
- option to pass DTMF through a conference and options that use
- DTMF to activate various features. The problem was that the BEGIN
- frame would be passed through, but the END frame would get
- intercepted to activate a feature. Then, the other conference
- members would hear DTMF for forever, which they didn't seem to
- like very much. (closes issue #10400, reported by stevefeinstein,
- fixed by me)
-
-2007-08-08 19:29 +0000 [r78646] Jason Parker <jparker@digium.com>
-
- * doc/jabber.txt: Fix mogs email address.
-
-2007-08-08 18:16 +0000 [r78575-78620] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Fixed some compiler warnings so that
- compiling with dev-mode and IMAP storage would not have any
- errors. This section of code may get changed again shortly since
- my change uncovers a rather silly bit of logic.
-
- * apps/app_queue.c: Changing a bit of logic so that someone will
- NEVER exit the queue on timeout unless they have enabled the 'n'
- option. This commit relates to issue #10320. Thanks to
- jfitzgibbon for detailing the idea behind this code change.
-
-2007-08-08 13:51 +0000 [r78569] Joshua Colp <jcolp@digium.com>
-
- * configs/sip.conf.sample: (closes issue #10335) Reported by:
- adamgundy Update sip.conf to include another scenario where
- directrtpsetup will fail.
-
-2007-08-07 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.10 released.
-
-2007-08-07 20:57 +0000 [r78488] Russell Bryant <russell@digium.com>
-
- * res/res_config_odbc.c: Fix the build of this module on 64-bit
- platforms
-
-2007-08-07 19:43 +0000 [r78450] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: The logic behind inboxcount's return value
- was reversed in has_voicemail and message_count. (closes issue
- #10401, reported by st1710, patched by me)
-
-2007-08-07 19:34 +0000 [r78437] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * res/res_odbc.c: Don't free the environment handle when the
- connection fails, because other connections might be depending
- upon it
-
-2007-08-07 19:11 +0000 [r78416] Jason Parker <jparker@digium.com>
-
- * channels/chan_sip.c: Allow chan_sip to build in devmode
-
-2007-08-07 19:09 +0000 [r78415] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_voicemail.c, res/res_config_odbc.c,
- apps/app_directory.c: Reconnection doesn't happen automatically
- when a DB goes down (fixes issue #9389)
-
-2007-08-07 18:25 +0000 [r78375] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Properly check the capabilities count to
- avoid a segfault. (ASA-2007-019)
-
-2007-08-07 17:45 +0000 [r78371] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 78370 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r78370 | russell | 2007-08-07 12:44:04 -0500 (Tue, 07 Aug 2007) |
- 4 lines Revert patch committed for issue #9660. It broke E&M
- trunks. (closes issue #10360) (closes issue #10364) ........
-
-2007-08-06 21:41 +0000 [r78275] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: Add additional DTMF log messages to help when
- debugging issues.
-
-2007-08-06 20:44 +0000 [r78184-78242] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix an issue where dynamic threads can get
- free'd, but still exist in the dynamic thread list. (closes issue
- #10392, patch from Mihai, with credit to his colleague, Pete)
-
- * include/asterisk/linkedlists.h: Fix the return value of
- AST_LIST_REMOVE(). This shouldn't be causing any problems,
- though, because the only code that uses the return value only
- checks to see if it is NULL. (closes issue #10390, pointed out by
- mihai)
-
-2007-08-06 16:32 +0000 [r78182] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: It is possible for a transfer to occur
- before the remote device has our tag in which case they send none
- in the transfer. In this case we need to not fail the transfer
- dialog lookup.
-
-2007-08-06 16:30 +0000 [r78180] Jason Parker <jparker@digium.com>
-
- * main/config.c: Fix an issue with using UpdateConfig (manager
- action) where escaped semicolons in a config would be converted
- to just semicolons (\; to ;) Issue 9938
-
-2007-08-06 15:27 +0000 [r78166-78172] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: (closes issue #10355) Reported by: wdecarne Now that
- we pass through RTP timestamp information we need to make the
- allowed timestamp skew considerably less. There are situations
- where a source may change and due to the timestamp difference the
- receiver will experience an audio gap since we did not indicate
- by setting the marker bit that the source changed.
-
- * configure, configure.ac: (closes issue #10383) Reported by: rizzo
- Include stdlib.h so NULL gets defined for gethostbyname_r checks.
-
-2007-08-06 13:33 +0000 [r78164] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: Fixed a mistake I made in realtime_peer
- which caused it to return NULL every time. Thanks to Jon Fealy
- for emailing me the correction.
-
-2007-08-05 14:18 +0000 [r78146] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * cdr/cdr_pgsql.c: Portability fix for devmode compiling (closes
- bug #10382)
-
-2007-08-05 04:15 +0000 [r78143] Russell Bryant <russell@digium.com>
-
- * include/asterisk/lock.h: Fix compilation failure when
- MALLOC_DEBUG is enabled, but DEBUG_THREADS is not
-
-2007-08-05 03:29 +0000 [r78139] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * channels/chan_sip.c: If peer is not found, the error message is
- misleading (should be peer not found, not ACL failure)
-
-2007-08-03 20:25 +0000 [r78103] Mark Michelson <mmichelson@digium.com>
-
- * main/config.c, channels/chan_sip.c, include/asterisk/config.h:
- Changed the behavior of sip's realtime_peer function to match the
- corresponding way of matching for non-realtime peers. Now matches
- are made on both the IP address and port number, or if the
- insecure setting is set to "port" then just match on the IP
- address. In order to accomplish this, I also added a new API
- call, ast_category_root, which returns the first variable of an
- ast_category struct
-
-2007-08-03 20:14 +0000 [r78028-78101] Russell Bryant <russell@digium.com>
-
- * apps/app_voicemail.c: (closes issue #10194) Reported by:
- blitzrage Patches: bug0010194 uploaded by vovochka Tested by:
- blitzrage Fix a problem when you call Voicemail() with multiple
- mailboxes specified and ODBC_STORAGE is in use. The audio part of
- the message was only given to the first mailbox specified.
-
- * main/utils.c, include/asterisk/lock.h, main/astmm.c: Add some
- improvements to lock debugging. These changes take effect with
- DEBUG_THREADS enabled and provide the following: * This will keep
- track of which locks are held by which thread as well as which
- lock a thread is waiting for in a thread-local data structure. A
- reference to this structure is available on the stack in the
- dummy_start() function, which is the common entry point for all
- threads. This information can be easily retrieved using gdb if
- you switch to the dummy_start() stack frame of any thread and
- print the contents of the lock_info variable. * All of the
- thread-local structures for keeping track of this lock
- information are also stored in a list so that the information can
- be dumped to the CLI using the "core show locks" CLI command.
- This introduces a little bit of a performance hit as it requires
- additional underlying locking operations inside of every
- lock/unlock on an ast_mutex. However, the benefits of having this
- information available at the CLI is huge, especially considering
- this is only done in DEBUG_THREADS mode. It means that in most
- cases where we debug deadlocks, we no longer have to request
- access to the machine to analyze the contents of ast_mutex_t
- structures. We can now just ask them to get the output of "core
- show locks", which gives us all of the information we needed in
- most cases. I also had to make some additional changes to astmm.c
- to make this work when both MALLOC_DEBUG and DEBUG_THREADS are
- enabled. I disabled tracking of one of the locks in astmm.c
- because it gets used inside the replacement memory allocation
- routines, and the lock tracking code allocates memory. This
- caused infinite recursion.
-
- * channels/chan_iax2.c: Only pass through HOLD and UNHOLD control
- frames when the mohinterpret option is set to "passthrough". This
- was pointed out by Kevin in the middle of a training session.
-
- * channels/chan_iax2.c: Don't reuse the timespec that was set to 0
- in the previous timedwait as it will just return immediately.
- Also, fix some logic so the thread's lock isn't unlocked twice in
- the weird case of dynamic threads getting acquired right after a
- timeout. (pointed out by SteveK)
-
-2007-08-02 21:53 +0000 [r77993-77996] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c, configs/skinny.conf.sample: Make sure we
- actually allow 6 chars to be sent. Also make note of the "A"
- option of date format. Issue 9779, modifications by DEA, wedhorn,
- and myself.
-
- * channels/chan_skinny.c: If a device disconnects, the session will
- go away. If this happens during call setup, we need to give up.
- Issue 10325.
-
-2007-08-02 19:25 +0000 [r77949] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix the case where a dynamic thread times
- out waiting for something to do during the first time it runs.
- This shouldn't ever happen, but we should account for it anyway.
- (pointed out by pete, who works with mihai)
-
-2007-08-02 18:42 +0000 [r77947] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Make sure we clear the prompt status
- message on a hangup. Also rearrange messages to better fit with
- what a wireshark trace shows it should be. Issue 10299, initial
- patch and solution by sbisker, modified by me to fit with
- wireshark trace.
-
-2007-08-02 18:21 +0000 [r77945] Steve Murphy <murf@digium.com>
-
- * main/fskmodem.c, /: Merged revisions 77942 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r77942 | murf | 2007-08-02 11:56:37 -0600 (Thu, 02 Aug 2007) | 1
- line This patch hopefully solves 10141; The user is running with
- it, and it doesn't appear to harm asterisk's operation, and may
- prevent a crash. I'll store it in 1.2, as we have shut down
- support on 1.2, but since I developed the patch before support
- finished, and it might affect 1.4 and trunk, I'm going ahead with
- it. ........
-
-2007-08-02 18:04 +0000 [r77939-77943] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix another race condition in the handling
- of dynamic threads. If the dynamic thread timed out waiting for
- something to do, but was acquired to perform an action
- immediately afterwords, then wait on the condition again to give
- the other thread a chance to finish setting up the data for what
- action this thread should perform. Otherwise, if it immediately
- continues, it will perform the wrong action. (reported on IRC by
- mihai, patch by me) (related to issue #10289)
-
- * channels/chan_iax2.c: Add another sanity check to
- vnak_retransmit(). This check ensures that frames that have
- already been marked for deletion don't get retransmitted. (closes
- issue #10361, patch from mihai)
-
-2007-08-02 15:15 +0000 [r77890-77894] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Make sure that we show the correct
- extension if dialed from a macro "From: 5555" rather than "From:
- s" Issue 10358, initial patch by DEA, reworked by me to use S_OR,
- tested by sbisker
-
- * channels/chan_skinny.c: Put in some additional debug information
- for softkey/stimulus messages. Issue 10291, patch by DEA.
-
-2007-08-01 22:16 +0000 [r77887] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix some race conditions which have been
- causing weird problems in chan_iax2. The most notable problem is
- that people have been seeing storms of VNAK frames being sent due
- to really old frames mysteriously being in the retransmission
- queue and never getting removed. It was possible that a dynamic
- thread got created, but did not acquire its lock before the
- thread that created it signals it to perform an action. When this
- happens, the thread will sleep until it hits a timeout, and then
- get destroyed. So, the action never gets performed and in some
- cases, means a frame doesn't get transmitted and never gets freed
- since the scheduler never gets a chance to reschedule
- transmission. Another less severe race condition is in the
- handling of a timeout for a dynamic thread. It was possible for
- it to be acquired to perform at action at the same time that it
- hit a timeout. When this occurs, whatever action it was acquired
- for would never get performed. (patch contributed by Mihai and
- SteveK) (closes issue #10289) (closes issue #10248) (closes issue
- #10232) (possibly related to issue #10359)
-
-2007-08-01 22:14 +0000 [r77886] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_voicemail.c: Voicemail with ODBC_STORAGE defined does
- not compile cleanly (missing def)
-
-2007-08-01 21:08 +0000 [r77883] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Fix an issue that caused one-way audio on
- some newer devices (specifically the 7921), due to sending
- packets in the wrong order during hangup. Also make sure we clear
- tones/messages on the correct line/instance. Issue 10291, patch
- by DEA, tested by sbisker and myself.
-
-2007-08-01 18:08 +0000 [r77863-77871] Joshua Colp <jcolp@digium.com>
-
- * main/cli.c: (closes issue #10351) Reported by: ftarz Some
- platforms don't like it when you pass NULL to vsnprintf so pass
- "" instead.
-
- * include/asterisk/threadstorage.h, channels/chan_mgcp.c,
- apps/app_voicemail.c, main/acl.c, utils/smsq.c,
- channels/chan_iax2.c: Add some fixes for building on Solaris.
-
- * main/utils.c: Whoops, I meant R_5 not R5.
-
- * configure, configure.ac: And for my last trick... make sure that
- if gethostbyname_r is exported by a library that it is used.
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac,
- main/utils.c: Extend autoconf logic to determine which version of
- gethostbyname_r is on the system.
-
-2007-08-01 14:08 +0000 [r77852-77854] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Fixes an issue I introduced to queues wherein a
- queue with joinempty=yes would kick people out of the queue
- because of erroneously thinking the 'n' option was in use.
- (closes issue #10320, reported by jfitzgibbon, patched by me,
- tested by blitzrage and me) Thank you blitzrage for all the
- testing you've done lately with queues! It's much appreciated!
-
- * apps/app_queue.c: If a queue uses dynamic realtime members, then
- the member list should be updated after each attempt to call the
- queue. This fixes an issue where if a caller calls into a queue
- where no one is logged in, they would wait forever even if a
- member logged in at some point. (closes issue #10346, reported by
- and tested by blitzrage, patched by me)
-
-2007-07-31 21:09 +0000 [r77845-77846] Jim Dixon <telesistant@hotmail.com>
-
- * apps/app_rpt.c: Much newer version, 0.70 with much additions
-
- * main/dsp.c, channels/chan_zap.c: Made VAST improvements in DTMF
- receiver in RADIO_RELAX mode (thanx Steve W9SH), and oversight in
- logic in TONE_VERIFY/RELAX mode in chan_zap.
-
-2007-07-31 20:59 +0000 [r77844] Steve Murphy <murf@digium.com>
-
- * /, contrib/scripts/ast_grab_core: Merged revisions 77842 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r77842 | murf | 2007-07-31 13:19:35 -0600 (Tue, 31 Jul 2007) | 1
- line This probably isn't super-general, but it's a first stab at
- using kill -11 to generate a core file instead of gcore. ........
-
-2007-07-31 16:17 +0000 [r77831] Joshua Colp <jcolp@digium.com>
-
- * res/res_speech.c, include/asterisk/speech.h: Add a flag to the
- speech API that allows an engine to set whether it received
- results or not.
-
-2007-07-31 15:53 +0000 [r77827] Kevin P. Fleming <kpfleming@digium.com>
-
- * build_tools/cflags.xml: DETECT_DEADLOCKS can't be enabled without
- DEBUG_THREADS or it does nothing
-
-2007-07-31 15:21 +0000 [r77824] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: This patch makes Asterisk send 100 Trying
- provisional responses upon receipt of re-invites. This makes it
- so that if there are two or more Asterisk servers between
- endpoints, the Asterisk servers will not keep retransmitting the
- re-invites. (closes issue #10274, reported by cstadlmann, patched
- by me with approval from file)
-
-2007-07-30 20:17 +0000 [r77795] Jason Parker <jparker@digium.com>
-
- * main/say.c: Applications like SayAlpha() should not hang up the
- channel if you request an "unknown" character such as a comma.
- Instead, skip the character and move on. Issue 10083, initial
- patch by jsmith, modified by me.
-
-2007-07-30 20:16 +0000 [r77785-77794] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix an issue that could potentially cause
- corruption of the global iax frame queue. In the network_thread()
- loop, it traverses the list using the AST_LIST_TRAVERSE_SAFE
- macro. However, to remove an element of the list within this
- loop, it used AST_LIST_REMOVE, instead of
- AST_LIST_REMOVE_CURRENT, which I believe could leave some of the
- internal variables of the SAFE macro invalid. Mihai says that he
- already made this change in his local copy and it didn't help his
- VNAK storm issues, but I still think it's wrong. :)
-
- * res/res_agi.c: (closes issue #10279) Reported by: seanbright
- Patches: res_agi.carefulwrite.1.4.07252007.patch uploaded by
- seanbright (license 71) res_agi.carefulwrite.trunk.07252007.patch
- uploaded by seanbright (license 71) Allow the "agi_network: yes"
- line to be printed out in the AGI debug output. Also, allow
- partial writes to be handled when writing out this line just like
- it is for all of the others.
-
- * main/channel.c: file and I both committed changes for issue
- #10301. Remove a duplicated assignment to restore the original
- value of the previous channel.
-
-2007-07-30 18:43 +0000 [r77783] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, res/res_agi.c: Merged revisions 77782 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r77782 | tilghman | 2007-07-30 13:40:54 -0500 (Mon, 30 Jul 2007)
- | 2 lines Revert change in revision 71656, even though it fixed a
- bug, because many people were depending upon the (broken)
- behavior. ........
-
-2007-07-30 17:29 +0000 [r77780] Russell Bryant <russell@digium.com>
-
- * main/channel.c: (closes issue #10301) Reported by: fnordian
- Patches: asterisk-1.4.9-channel.c.patch uploaded by fnordian
- (license 110) Additional changes by me Fix some problems in
- channel_find_locked() which can cause an infinite loop. The
- reference to the previous channel is set to NULL in some cases.
- These changes ensure that the reference to the previous channel
- gets restored before needing it again. I'm not convinced that the
- code that is setting it to NULL is really the right thing to do.
- However, I am making these changes to fix the obvious problem and
- just leaving an XXX comment that it needs a better explanation
- that what is there now.
-
-2007-07-30 17:11 +0000 [r77768-77778] Joshua Colp <jcolp@digium.com>
-
- * res/res_features.c: (closes issue #10327) Reported by: kkiely
- Instead of directly mucking with the extension/context/priority
- of the channel we are transferring when it has a PBX simply call
- ast_async_goto on it. This will ensure that the channel gets
- handled properly and sent to the right place.
-
- * main/channel.c: (closes issue #10301) Reported by: fnordian
- Patches: asterisk-1.4.9-channel.c.patch uploaded by fnordian
- (license 110) Restore previous behavior where if we failed to
- lock the channel we wanted we would return to exactly the same
- point as if we had just reentered the function.
-
- * /, apps/app_macro.c: Merged revisions 77767 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r77767 | file | 2007-07-30 11:50:02 -0300 (Mon, 30 Jul 2007) | 4
- lines (closes issue #10334) Reported by: ramonpeek Pass through
- the return value from macro_exec through the MacroIf application.
- ........
-
-2007-07-27 18:15 +0000 [r77571] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * res/res_odbc.c: Missing newline
-
-2007-07-27 17:04 +0000 [r77536-77540] Joshua Colp <jcolp@digium.com>
-
- * cdr/cdr_pgsql.c: (closes issue #10310) Reported by: prashant_jois
- Patches: cdr_pgsql.patch uploaded by prashant (license 114)
- Finish the Postgresql connection after the log messages are
- printed so we don't access invalid memory.
-
- * channels/chan_sip.c: (closes issue #10323) Reported by: julianjm
- Patches: chan_sip_device_state_hold_fix.v1.diff.txt uploaded by
- julianjm (license 99) Clear ONHOLD flag when decrementing the
- onHold peer count. If we did not do this the count may keep
- decreasing.
-
-2007-07-27 14:30 +0000 [r77490] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: "re-invite" was misspelled
-
-2007-07-26 23:19 +0000 [r77460] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: (closes issue #10302) Reported by: litnialex If a
- DTMF end frame comes from a channel without a begin and it is
- going to a technology that only accepts end frames (aka INFO)
- then use the minimum DTMF duration if one is not in the frame
- already.
-
-2007-07-26 22:16 +0000 [r77424-77429] Kevin P. Fleming <kpfleming@digium.com>
-
- * doc/mp3.txt: change protocol for downloads as well
-
- * doc/mp3.txt, sounds/Makefile: use new canonical name for download
- server
-
-2007-07-26 21:23 +0000 [r77410] Russell Bryant <russell@digium.com>
-
- * Makefile, build_tools/make_buildopts_h: AST_DEVMODE was defined
- in trunk, but not in 1.4. When Asterisk is compiled under dev
- mode, AST_DEVMODE will get defined in buildopts.h. Change 1.4 to
- define it in the same way that trunk does. Also, revert the
- change that added this define in the Makefile The advantage to
- doing it this way is that buildopts.h gets installed when you
- install Asterisk. Then, when building any out of tree modules, or
- building asterisk-addons, these modules know which options the
- rest of Asterisk was built with.
-
-2007-07-26 20:35 +0000 [r77380] Mark Michelson <mmichelson@digium.com>
-
- * Makefile, main/logger.c: Fixes to get ast_backtrace working
- properly. The AST_DEVMODE macro was never defined so the majority
- of ast_backtrace never attempted compilation. The makefile now
- defines AST_DEVMODE if configure was run with --enable-dev-mode.
- Also, changes were made to acccomodate 64 bit systems in
- ast_backtrace. Thanks to qwell, kpfleming, and Corydon76 for
- their roles in allowing me to get this committed
-
-2007-07-26 19:32 +0000 [r77348-77350] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * main/logger.c: Missed one
-
- * main/logger.c: Oops, that builtin define should be all-lowercase.
-
-2007-07-26 18:30 +0000 [r77318] Mark Michelson <mmichelson@digium.com>
-
- * cdr/cdr_pgsql.c: Two consecutive calls to PQfinish could occur,
- meaning free gets called on the same variable twice. This patch
- sets the connection to NULL after calls to PQfinish so that the
- problem does not occur. Also in this patch, prashant_jois
- informed me that it is safe to pass a null pointer to PQfinish,
- so I have removed the check for conn's existence from
- my_unload_module. (closes issue 10295, reported by junky, patched
- by me with input from prashant_jois)
-
-2007-07-25 22:39 +0000 [r77191] Steve Murphy <murf@digium.com>
-
- * apps/app_meetme.c: This fix solves problem with intense squelch
- noise when someone joins conf in bug 9430; We repro'd the problem
- with meetme opts of 'CciMo'; Josh Colp supplied this patch, and
- I'm applying it. It looks like playing the recorded username will
- louse up the next thing played into the channel. Josh rearranged
- the code so as to start things over before playing data directly
- into the conference.
-
-2007-07-25 22:16 +0000 [r77176] Joshua Colp <jcolp@digium.com>
-
- * apps/app_speech_utils.c: (closes issue #10303) Reported by: jtodd
- Add SPEECH_DTMF_TERMINATOR variable so the user can specify the
- digit to terminate a DTMF string with. If none is specified then
- no terminator will be used.
-
-2007-07-25 21:52 +0000 [r77154] Mark Michelson <mmichelson@digium.com>
-
- * main/channel.c: chan->emulate_dtmf_duration is an unsigned int,
- not a signed int, so use %u instead of %d in the format string
-
-2007-07-25 20:23 +0000 [r77116-77136] Jason Parker <jparker@digium.com>
-
- * /: so are my fingers...
-
- * /: autotagexternals script is still obviously misbehaving...
-
- * /: use autotagged externals
-
-2007-07-25 17:14 +0000 [r77071] Joshua Colp <jcolp@digium.com>
-
- * configure, acinclude.m4: Fix autoconf logic for finding OpenH323
- when it is not in the first place searched (/usr/share/openh323).
-
-2007-07-25 09:34 +0000 [r77022] Luigi Rizzo <rizzo@icir.org>
-
- * main/rtp.c: set the sequence number in a frame for all frame
- types
-
-2007-07-25 00:18 +0000 [r76983] Steve Murphy <murf@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 76978 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r76978 | murf | 2007-07-24 18:07:24 -0600 (Tue, 24 Jul 2007) | 1
- line this fixes bug 10293, where the error message because
- defaultzone or loadzone was not defined was confusing ........
-
-2007-07-24 22:12 +0000 [r76891-76937] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, include/asterisk/lock.h: Merged revisions 76934 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r76934 | tilghman | 2007-07-24 17:11:33 -0500 (Tue, 24
- Jul 2007) | 2 lines Oops, res contains the error code, not errno.
- I was wondering why a mutex was reporting "No such file or
- directory"... ........
-
- * main/app.c: Found another place where we should be using the
- umask (thanks jcmoore)
-
-2007-07-24 Jason Parker <jparker@digium.com>
-
- * Asterisk 1.4.9 released.
-
-2007-07-24 16:42 +0000 [r76803-76805] Jason Parker <jparker@digium.com>
-
- * channels/chan_iax2.c: Don't create the Asterisk channel until we
- are starting the PBX on it. (ASA-2007-018)
-
-2007-07-24 16:26 +0000 [r76801] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Added a membercount variable to call_queue
- struct which keeps track of the number of logged in members in a
- particular queue. This makes it so that the 'n' option for
- Queue() can act properly depending on which strategy is used. If
- the strategy is roundrobin, rrmemory, or ringall, we want to ring
- each phone once before moving on in the dialplan. However, if any
- other strategy is used, we will only ring one phone since it
- cannot be guaranteed that a different phone will ring on
- subsequent attempts to ring a phone. As a side effect of this,
- the QUEUE_MEMBER_COUNT dialplan function now just reads the
- membercount variable instead of traversing through the member
- list to figure out how many members there are. Special thanks to
- blitzrage for helping to test this out. (closes issue #10127,
- reported by bcnit, patched by me, tested by blitzrage)
-
-2007-07-23 22:38 +0000 [r76708] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_voicemail.c: It was our stated intention for 1.4 that
- files created in app_voicemail should depend upon the umask.
- Unfortunately, mkstemp() creates files with mode 0600, regardless
- of the umask. This corrects that deficiency.
-
-2007-07-23 18:59 +0000 [r76656] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Fix some incorrect softkey labels in
- messages. Don't try to play dialtone in some unimplemented
- features.
-
-2007-07-23 18:29 +0000 [r76654] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 76653 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r76653 | file | 2007-07-23 15:28:13 -0300 (Mon, 23 Jul
- 2007) | 4 lines (closes issue #5866) Reported by: tyler Do not
- force channel format changes when a generator is present. The
- generator may have changed the formats itself and changing them
- back would cause issues. ........
-
-2007-07-23 17:57 +0000 [r76620] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Don't try to queue up hold/unhold frames
- on a non-existent channel. Issue 10276.
-
-2007-07-23 17:48 +0000 [r76519-76618] Joshua Colp <jcolp@digium.com>
-
- * apps/app_morsecode.c: Allow app_morsecode to build on PPC Linux
- by putting the value of the digit char in an int.
-
- * /, channels/chan_sip.c: Merged revisions 76560 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r76560 | file | 2007-07-23 11:32:07 -0300 (Mon, 23 Jul 2007) | 6
- lines (closes issue #10236) Reported by: homesick Patches:
- rpid_1.4_75840.patch uploaded by homesick (license 91) Accept
- Remote Party ID on guest calls. ........
-
- * channels/chan_skinny.c: (closes issue #10268) Reported by:
- mvanbaak Patches: chan_skinny_openbsd.diff uploaded by mvanbaak
- (license 7) Add another OS that has to use the Macros for byte
- ordering.
-
-2007-07-23 12:25 +0000 [r76485] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Use a signed integer for storing the number
- of bytes in the packet read from the network. Using an unsigned
- value here made it impossible to handle an error returned from
- recvfrom(). Furthermore, in the case that recvfrom() did return
- an error, this would cause a crash due to a heap overflow.
- (closes issue #10265, reported by and fix suggested by
- timrobbins)
-
-2007-07-21 02:02 +0000 [r76227] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 76226 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r76226 | russell | 2007-07-20 21:01:46 -0500 (Fri, 20 Jul 2007) |
- 4 lines Backport a fix for a memory leak that was fixed in trunk
- in reivision 76221 by rizzo. The memory used for the localaddr
- list was not freed during a configuration reload. ........
-
-2007-07-20 21:36 +0000 [r76211] Steve Murphy <murf@digium.com>
-
- * sounds/Makefile: This patch from 10249 is worth applying! It
- prevents downloading sound files if they are already downloaded.
- Darn Practical, if you ask me
-
-2007-07-20 21:03 +0000 [r76174-76178] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Allow getting a call from an existing
- "sub" channel. Cancel ringing if endpoint hangs up before
- answering. Fixes were backported from trunk (there was apparently
- a bit of confusion during merge of a previous patch). (closes
- issue #10241)
-
- * main/manager.c: Eliminate a compiler warning with gcc 4.2 by
- constifying a char *
-
- * channels/chan_skinny.c: It's possible for sub->owner to be NULL
- here if you cancel the call immediately after/during sending a
- digit.
-
-2007-07-20 18:42 +0000 [r76139] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_directory.c: When using users.conf for the entries in
- the directory, if multiple users had the same last name, only the
- first user listed would be available in the directory. (closes
- issue #10200, reported by mrskippy, patched by me)
-
-2007-07-20 18:22 +0000 [r76132] Russell Bryant <russell@digium.com>
-
- * main/channel.c: Use the define that specifies the default length
- of an artificially created DTMF digit in the ast_senddigit()
- function. The define is set to 100ms by default, which is the
- same thing that this function was using. But, using the define
- lets changes take effect in this case, as well as the others
- where it was already used.
-
-2007-07-20 17:20 +0000 [r76054-76087] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 76080 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r76080 | file | 2007-07-20 14:16:48 -0300 (Fri, 20 Jul 2007) | 6
- lines (closes issue #10247) Reported by: fkasumovic Patches:
- chan_sip.patch uploaded by fkasumovic (license #101) Drop any
- peer realm authentication entries when reloading so multiple
- entries do not get added to the peer. ........
-
- * res/res_convert.c: (closes issue #10246) Reported by: fkasumovic
- Patches: res_conver.patch uploaded by fkasumovic (license #101)
- Use the last occurance of . to find the extension, not the first
- occurance.
-
- * apps/app_queue.c: Move makeannouncement variable declaration to
- proper place.
-
-2007-07-19 20:36 +0000 [r75980] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Remove some duplicate code.
-
-2007-07-19 18:59 +0000 [r75969-75978] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: The diff on this looks pretty big but all I did
- was remove a pointless if statement (always evaluates true).
-
- * apps/app_queue.c: Changes in handling return values of several
- functions in app_queue. This all started as a fix for issue
- #10008 but now includes all of the following changes: 1.
- Simplifying the code to handle positive return values from ast
- API calls. 2. Removing the background_file function. 3. The fix
- for issue #10008 (closes issue #10008, reported and patched by
- dimas)
-
-2007-07-19 15:53 +0000 [r75928] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 75927 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r75927 | russell | 2007-07-19 10:49:42 -0500 (Thu, 19 Jul 2007) |
- 6 lines When processing full frames, take sequence number
- wraparound into account when deciding whether or not we need to
- request retransmissions by sending a VNAK. This code could cause
- VNAKs to be sent erroneously in some cases, and to not be sent in
- other cases when it should have been. (closes issue #10237,
- reported and patched by mihai) ........
-
-2007-07-18 22:59 +0000 [r75807] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Need to make sure we set milliseconds and
- timestamp - pointed out by the recent ast_ time stuff from
- Tilghman
-
-2007-07-18 21:09 +0000 [r75759] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 75757 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r75757 | russell | 2007-07-18 16:09:13 -0500 (Wed, 18 Jul 2007) |
- 5 lines When traversing the queue of frames for possible
- retransmission after receiving a VNAK, handle sequence number
- wraparound so that all frames that should be retransmitted
- actually do get retransmitted. (issue #10227, reported and
- patched by mihai) ........
-
-2007-07-18 20:40 +0000 [r75749] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_voicemail.c, /: Merged revisions 75748 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r75748 | tilghman | 2007-07-18 15:31:36 -0500 (Wed, 18 Jul 2007)
- | 2 lines Store prior to copy (closes issue #10193) ........
-
-2007-07-18 20:17 +0000 [r75732] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Umm, why are we transmitting dialtone on
- cfwdall?
-
-2007-07-18 20:00 +0000 [r75712] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c, channels/chan_sip.c, channels/chan_agent.c,
- pbx/pbx_realtime.c: Backport GCC 4.2 fixes. Without these
- Asterisk won't build under devmode using GCC 4.2.
-
-2007-07-18 19:54 +0000 [r75707-75711] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Fixes for 7935/7936 conference phones.
- Issue 9245, patch by slimey.
-
- * channels/chan_skinny.c: Fix issues with new 79x1 phones. Issue
- 9887, patches by DEA
-
-2007-07-18 17:56 +0000 [r75658] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 75657 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r75657 | dhubbard | 2007-07-18 12:48:33 -0500 (Wed, 18 Jul 2007)
- | 1 line removed the word 'pissed' from ast_log(...) function
- call for BE-90 ........
-
-2007-07-18 15:44 +0000 [r75583-75623] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Few more places that needs to check for
- onhold state.
-
- * channels/chan_sip.c: (closes issue #10165) Reported by: elandivar
- It is possible for hold status to exist without call limits set,
- so we need to ensure update_call_counter is executed regardless.
-
- * channels/chan_h323.c: Don't bother reloading chan_h323 if it did
- not load successfully in the first place. This would otherwise
- cause a crash.
-
- * pbx/pbx_dundi.c: (closes issue #10224) Reported by: irroot Record
- the threadid of each running thread before shutting them down as
- the thread themselves may change the value.
-
-2007-07-18 12:29 +0000 [r75529] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_meetme.c: Using a freed frame causes crashes (closes
- issue #9317)
-
-2007-07-17 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.8 released.
-
-2007-07-17 20:57 +0000 [r75441-75450] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_skinny.c: Merged revisions 75449 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r75449 | russell | 2007-07-17 15:57:09 -0500 (Tue, 17
- Jul 2007) | 3 lines Properly check for the length in the skinny
- packet to prevent an invalid memcpy. (ASA-2007-016) ........
-
- * main/rtp.c: cast arguments to ast_log so that it builds without
- warnings for me
-
- * channels/iax2-parser.c, channels/iax2-parser.h, /,
- channels/chan_iax2.c: Merged revisions 75444 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r75444 | russell | 2007-07-17 15:45:27 -0500 (Tue, 17 Jul 2007) |
- 5 lines Ensure that when encoding the contents of an ast_frame
- into an iax_frame, that the size of the destination buffer is
- known in the iax_frame so that code won't write past the end of
- the allocated buffer when sending outgoing frames. (ASA-2007-014)
- ........
-
- * /, channels/chan_iax2.c: Merged revisions 75440 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r75440 | russell | 2007-07-17 15:41:41 -0500 (Tue, 17 Jul 2007) |
- 4 lines After parsing information elements in IAX frames, set the
- data length to zero, so that code later on does not think it has
- data to copy. (ASA-2007-015) ........
-
-2007-07-17 20:40 +0000 [r75439] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Ensure that the pointer to STUN data does not go to
- unaccessible memory. (ASA-2007-017)
-
-2007-07-17 20:33 +0000 [r75437] Russell Bryant <russell@digium.com>
-
- * res/res_agi.c: (issue #10210) Reported by: juggie Patches:
- 10210-1.4-grr.patch uploaded by juggie (license #24) Tested by:
- juggie, blitzrage Log a warning if someone uses DeadAGI on a live
- channel.
-
-2007-07-17 20:03 +0000 [r75405] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_dial.c: Fixing an error I made earlier. ast_fileexists
- can return -1 on failure, so I need to be sure that we only enter
- the if statement if it is successful. Related to my fix to issue
- #10186
-
-2007-07-17 20:01 +0000 [r75401-75403] Russell Bryant <russell@digium.com>
-
- * main/pbx.c: (closes issue #10209) Reported by: juggie Patches:
- 10209-trunk-2.patch uploaded by juggie Tested by: juggie,
- blitzrage In ast_pbx_run(), mark a channel as hung up after an
- application returned -1, or when it runs out of extensions to
- execute. This is so that code can detect that this channel has
- been hung up for things like making sure DeadAGI is used on
- actual dead channels, and is beneficial for other things, like
- making sure someone doesn't try to start spying on a channel that
- is about to go away.
-
- * res/res_agi.c: Remove a duplicated newline character in AGI debug
- output. (closes issue #10207, patch by seanbright)
-
-2007-07-16 20:53 +0000 [r75258-75306] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/dns.c, /: Merged revisions 75304 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r75304 | kpfleming | 2007-07-16 15:46:58 -0500 (Mon, 16 Jul 2007)
- | 3 lines provide proper copyright/license attribution for this
- structure that was copied from a BSD-licensed header file long,
- long ago... ........
-
- * /: another fix that is not needed here (finishing up 75251)
-
-2007-07-16 18:16 +0000 [r75253] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_dial.c: Restoring functionality from 1.2 wherein
- Retrydial will not exit if there is no announce file specified.
- This change makes it so that if there is no announce file
- specified, the application will continue until finished (or
- caller hangs up). If a bogus announce file is specified, then a
- warning message will be printed saying that the file could not be
- found, but execution will still continue. (closes issue #10186,
- reported by jon, patched by me)
-
-2007-07-16 18:12 +0000 [r75252] Kevin P. Fleming <kpfleming@digium.com>
-
- * /: block change that is not relevant here
-
-2007-07-13 20:36 +0000 [r75108] Russell Bryant <russell@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 75107 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r75107 | russell | 2007-07-13 15:35:22 -0500 (Fri, 13
- Jul 2007) | 3 lines Fix a couple potential minor memory leaks.
- load_moh_classes() could return without destroying the loaded
- configuration. ........
-
-2007-07-13 20:15 +0000 [r75078] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_chanspy.c, /: Merged revisions 75066 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r75066 | mmichelson | 2007-07-13 15:10:39 -0500 (Fri, 13 Jul
- 2007) | 5 lines Fixed an issue where chanspy flags were
- uninitialized if no options were passed. What triggered this
- investigation was an IRC chat where some people's quiet flags
- were set while others' weren't even though none of them had
- specified the q option. ........
-
-2007-07-13 20:10 +0000 [r75053-75067] Russell Bryant <russell@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 75059 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r75059 | russell | 2007-07-13 15:07:21 -0500 (Fri, 13
- Jul 2007) | 6 lines Ensure that adding a user to the list of
- users of a specific music on hold class is not done at the same
- time as any of the other operations on this list to prevent list
- corruption. Using the global moh_data lock for this is not ideal,
- but it is what is used to protect these lists everywhere else in
- the module, and I am only changing what is necessary to fix the
- bug. ........
-
- * channels/chan_zap.c, /: Merged revisions 75052 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r75052 | russell | 2007-07-13 14:10:00 -0500 (Fri, 13 Jul 2007) |
- 12 lines (closes issue #9660) Reported by: mmacvicar Patches
- submitted by: bbryant, russell Tested by: mmacvicar, marco,
- arcivanov, jmhunter, explidous When using a TDM400P (and probably
- other analog cards) there was a chance that you could hang up and
- pick the phone back up where it has been long enough to be not
- considered a flash hook, but too soon such that the device
- reports that it is busy and the person on the phone will only
- hear silence. This patch makes chan_zap more tolerant of this and
- gives the device a couple of seconds to succeed so the person on
- the phone happily gets their dialtone. ........
-
-2007-07-12 23:00 +0000 [r74998] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_agent.c: Change to my previous fix regarding agent
- logoff soft. Now uses deferlogoff instead of loginstart since
- loginstart is used after logoff. Thanks to makoto for pointing
- this out and suggesting the fix. (closes issue #10178, reported
- and patched by makoto, with modification by me)
-
-2007-07-12 20:42 +0000 [r74955] Steve Murphy <murf@digium.com>
-
- * channels/chan_sip.c: This patch resolves 10143; thanks to irroot
- for the patch; looked acceptable. Let the community decide if it
- messes things up
-
-2007-07-12 19:17 +0000 [r74888-74922] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: Whoops... didn't want this to be returned to 0
- each iteration.
-
- * main/channel.c: When waiting for a digit ensure that a begin
- frame was received with it, not just an end frame. (issue #10084
- reported by rushowr)
-
-2007-07-12 16:53 +0000 [r74839-74866] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: It helps if I actually add this stuff for
- the 7921 too - otherwise it won't actually do much of anything.
-
- * channels/chan_skinny.c: Add device ID for 7921 wireless skinny
- phone
-
- * channels/chan_skinny.c: Fix dialing in skinny that was broken in
- some cases. Issue 10136, fix provided by DEA.
-
-2007-07-12 15:53 +0000 [r74815] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 74814 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r74814 | file | 2007-07-12 12:51:24 -0300 (Thu, 12 Jul
- 2007) | 2 lines Only print out a warning for situations where it
- is actually helpful. (issue #10187 reported by denke) ........
-
-2007-07-11 22:57 +0000 [r74767] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 74766 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r74766 | russell | 2007-07-11 17:53:26 -0500 (Wed, 11 Jul 2007) |
- 5 lines The function make_trunk() can fail and return -1 instead
- of a valid new call number. Fix the uses of this function to
- handle this instead of treating it as the new call number. This
- would cause a deadlock and memory corruption. (possible cause of
- issue #9614 and others, patch by me) ........
-
-2007-07-11 21:14 +0000 [r74722] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 74719 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r74719 | mmichelson | 2007-07-11 16:12:30 -0500 (Wed, 11
- Jul 2007) | 5 lines The cli command "agent logoff Agent/x soft"
- did not work...at all. Now it does. (closes issue #10178,
- reported and patched by makoto, with slight modification for 1.4
- and trunk by me) ........
-
-2007-07-11 18:34 +0000 [r74657] Russell Bryant <russell@digium.com>
-
- * res/res_config_odbc.c, /: Merged revisions 74656 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r74656 | russell | 2007-07-11 13:33:23 -0500 (Wed, 11
- Jul 2007) | 4 lines Make sure that the ESCAPE immediately follows
- the condition that uses LIKE. This fixes realtime extensions with
- ODBC. (closes issue #10175, reported by stuarth, patch by me)
- ........
-
-2007-07-11 18:18 +0000 [r74628-74642] Steve Murphy <murf@digium.com>
-
- * Makefile: This fixes 10172, where the entire man8 dir gets
- removed during an uninstall of asterisk
-
- * utils/expr2.testinput, doc/channelvariables.txt, UPGRADE.txt:
- further reversion of previously applied floating point stuff for
- expr2
-
-2007-07-11 17:16 +0000 [r74515-74590] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_phone.c, configure,
- include/asterisk/autoconfig.h.in, configure.ac: Instead of
- figuring out kernel versions that have compiler.h and not...
- let's just use autoconf to check for it's presence. (issue #10174
- reported by francesco_r)
-
- * channels/chan_phone.c: Only check if we need to do a SIGMA based
- tone generation if we have a card. (issue #10179 reported by
- mikowhy)
-
-2007-07-10 23:32 +0000 [r74476] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Forwarding a message with IMAP storage was
- storing the message in the sender's box instead of the forwarded
- mailbox. (closes issue #10138, reported and patched by jaroth)
-
-2007-07-10 19:58 +0000 [r74374-74428] Jason Parker <jparker@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 74427 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r74427 | qwell | 2007-07-10 14:57:20 -0500 (Tue, 10 Jul 2007) | 6
- lines Fix an issue where it was possible to have a service level
- of over 100% Between the time recalc_holdtime and update_queue
- was called, it was possible that the call could have been hungup.
- Move both additions to the same place, so this won't happen.
- Issue 10158, initial patch by makoto, modified by me. ........
-
- * main/dns.c: Don't use #if to check if something is defined - use
- #ifdef instead. Pointed out by kpfleming
-
- * /, channels/chan_agent.c: Merged revisions 74376 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r74376 | qwell | 2007-07-10 14:03:45 -0500 (Tue, 10 Jul
- 2007) | 4 lines Fix an issue with wrapuptime not working when
- using AgentLogin. Issue 10169, patch by makoto, with a minor mod
- by me to not re-break issue 9618 ........
-
- * main/dns.c, /, configure, include/asterisk/autoconfig.h.in,
- configure.ac: Merged revisions 74373 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r74373 | qwell | 2007-07-10 13:37:23 -0500 (Tue, 10 Jul 2007) | 5
- lines Use res_ndestroy on systems that have it. Otherwise, use
- res_nclose. This prevents a memleak on NetBSD - and possibly
- others. Issue 10133, patch by me, reported and tested by scw
- ........
-
-2007-07-10 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.7.1 released.
-
-2007-07-10 16:00 +0000 [r74323] Russell Bryant <russell@digium.com>
-
- * res/res_musiconhold.c: fix an uninitialized variable
-
-2007-07-10 15:38 +0000 [r74317] Jason Parker <jparker@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 74316 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r74316 | qwell | 2007-07-10 10:37:54 -0500 (Tue, 10 Jul 2007) | 4
- lines Fix a small typo in description in of Voicemail()
- application. Issue 10170, patch by casper. ........
-
-2007-07-10 15:31 +0000 [r74314] Russell Bryant <russell@digium.com>
-
- * res/res_config_odbc.c, /: Merged revisions 74313 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r74313 | russell | 2007-07-10 10:30:20 -0500 (Tue, 10
- Jul 2007) | 3 lines Only use ESCAPE when LIKE is used. (issue
- #10075, this part reported by jmls on IRC, patch by me) ........
-
-2007-07-10 14:50 +0000 [r74262-74265] Joshua Colp <jcolp@digium.com>
-
- * /, main/app.c: Merged revisions 74264 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r74264 | file | 2007-07-10 11:48:00 -0300 (Tue, 10 Jul 2007) | 2
- lines Ensure the group information category exists before trying
- to do a string comparison with it. (issue #10171 reported by
- mlegas) ........
-
- * channels/chan_sip.c: Only spit out an inringing warning message
- when it is applicable. Since call limits are already toast in
- realtime let's not scare the user if they are using it. (issue
- #10166 reported by bcnit)
-
-2007-07-09 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.7 released.
-
-2007-07-09 21:31 +0000 [r74162-74211] Russell Bryant <russell@digium.com>
-
- * configure, configure.ac: Update the configure script to check for
- a required function that is not present in the 1.2 version of
- libpri. This will prevent the configure script from thinking that
- it has compatible libpri support for Asterisk 1.4, when it
- actually does not because the installed version is from 1.2.
-
- * res/res_musiconhold.c: (closes issue #10123) Reported by:
- blitzrage Patches submitted by: juggie, qwell, me Tested by:
- blitzrage When trying to find a music on hold class to use, try
- all of the options, instead of only the first one that is set.
- Also, change the MusicOnHold applications to not hang up on the
- channel when a class can not be found.
-
-2007-07-09 20:19 +0000 [r74159] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 74158 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r74158 | qwell | 2007-07-09 15:18:15 -0500 (Mon, 09 Jul 2007) | 8
- lines Several chan_zap options were not working on reload because
- they were arbitrarily disallowed when reloading some/most PRI
- options (such as signalling) was disallowed. Options such as
- polarityonanswerdelay and answeronpolarityswitch can safely be
- changed on a reload. This corrects that behavior. Issue 9186,
- patch by tzafrir. ........
-
-2007-07-09 18:38 +0000 [r74120-74122] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Forgot to get rid of an extraneous debug
- message.
-
- * apps/app_queue.c: The n option for Queue should make the queue
- exit immediately after failure to reach any members and should
- not be dependent on the timeout value passed to Queue (closes
- issue #10127, reported by bcnit, repaired by me)
-
-2007-07-09 15:32 +0000 [r74082] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_skinny.c: Only destroy the scheduler context if it
- was allocated. (issue #10124 reported by gzero)
-
-2007-07-09 14:57 +0000 [r74047] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Fixed a logic error in leave_voicemail.
- Pass the mailbox instead of the context to inbox_count when the
- context is "default." (closes issue #10135, reported by yannj,
- repaired by me)
-
-2007-07-09 14:49 +0000 [r74043-74045] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_skinny.c, pbx/pbx_dundi.c: Few minor thread
- synchronization tweaks. (issue #10124 reported by gzero)
-
- * configure, acinclude.m4: Use AC_CHECK_HEADER to check for
- ptlib/openh323 to allow for cross compiling. (issue #9675
- reported by zandbelt)
-
-2007-07-09 04:03 +0000 [r73985] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * main/ast_expr2f.c: Doxygen formatting fixes; fixes errors while
- 'make progdocs'. (Closes issue #10104)
-
-2007-07-09 03:13 +0000 [r73930-73980] Joshua Colp <jcolp@digium.com>
-
- * main/cdr.c: Give Agent channel names priority when doing CDR
- merging. (issue #10011 reported by krtorio)
-
- * pbx/pbx_config.c: Add a few sanity checks when writing out the
- dialplan. (issue #10157 reported by dome)
-
-2007-07-08 09:47 +0000 [r73849] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: While tracking down a bug, I need some more
- history. Dumphistory is very useful, indeed.
-
-2007-07-06 23:02 +0000 [r73769] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 73768 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r73768 | russell | 2007-07-06 18:01:22 -0500 (Fri, 06 Jul 2007) |
- 4 lines If a sip_pvt struct has already registered an extension
- state callback, remove the old one before adding a new one. If
- this isn't done, Asterisk will crash. (issue #10120) ........
-
-2007-07-06 16:36 +0000 [r73727] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Fixing a rare case which causes voicemail
- to crash when compiled with IMAP storage. inboxcount has the
- possibility of finding an "interactive" vm_state when no
- persistent "non-interactive" vm_state exists for that mailbox. If
- this should happen when someone attempts to leave a message, it
- results in a crash. This patch, along with my commit in revision
- 72670 fix issue 10053, reported by jaroth. closes issue #10053
-
-2007-07-06 16:12 +0000 [r73679-73696] Russell Bryant <russell@digium.com>
-
- * res/res_config_odbc.c, /: Merged revisions 73684 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r73684 | russell | 2007-07-06 11:06:27 -0500 (Fri, 06
- Jul 2007) | 8 lines (closes issue #10075) Reported by: apsaras
- Patches submitted by: Corydon76 Tested by: apsaras Fix a problem
- with MSSQL 2005 by explicitly stating that '\' is being used as
- an escape character. ........
-
- * /, channels/chan_sip.c: Merged revisions 73678 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r73678 | russell | 2007-07-06 10:55:41 -0500 (Fri, 06 Jul 2007) |
- 7 lines (closes issue #10125) Reported by: makoto Patches
- submitted by: makoto This fixes a crash in chan_sip that happens
- when the bindaddr setting is not valid on Asterisk startup, gets
- fixed, and then a reload gets issued. ........
-
-2007-07-06 15:27 +0000 [r73675] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 73674 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r73674 | mmichelson | 2007-07-06 10:26:40 -0500 (Fri, 06
- Jul 2007) | 5 lines Fixed a bug wherein agents get stuck busy.
- (issue 9618, reported by jiddings, patched by moi) closes issue
- #9618 ........
-
-2007-07-06 03:34 +0000 [r73551-73629] Russell Bryant <russell@digium.com>
-
- * BUGS: fix a little spelling error
-
- * channels/chan_sip.c: Fix a crash in chan_sip. Don't try to stop
- the monitor thread if it was never started. (closes issue #10124,
- reported by gzero, fixed by me)
-
- * channels/chan_iax2.c: copy from the correct buffer when deferring
- a full frame (related to issue #9937)
-
- * channels/chan_iax2.c: * Store the call number that a thread is
- processing without the full frame bit set to ease debugging *
- When deferring a full frame for processing, stick it into the
- queue for the thread that is processing frames for that call, not
- the one that read the current frame and is about to go back into
- the idle list (related to issue #9937)
-
-2007-07-05 22:20 +0000 [r73548] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 73547 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r73547 | kpfleming | 2007-07-05 17:11:51 -0500 (Thu, 05 Jul 2007)
- | 2 lines we shouldn't allow G.723.1 endpoints to use VAD, just
- like we don't support it for G.729 ........
-
-2007-07-05 20:50 +0000 [r73512] Russell Bryant <russell@digium.com>
-
- * res/res_features.c: Pass HOLD and UNHOLD frames to the other
- channel when they are returned from a native bridge function.
- This fixes a problem where when two zap channels are natively
- bridged and one does a flash hook, the other channel did not
- receive music on hold. (Reported to me directly by Doug Bailey at
- Digium)
-
-2007-07-05 19:18 +0000 [r73467] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 73466 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r73466 | file | 2007-07-05 16:15:18 -0300 (Thu, 05 Jul 2007) | 2
- lines Copy language information to the dialog structure when
- calling a peer for situations where a PBX may be started on the
- dialed channel. (issue #10121 reported by clegall_proformatique)
- ........
-
-2007-07-05 15:59 +0000 [r73400] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Correcting a minor CLI bug I found. When
- issuing the queue show command, if you type queue show and then
- press tab, you can continue pressing tab and it will keep
- auto-completing queue names even though only 1 queue can be used
- as an argument.
-
-2007-07-05 15:28 +0000 [r73398] Russell Bryant <russell@digium.com>
-
- * channels/chan_vpb.cc, channels/Makefile: Make this module build
- for me in dev-mode
-
-2007-07-05 14:21 +0000 [r73316-73355] Joshua Colp <jcolp@digium.com>
-
- * apps/app_chanspy.c, main/channel.c, /: Merged revisions 73349 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r73349 | file | 2007-07-05 11:19:14 -0300 (Thu, 05 Jul 2007) | 2
- lines Tweak spy locking. (issue #9951 reported by welles)
- ........
-
- * channels/chan_local.c, /: Merged revisions 73318 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r73318 | file | 2007-07-05 10:26:02 -0300 (Thu, 05 Jul
- 2007) | 2 lines Actually check to make sure a PBX was started on
- one of the Local channels instead of blindly assuming it was.
- (issue #10112 reported by makoto) ........
-
- * /, apps/app_queue.c: Merged revisions 73315 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r73315 | file | 2007-07-05 10:19:17 -0300 (Thu, 05 Jul 2007) | 2
- lines Reset ServicelevelPerf variable back to 0 if we are unable
- to calculate it each time... otherwise we will get previous
- values. (issue #10117 reported by noriyuki) ........
-
-2007-07-04 14:53 +0000 [r73208-73253] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, /: Merged revisions 73252 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r73252 | crichter | 2007-07-04 16:50:58 +0200 (Mi, 04
- Jul 2007) | 1 line bchannel configurations like echocancel and
- volume control, need to be setuped on inbound calls too. ........
-
- * channels/chan_misdn.c, /: Merged revisions 73207 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r73207 | crichter | 2007-07-04 10:20:54 +0200 (Mi, 04
- Jul 2007) | 1 line bad bug in overlapdial case, we called
- start_pbx multiple times, because the state wasn't changed..
- ........
-
-2007-07-03 20:17 +0000 [r73143] Steve Murphy <murf@digium.com>
-
- * main/ast_expr2.fl, main/ast_expr2.c, main/Makefile,
- main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2f.c: Removing
- expr floating patch from 1.4; too much of a behavior change. If
- you want this fix, try trunk instead. bug 9508.
-
-2007-07-03 15:42 +0000 [r73104-73106] Jason Parker <jparker@digium.com>
-
- * /: What the heck. This should not have happened.
-
- * /: use autotagged externals
-
-2007-07-03 12:38 +0000 [r73053] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_dial.c, /: Merged revisions 73052 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r73052 | tilghman | 2007-07-03 07:34:14 -0500 (Tue, 03 Jul 2007)
- | 2 lines RetryDial should accept a 0 argument, but it does not,
- because atoi does not distinguish between 0 and error (closes
- issue #10106) ........
-
-2007-07-03 08:17 +0000 [r73005] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /: Merged revisions 73004 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r73004 | crichter | 2007-07-03 10:04:35 +0200 (Di, 03
- Jul 2007) | 1 line fixed issue, that misdn_l2l1_check could only
- be called from mISDN Source channels.. #9449 ........
-
-2007-07-02 20:16 +0000 [r72933] Steve Murphy <murf@digium.com>
-
- * main/ast_expr2.fl, main/ast_expr2.c, utils/expr2.testinput,
- main/Makefile, main/ast_expr2.h, main/ast_expr2.y,
- main/ast_expr2f.c, doc/channelvariables.txt, UPGRADE.txt: support
- for floating point numbers added to ast_expr2 $\[...\] exprs.
- Fixes bug 9508, where the expr code fails with fp numbers. The
- MATH function returns fp numbers by default, so this fix is
- considered necessary.
-
-2007-07-02 18:18 +0000 [r72926] Russell Bryant <russell@digium.com>
-
- * main/manager.c: Remove a bogus comment and add proper locking to
- the handler function for the CLI command to show information on
- manager actions.
-
-2007-07-02 14:32 +0000 [r72888] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: Added additional DTMF debug messages for when
- emulation occurs.
-
-2007-07-02 08:41 +0000 [r72850-72852] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
- revisions 72585 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72585 | crichter | 2007-06-29 15:08:26 +0200 (Fr, 29 Jun 2007) |
- 1 line check if the bchannel stack id is already used, if so
- don't use it a second time. Also added a release_chan lock, so
- that the same chan_list object cannot be freed twice. chan_misdn
- does not crash anymore on heavy load with these changes. ........
-
- * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
- channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c:
- Merged revisions 72099 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72099 | crichter | 2007-06-27 15:22:37 +0200 (Mi, 27 Jun 2007) |
- 1 line simplified generation for dummy bchannels, also we mark
- them as dummies, so they are not used later as real-bchannels,
- optimized the RESTART mechanisms, we block a channel now on
- cause:44, and send out a RESTART automatically, then on reception
- of RESTART_ACKNOWLEDGE we unblock the channel again. ........
-
- * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, /: Merged
- revisions 72087 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72087 | crichter | 2007-06-27 11:26:53 +0200 (Mi, 27 Jun 2007) |
- 1 line simplified channel finding and locking a lot. removed
- unnecessary #ifdefed areas. ........
-
-2007-07-01 23:52 +0000 [r72806] Russell Bryant <russell@digium.com>
-
- * pbx/pbx_spool.c, /: Merged revisions 72805 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72805 | russell | 2007-07-01 18:51:34 -0500 (Sun, 01 Jul 2007) |
- 5 lines When appending lines to call files to keep track of
- retries, write a leading newline just in case the original call
- file did not have a newline at the end. This fix is in response
- to a problem I saw reported on the asterisk-users mailing list.
- ........
-
-2007-06-30 16:50 +0000 [r72705-72766] Russell Bryant <russell@digium.com>
-
- * configure, configure.ac: Tweak the configure script so that error
- output isn't spewed to the console when searching for GTK2 libs,
- and they aren't found.
-
- * formats/format_pcm.c: give format_pcm a more concise destription
-
-2007-06-29 19:07 +0000 [r72665] Luigi Rizzo <rizzo@icir.org>
-
- * main/utils.c: Use !defined(HAVE_GETHOSTBYNAME_R) to check for
- absence of the function. This was already done in trunk.
-
-2007-06-29 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.6 released.
-
-2007-06-29 14:26 +0000 [r72597-72599] Joshua Colp <jcolp@digium.com>
-
- * main/cdr.c: Minor change for older GCC versions.
-
- * Makefile, configure, configure.ac, makeopts.in: Backport fix for
- GCC versions without support for declaration-after-statement.
-
-2007-06-29 04:47 +0000 [r72554-72556] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * main/manager.c: Issue 10055 - Change memory allocation to use the
- heap for a command, since the output has the potential to
- overflow the stack (as it did here)
-
- * res/res_jabber.c: Fix 1.4 breakage
-
-2007-06-28 19:44 +0000 [r72493] Russell Bryant <russell@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in: regenerate the
- configure script for rizzo
-
-2007-06-28 19:29 +0000 [r72453-72489] Luigi Rizzo <rizzo@icir.org>
-
- * configure.ac: add a check for gethostbyname_r so we can simplify
- the handling e.g. in utils.c Also add comments on a couple of
- features which are not working on FreeBSD. All the above has been
- already done in trunk so the merge must be blocked. Can someone
- please regenerate ./configure ?
-
- * Makefile, channels/chan_zap.c, main/say.c: Add
- -Wdeclaration-after-statement to AST_DEVMODE flags to catch
- variable declarations in the middle of a block. Fix the few
- instances of the above spotted out by the compiler. All of this
- has been already done or is not applicable in trunk, so the merge
- of this change will be blocked.
-
- * apps/app_meetme.c: cast a time_t so that it does not conflict
- with the print format. This change was already done on trunk so
- this change needs to be blocked from merging.
-
-2007-06-27 23:29 +0000 [r72383] Brett Bryant <bbryant@digium.com>
-
- * main/asterisk.c, /: Merged revisions 72373 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72373 | bbryant | 2007-06-27 18:22:13 -0500 (Wed, 27 Jun 2007) |
- 3 lines Reinstating patch. This actually fixes the problem,
- however I was running a development branch without it and
- mistakenly thought it wasn't fixed. Fixes issue #10010, and
- #9654: 100% CPU usage caused by an asterisk console losing it's
- controlling terminal. ........
-
-2007-06-27 23:25 +0000 [r72381] Joshua Colp <jcolp@digium.com>
-
- * apps/app_mixmonitor.c, /: Merged revisions 72378 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r72378 | file | 2007-06-27 19:24:01 -0400 (Wed, 27 Jun
- 2007) | 2 lines Update documentation to clarify variable usage
- with MixMonitor. (issue #9494 reported by netoguy) ........
-
-2007-06-27 23:03 +0000 [r72335] Brett Bryant <bbryant@digium.com>
-
- * main/asterisk.c, /: Merged revisions 72333 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72333 | bbryant | 2007-06-27 17:58:53 -0500 (Wed, 27 Jun 2007) |
- 2 lines Reverted changes for earlier revisions 72259 to 72261.
- Issue #9654, #10010 ........
-
-2007-06-27 22:58 +0000 [r72328-72331] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_gtalk.c: Make payload IDs for iLBC/Speex match to
- our list. Since these are dynamic payloads the other side
- shouldn't care. (issue #9426 reported by irroot)
-
- * /, apps/app_queue.c: Merged revisions 72327 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72327 | file | 2007-06-27 18:43:11 -0400 (Wed, 27 Jun 2007) | 2
- lines Fix issue where queue log events might be missing. (issue
- #7765 reported by mtryfoss) ........
-
-2007-06-27 21:08 +0000 [r72272] Russell Bryant <russell@digium.com>
-
- * /, pbx/pbx_config.c: Merged revisions 72267 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72267 | russell | 2007-06-27 16:06:45 -0500 (Wed, 27 Jun 2007) |
- 5 lines Fix a minor issue with parsing the priority number. You
- could have as much whitespace as you want around a numeric
- priority, but you couldn't have any whitespace around a special
- priority like "n" or "hint". (issue #10039, reported by mitheloc,
- fixed by me) ........
-
-2007-06-27 20:46 +0000 [r72260] Brett Bryant <bbryant@digium.com>
-
- * main/asterisk.c, /: Merged revisions 72259 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72259 | bbryant | 2007-06-27 15:43:53 -0500 (Wed, 27 Jun 2007) |
- 4 lines Fixes 100% load when controlling terminal disappears.
- Issue #9654, #10010 ........
-
-2007-06-27 20:25 +0000 [r72257] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /: Merged revisions 72256 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72256 | file | 2007-06-27 16:23:24 -0400 (Wed, 27 Jun 2007) | 2
- lines I may possibly get shot for doing this... but... defer CDR
- processing until after the channel has been dealt with. This
- should eliminate all of the issues with channels going funky
- (SIP/PRI) when you are posting CDRs to a database that is either
- slow or unavailable and do not want to enable batching. ........
-
-2007-06-27 19:13 +0000 [r72205] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_zap.c: use the proper type for storing group number
- bits so that if someone specifies 'group=42' it will actually
- work instead of being silently ignored
-
-2007-06-27 18:40 +0000 [r72182-72185] Jason Parker <jparker@digium.com>
-
- * apps/app_voicemail.c: Fix another problem in voicemail with
- missing symbols. Issue 10074, patch by kryptolus, extended to
- include #if 0'd blocks (just in case)
-
-2007-06-27 17:31 +0000 [r72148] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: Make the ast_read_noaudio API call behave better
- under circumstances where DTMF emulation was happening and a
- generator was setup. (issue #10065 reported by stevefeinstein)
-
-2007-06-27 17:10 +0000 [r72125] Jason Parker <jparker@digium.com>
-
- * channels/chan_gtalk.c: Don't modify a variable that we don't want
- modified. Make a copy of it instead. Issue 10029, patch by
- phsultan with slight modifications by me (to remove needless
- casts).
-
-2007-06-27 16:34 +0000 [r72112] Russell Bryant <russell@digium.com>
-
- * main/rtp.c: Only output debug information related to RTCP
- timestamps when RTCP debug is turned on (issue #10066, patch by
- me)
-
-2007-06-27 07:58 +0000 [r72042] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, /: Merged revisions 72040-72041 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r72040 | crichter | 2007-06-27 09:49:27 +0200 (Mi, 27 Jun 2007) |
- 1 line for inbound TE calls, we setup the bchannel when we get
- the CONNECT_ACKNOWLEDGE, to make sure mISDN has everything ready.
- removed some #if 0 areas which weren't used anymore. ........
- r72041 | crichter | 2007-06-27 09:54:30 +0200 (Mi, 27 Jun 2007) |
- 1 line isdn_lib.c didn't compile ........
-
-2007-06-27 00:58 +0000 [r72006] Joshua Colp <jcolp@digium.com>
-
- * pbx/pbx_dundi.c: Make unloading of pbx_dundi actually work.
-
-2007-06-26 23:02 +0000 [r71953] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Removing a pointless line. This variable
- was already set earlier and between then and this line, there is
- no way that the values on the right side of the assignment could
- have changed.
-
-2007-06-26 20:36 +0000 [r71915] Jason Parker <jparker@digium.com>
-
- * main/rtp.c: Don't dereference a pointer that may be NULL here.
- Issue 10017.
-
-2007-06-26 19:00 +0000 [r71877] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: A few changes, the ultimate goal of which
- is to keep better track of the number of messages that a mailbox
- currently has. A description of the changes: 1. Changed the
- "updated" field of the vm_state struct to act more as a binary
- semaphore than a counting semaphore, since its current
- implementation made the inboxcount function not work properly.
- This change falls in line with a change made by UPenn with their
- IMAP setup and helps to sync our changes with theirs. 2.
- Eliminated some redundant calls to get_vm_state_by_mailbox inside
- leave_voicemail 3. Use the play_folder variable to keep track of
- the number of old and new messages in a mailbox as the messages
- are deleted 4. Added an increment to the number of new messages
- that was not there previously in the leave_voicemail function
-
-2007-06-26 15:47 +0000 [r71796] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Fixing bug where the authuser was
- mistakenly pulled from the mailbox string instead of the IMAP
- user. (closes issue 10054, reported and patched by jaroth)
-
-2007-06-26 12:27 +0000 [r71657-71751] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_voicemail.c, /: Merged revisions 71750 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r71750 | tilghman | 2007-06-26 07:25:58 -0500 (Tue, 26 Jun 2007)
- | 2 lines Issue 10062 - Trying to move a message without
- selecting one first results in memory corruption ........
-
- * /, res/res_agi.c: Merged revisions 71656 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r71656 | tilghman | 2007-06-25 13:12:37 -0500 (Mon, 25 Jun 2007)
- | 2 lines Issue 10035 - handle_exec returns a result inconsistent
- with all of the other AGI commands ........
-
-2007-06-25 14:13 +0000 [r71522-71576] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_h323.c: Build a peer as well when hash323 is
- enabled in users.conf (issue #9599 reported by asagage)
-
- * channels/chan_agent.c: Minor tweak for queueing up the unhold
- frame... this will teach me to do bugs while half asleep. (issue
- #10046 reported by dimas)
-
-2007-06-25 12:40 +0000 [r71519] Russell Bryant <russell@digium.com>
-
- * doc/asterisk-mib.txt: Fix a typo in the Asterisk mib. (issue
- #10048, Matti)
-
-2007-06-25 01:10 +0000 [r71412-71430] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 71414 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r71414 | file | 2007-06-24 21:02:49 -0400 (Sun, 24 Jun 2007) | 2
- lines Ignore other URIs after the first in a 300 Multiple Choice
- response. (issue #10041 reported by homesick) ........
-
- * main/cdr.c: Fix it so 1.4 actually compiles on my box.
-
- * channels/chan_agent.c: Check to make sure the channel pointer is
- present before queueing up an unhold frame on it. (issue #10046
- reported by dimas)
-
-2007-06-24 20:16 +0000 [r71362-71371] Russell Bryant <russell@digium.com>
-
- * build_tools/prep_tarball: Include the menuselect-tree file in
- tarballs to make builds from tarballs a little bit faster
-
- * main/asterisk.c, /: Merged revisions 71358 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r71358 | russell | 2007-06-24 15:04:21 -0500 (Sun, 24 Jun 2007) |
- 2 lines Revert the patch from issue 9654 due to an unexpected
- side effect ........
-
-2007-06-24 17:50 +0000 [r71289-71291] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * res/res_features.c: Issue 10044 - chan->cdr is NULL here, so
- peer->cdr is what we really wanted to use
-
- * main/db.c, main/manager.c, /: Merged revisions 71288 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r71288 | tilghman | 2007-06-24 12:32:21 -0500 (Sun, 24
- Jun 2007) | 2 lines Issue 10043 - There is a legitimate need to
- be able to set variables to the empty string. ........
-
-2007-06-23 03:29 +0000 [r71230] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, res/res_features.c: This patch is meant to fix 8433;
- where clid and src are lost via bridging.
-
-2007-06-22 22:44 +0000 [r71214] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
- channels/chan_misdn.c, /: Merged revisions 70341 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r70341 | crichter | 2007-06-20 17:29:09 +0200 (Mi, 20
- Jun 2007) | 1 line fixed a bug that was introduced by copy and
- paste in the last commit ..bchannels weren't cleaned properly.
- ........
-
-2007-06-22 15:38 +0000 [r71096-71123] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
- revisions 70672 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r70672 | crichter | 2007-06-21 15:11:29 +0200 (Do, 21 Jun 2007) |
- 1 line we activate the bchannels in TE mode on incoming calls
- only when we want to connect the call. ........
-
- * channels/misdn/isdn_lib.c, /: Merged revisions 70342 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r70342 | crichter | 2007-06-20 17:42:39 +0200 (Mi, 20
- Jun 2007) | 1 line forgot one place .. ........
-
- * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
- channels/chan_misdn.c, /: Merged revisions 70311 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r70311 | crichter | 2007-06-20 16:47:59 +0200 (Mi, 20
- Jun 2007) | 1 line on receiption of cause:44 we mark the channel
- as in use and inform the user about the situation, we need to
- test the RESTART stuff then. Also shuffled the
- empty_chan_in_stack function after the bchannel cleaning
- functions, to avoid race conditions. ........
-
- * channels/chan_misdn.c, /: Merged revisions 69887 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r69887 | crichter | 2007-06-19 15:23:04 +0200 (Di, 19
- Jun 2007) | 1 line when we send out a SETUP, but get no response,
- we should cleanup everything after reception of a hangup.
- ........
-
- * /, channels/misdn/isdn_msg_parser.c: Merged revisions 69053 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r69053 | crichter | 2007-06-13 11:55:54 +0200 (Mi, 13 Jun 2007) |
- 1 line restart indicator 0x80 is correct, at least that's what
- libpri does. ........
-
- * channels/chan_misdn.c, /: Merged revisions 68887 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r68887 | crichter | 2007-06-12 10:35:22 +0200 (Di, 12
- Jun 2007) | 1 line if the bridged partner is mISDN too we should
- not send dtmf tones, they are transmitted inband always ........
-
- * channels/chan_misdn.c, /: Merged revisions 68874 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r68874 | crichter | 2007-06-12 09:48:52 +0200 (Di, 12
- Jun 2007) | 1 line if we have already some digits, we just stop
- the tones. ........
-
-2007-06-22 15:00 +0000 [r71068] Jason Parker <jparker@digium.com>
-
- * apps/app_speech_utils.c, /, res/res_agi.c, main/file.c: Merged
- revisions 71065 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r71065 | qwell | 2007-06-22 09:52:18 -0500 (Fri, 22 Jun 2007) | 4
- lines Fix a few silly usages of ast_playstream() - it only ever
- returns 0... Issue 10035 ........
-
-2007-06-22 14:53 +0000 [r71066] Brett Bryant <bbryant@digium.com>
-
- * main/asterisk.c, /: Merged revisions 71064 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r71064 | bbryant | 2007-06-22 09:39:34 -0500 (Fri, 22 Jun 2007) |
- 10 lines Fixed infinite loop when controlling terminal was lost
- and return value of input function wasn't checked for errors.
- This would cause 100% cpu to be taken up. (closes issue #9654,
- issue #10010) Reported by: mnicholson, and eserra Idea for the
- patch from mnicholson, patched by me ........
-
-2007-06-22 14:10 +0000 [r71063] Steve Murphy <murf@digium.com>
-
- * main/cdr.c: My conditions for merging amaflags info was naive;
- DOCUMENTATION is the default, although null is possible; theft of
- user-settable fields is not good. Just copy them, leave them
- alone.
-
-2007-06-22 03:14 +0000 [r71003] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix a small typo which ... well ...
- completely broke chan_iax2. oops! (issue #9937, patch by me)
-
-2007-06-21 22:34 +0000 [r70949] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, /: Merged revisions 70948 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r70948 | murf | 2007-06-21 16:29:50 -0600 (Thu, 21 Jun 2007) | 1
- line This little fix is in response to bug 10016, but may not
- cure it. The code is wrong, clearly. In a situation where you set
- the CDR's amaflags, and then ForkCDR, and then set the new CDR's
- amaflags to some other value, you will see that all CDRs have had
- their amaflags changed. This is not good. So I fixed it. ........
-
-2007-06-21 21:40 +0000 [r70899] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 70898 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r70898 | file | 2007-06-21 17:37:55 -0400 (Thu, 21 Jun 2007) | 2
- lines Don't explode if the gain option is specified without a
- value. (issue #9274 reported by mfarver) ........
-
-2007-06-21 21:14 +0000 [r70866-70883] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Put the thread reading from the socket back
- in the idle list if it deferred the processing of a full frame to
- another thread
-
- * channels/chan_iax2.c: If a full frame is received while one of
- the iax2 threads is in the middle of handling a full frame for
- the same call, queue it up for processing by that same thread
- later instead of dropping it. (issue #9937, patch by me)
-
-2007-06-21 20:19 +0000 [r70841] Steve Murphy <murf@digium.com>
-
- * cdr/cdr_custom.c, /: Merged revisions 70804 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r70804 | murf | 2007-06-21 13:13:17 -0600 (Thu, 21 Jun 2007) | 1
- line it was pointed out that the cdr_custom config load could get
- a lock, and under certain circumstances, would never release it.
- I also noted that the situation where more than one mapping spec
- was warned about, but did not ignore further mappings as it had
- promised. I think I have fixed both situations. ........
-
-2007-06-21 19:49 +0000 [r70808] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: When volgain is used don't leave a
- temporary file behind. (Closes Issue 8514, Reported and patched
- by ulogic, code reviewed by Jason Parker)
-
-2007-06-21 15:22 +0000 [r70727] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Do not Packet2Packet bridge if packetization settings
- do not allow it. (issue #9117 reported by phsultan)
-
-2007-06-21 15:21 +0000 [r70726] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c: Remove a couple of duplicate unlocks
-
-2007-06-21 13:58 +0000 [r70677] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c: Fix building with ODBC storage enabled.
- (issue #10025 reported by denisgalvao)
-
-2007-06-21 13:00 +0000 [r70656] Steve Murphy <murf@digium.com>
-
- * main/cdr.c: Via complaints aired in asterisk-users, I submit
- these changes, which allow cdr updates to see macro
- context/exten, whether hung up or not
-
-2007-06-20 23:32 +0000 [r70554-70612] Jason Parker <jparker@digium.com>
-
- * cdr/cdr_pgsql.c: Fix some potential memory leaks in cdr_pgsql.
- Issue 10020, patch by my, with credit to prashant_jois for
- pointing out the problem.
-
- * cdr/cdr_pgsql.c: Fix a stupid mistake in my last cdr_pgsql race
- condition fix
-
- * cdr/cdr_pgsql.c: Fix a race condition in cdr_pgsql that can occur
- when reloading the module. Issue 10022, patch by me, with credit
- to prashant_jois for finding the bug.
-
-2007-06-20 22:22 +0000 [r70552] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 70551 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r70551 | file | 2007-06-20 18:20:16 -0400 (Wed, 20 Jun 2007) | 2
- lines Don't overwrite the configured username setting upon a
- REGISTER. (issue #8565 reported by jsmith) ........
-
-2007-06-20 20:53 +0000 [r70494] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Make sure we clear the previously dialed
- number if it did not exist. Issue 9958.
-
-2007-06-20 19:29 +0000 [r70445] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_dial.c, /: Merged revisions 70444 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r70444 | tilghman | 2007-06-20 14:25:54 -0500 (Wed, 20 Jun 2007)
- | 2 lines Issue 9997 - Timelimit times out the wrong channel
- ........
-
-2007-06-20 18:46 +0000 [r70397] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 70396 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r70396 | russell | 2007-06-20 13:45:38 -0500 (Wed, 20 Jun 2007) |
- 5 lines Fix a problem where an established call would not be
- properly disconnected when a PRI disconnect is received depending
- on which cause code was received. (issue #9588, original patch by
- softins, updated patch from jtexter3, and some additional
- feedback from mhardeman) ........
-
-2007-06-20 17:52 +0000 [r70198-70360] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c, main/frame.c: Put the speex packetization values back
- in but disable it when setting up the smoother.
-
- * main/frame.c: Don't do packetization/smoother stuff with speex,
- it doesn't work.
-
-2007-06-20 00:03 +0000 [r70084-70164] Russell Bryant <russell@digium.com>
-
- * contrib/scripts/ast_grab_core: don't delete the backtrace in
- ast_grab_core
-
- * channels/chan_gtalk.c: Only attempt to queue a hangup on the
- owner channel if it actually exists. (issue #9795, patch from
- zandbelt)
-
-2007-06-19 18:23 +0000 [r70062] Steve Murphy <murf@digium.com>
-
- * main/channel.c, /: Merged revisions 70053 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r70053 | murf | 2007-06-19 12:07:59 -0600 (Tue, 19 Jun 2007) | 1
- line This fixes 9246, where channel variables are not available
- in the 'h' exten, on a 'ZOMBIE' channel. The fix is to
- consolidate the channel variables during a masquerade, and then
- copy the merged variables back onto the clone, so the zombie has
- the same vars that the 'original' has. ........
-
-2007-06-19 17:07 +0000 [r70003] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c, /: Merged revisions 69992 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r69992 | file | 2007-06-19 13:00:58 -0400 (Tue, 19 Jun 2007) | 2
- lines Handle the CC field in the RTP header. (issue #9384
- reported by DoodleHu) ........
-
-2007-06-19 16:24 +0000 [r69987] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /: Merged revisions 69986 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r69986 | file | 2007-06-19 12:21:29 -0400 (Tue, 19 Jun 2007) | 2
- lines Update BRIDGEPEER variable if set to the new channel name
- when a masquerade happens. (issue #9699 reported by dimas)
- ........
-
-2007-06-19 15:22 +0000 [r69944] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Fix a crash that could occur when handing
- device state changes. When the state of a device changes, the
- device state thread tells the extension state handling code that
- it changed. Then, the extension state code calls the callback in
- chan_sip so that it can update subscriptions to that extension. A
- pointer to a sip_pvt structure is passed to this function as the
- call which needs a NOTIFY sent. However, there was no locking
- done to ensure that the pvt struct didn't disappear during this
- process. (issue #9946, reported by tdonahue, patch by me, patch
- updated to trunk to use the sip_pvt lock wrappers by eliel)
-
-2007-06-19 13:55 +0000 [r69805-69895] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 69894 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r69894 | file | 2007-06-19 09:54:03 -0400 (Tue, 19 Jun 2007) | 2
- lines Perform an extra hangup check just in case. (issue #9589
- reported by bcnit) ........
-
- * /, res/res_features.c: Merged revisions 69846 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r69846 | file | 2007-06-19 08:57:55 -0400 (Tue, 19 Jun 2007) | 2
- lines Add parked call extension AFTER the parking slot has been
- announced, otherwise two threads will try to handle the same
- channel and it will go kaboom. (issue #9191 reported by japple)
- ........
-
- * main/callerid.c: Fix for building on PowerPC under Linux.
-
-2007-06-18 19:48 +0000 [r69796] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * channels/chan_sip.c: Issue 10005 - Segfault with missing
- arguments, plus fix a missing define for SIP INFO channels
-
-2007-06-18 19:00 +0000 [r69775-69794] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Don't count RTP timeout when involved in a
- T38 fax session. (issue #9222 reported by ivoc)
-
- * /, channels/chan_sip.c: Merged revisions 69765 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r69765 | file | 2007-06-18 14:13:03 -0400 (Mon, 18 Jun 2007) | 2
- lines Set the peer name on the dialog to the one configured in
- sip.conf and NOT the username to be used for authentication
- attempts. (issue #9967 reported by achauvin) ........
-
-2007-06-18 17:46 +0000 [r69744] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * contrib/scripts/safe_asterisk, /: Merged revisions 69743 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r69743 | tilghman | 2007-06-18 12:45:15 -0500 (Mon, 18 Jun 2007)
- | 2 lines Issue 9998 - Remove SIG prefix, since it's not
- supported by ksh ........
-
-2007-06-18 16:51 +0000 [r69708] Joshua Colp <jcolp@digium.com>
-
- * main/dnsmgr.c: Remember the DNS lookup done when dnsmgr is called
- for the first time so that it does not needlessly spit out
- changed messages when the host really didn't change.
-
-2007-06-18 16:35 +0000 [r69689-69702] Russell Bryant <russell@digium.com>
-
- * res/res_odbc.c, apps/app_voicemail.c, res/res_config_odbc.c,
- build_tools/menuselect-deps.in, configure, funcs/func_odbc.c,
- include/asterisk/autoconfig.h.in, configure.ac, cdr/cdr_odbc.c:
- To prevent 92138749238754 more reports of "I have unixodbc
- installed, but still can't build *_odbc.so!", check for ltdl
- directly, instead of just listing it as another library to
- include in the unixodbc check in the configure script. This also
- makes ltdl show up as a dependency in menuselect so people know
- what to go install. (related to issue #9989, patch by me)
-
- * build_tools/prep_moduledeps: Change the use of "echo -e" to
- "printf". On systems where /bin/sh is not bash, most of the lines
- in menuselect-tree were getting a "-e" at the beginning of every
- line. I'm surprised nobody noticed this, but I think the XML
- parser was being very nice and ignoring them.
-
-2007-06-18 16:04 +0000 [r69661-69668] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Don't defer the BYE till later on a transfer
- when the transfer itself goes kaboom and has no hope of working.
-
- * channels/chan_sip.c: Few minor transfer tweaks. We can't unlock
- something we never locked, and better handle a specific scenario
- with doing an attended transfer between two non-bridged calls.
-
-2007-06-18 15:46 +0000 [r69660] Russell Bryant <russell@digium.com>
-
- * Makefile: Tweak paths for BSD systems (issue #10001, stuarth)
-
-2007-06-18 13:55 +0000 [r69625] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Fix issue where it would be possible for the
- negotiated codecs to get set back to nothing. (issue #9992
- reported by yehavi)
-
-2007-06-15 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.4.5 released.
-
-2007-06-15 20:18 +0000 [r69579] Russell Bryant <russell@digium.com>
-
- * res/res_features.c: Fix a silly deadlock in res_features that I
- found while debugging on one of blitzrage's test machines. It was
- one of the situations where he was seeing hung channels, and may
- be the cause of some of the reports from other people. (related
- to issue #9235)
-
-2007-06-15 19:23 +0000 [r69558] Joshua Colp <jcolp@digium.com>
-
- * apps/app_speech_utils.c: Add support for setting the maximum
- length of acceptable DTMF in SpeechBackground.
-
-2007-06-15 15:27 +0000 [r69518] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c: The SLATRUNK_STATUS variable indicated
- "SUCCESS" for both an answer of the incoming call on the trunk,
- or if the trunk reached its ring timeout. This patch changes the
- variable to say "RINGTIMEOUT" in that case. (issue #9973,
- reported by n00dle, patch by me)
-
-2007-06-14 23:22 +0000 [r69434-69470] Jason Parker <jparker@digium.com>
-
- * main/config.c, /: Merged revisions 69469 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r69469 | qwell | 2007-06-14 18:21:45 -0500 (Thu, 14 Jun 2007) | 4
- lines Fix an issue where the line number in an unterminated
- comment block error message would show the wrong line number.
- "Reported" to me on #asterisk (somebody posted an error message,
- and I happened to catch it) ........
-
- * sounds/Makefile: Update to latest versions of sound files.
-
-2007-06-14 21:50 +0000 [r69392] Kevin P. Fleming <kpfleming@digium.com>
-
- * cdr/cdr_tds.c, cdr/cdr_csv.c, main/cdr.c, channels/chan_phone.c,
- cdr/cdr_sqlite.c, main/logger.c, main/callerid.c, cdr/cdr_odbc.c,
- main/asterisk.c, channels/chan_mgcp.c, cdr/cdr_manager.c,
- apps/app_voicemail.c, include/asterisk/utils.h, main/pbx.c,
- main/say.c, cdr/cdr_pgsql.c, cdr/cdr_radius.c,
- channels/chan_iax2.c: use ast_localtime() in every place
- localtime_r() was being used
-
-2007-06-14 21:08 +0000 [r69358] Russell Bryant <russell@digium.com>
-
- * main/say.c: Fix some problems with saying dates and times for the
- "tw" langauge (issue #9964, ljmid)
-
-2007-06-14 15:21 +0000 [r69259] Jason Parker <jparker@digium.com>
-
- * funcs/func_groupcount.c, /: Merged revisions 69258 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r69258 | qwell | 2007-06-14 10:15:53 -0500 (Thu, 14 Jun
- 2007) | 4 lines Change a quite broken while loop to a for loop,
- so "continue;" works as expected instead of eating 99% CPU...
- Issue 9966, patch by me. ........
-
-2007-06-13 21:19 +0000 [r69184-69222] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: Whoops...
-
- * channels/chan_iax2.c: Let's make chan_iax2 media only native
- transfers actually work. (issue #9376 reported by simone
- cittadini)
-
- * channels/iax2-parser.c: Add TXMEDIA to list so that it is
- properly displayed during iax2 packet output.
-
-2007-06-13 19:57 +0000 [r69183] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Move the logic for destroying a call when no
- response is received to a BYE outside of the block that checks
- for FLAG_FATAL to be set. This flag is only set when the packet
- is transmitted with the reliability set to XMIT_CRITICAL when the
- original packet is transmitted. A BYE is always sent with it set
- to XMIT_RELIABLE, meaning this code could never be encountered.
- This resulted in seeing some SIP channels that would never go
- away with the last packet sent being a BYE. (part of issue #9235,
- patch from jcmoore)
-
-2007-06-13 19:41 +0000 [r69181] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Contains a patch for fixing an encoding
- problem when using Outlook to view voicemail emails and
- attachments. This fix has also been tested on Thunderbird,
- Evolution, Pine, and Mutt. (Issue 9336, reported by marwick,
- patched by mutterc)
-
-2007-06-13 19:08 +0000 [r69128-69144] Joshua Colp <jcolp@digium.com>
-
- * apps/app_meetme.c: Really ignore NULL frames and check whether
- the channel hungup or not. (issue #9912 reported by junky)
-
- * /, main/app.c: Merged revisions 69127 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r69127 | file | 2007-06-13 14:12:48 -0400 (Wed, 13 Jun 2007) | 2
- lines Return group counting to previous behavior where you could
- only have one group per category. (issue #9711 reported by
- irroot) ........
-
-2007-06-13 16:56 +0000 [r69016-69071] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Clarify a bit of logic. This doesn't change
- behavior in any way, but it is helpful when following the logic
- to debug problems like 9235.
-
- * channels/chan_iax2.c: Fix a place where a chan_iax2 pvt struct
- was accessed without the lock held. This issue was reported to me
- via email by Dmitry Mishchenko. Thanks!
-
- * cdr/cdr_pgsql.c: Fix a memory leak pointed out by prashant_jois
- in #asterisk-bugs. PQclear() was not called on the result
- structure after doing a PQexec(). Also, fix up some formatting in
- passing.
-
-2007-06-12 19:36 +0000 [r69012-69014] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: Change the full frame dropping log message
- to debug to avoid future bug reports.
-
- * channels/chan_iax2.c: Schedule the sending of a PING packet a
- second later than previously so that it does not collide with the
- LAGRQ.
-
-2007-06-12 19:13 +0000 [r69010] Russell Bryant <russell@digium.com>
-
- * main/channel.c: In ast_channel_make_compatible(), just return if
- the channels' read and write formats already match up. There are
- code paths that call this function on a pair of channels multiple
- times. This made calls fail that were using g729 in some cases.
- The reason is that codec_g729a will unregister itself from the
- list of available translators will all licenses are in use. So,
- the first time the function got called, the right translation
- path was allocated. However, the second time it got called, the
- code would not find a translation path to/from g729 and make the
- call fail, even if the channel actually already had a g729
- translation path allocated. (SPD-32)
-
-2007-06-12 14:23 +0000 [r68922] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c, /: Merged revisions 68921 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r68921 | file | 2007-06-12 10:18:57 -0400 (Tue, 12 Jun 2007) | 2
- lines Bring RTP back to Asterisk at the end of a native bridge no
- matter what. ........
-
-2007-06-11 21:20 +0000 [r68814] Jason Parker <jparker@digium.com>
-
- * include/asterisk/time.h: Solaris 10 sometimes (?) needs this
- include in order to have NULL defined.
-
-2007-06-11 20:45 +0000 [r68781] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * apps/app_directory.c: Issue 9947 - fn2 was unused / incorrectly
- used
-
-2007-06-11 16:57 +0000 [r68733] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c:
- Merged revisions 68732 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r68732 | crichter | 2007-06-11 18:49:00 +0200 (Mo, 11 Jun 2007) |
- 1 line added check for NULL Pointer when calling misdn_new.
- Asterisk does not allow us to create channels anymore when stop
- gracefully is used :). also modified the restart_indicator to 0
- ........
-
-2007-06-11 14:33 +0000 [r68683] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /: Merged revisions 68682 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r68682 | file | 2007-06-11 10:29:58 -0400 (Mon, 11 Jun 2007) | 2
- lines Improve deadlock handling of the channel list. (issue #8376
- reported by one47) ........
-
-2007-06-11 10:29 +0000 [r68644] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
- channels/chan_misdn.c, /, channels/misdn/ie.c,
- channels/misdn/isdn_msg_parser.c: Merged revisions 68631 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r68631 | crichter | 2007-06-11 11:18:01 +0200 (Mo, 11 Jun 2007) |
- 1 line fixed problem that the dummybc chanels had no lock,
- checking for the lock now. Also fixed the channel restart stuff,
- we can now specify and restart particular channels too. ........
-
-2007-06-11 04:21 +0000 [r68595] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * pbx/pbx_config.c: "dialplan save" produced garbage in the config
- file
-
-2007-06-08 22:23 +0000 [r68527] Russell Bryant <russell@digium.com>
-
- * /, apps/app_dictate.c: Merged revisions 68526 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r68526 | russell | 2007-06-08 17:22:36 -0500 (Fri, 08 Jun 2007) |
- 4 lines Don't automatically hang up after running Dictate so that
- callers can exit cleanly using '#' (closes issue #9577, patch
- from Thomas Andrews) ........
-
-2007-06-08 15:52 +0000 [r68450] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_iax2.c: actually remember the type/subclass of full
- frames that are in process
-
-2007-06-08 00:17 +0000 [r68370-68401] Joshua Colp <jcolp@digium.com>
-
- * /, main/say.c: Merged revisions 68397 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r68397 | file | 2007-06-07 20:15:33 -0400 (Thu, 07 Jun 2007) | 2
- lines Don't call ast_waitstream_full when the control file
- descriptor and audio file descriptor are not set, simply call
- ast_waitstream! (issue #8530 reported by rickead2000) ........
-
- * main/dnsmgr.c, /: Merged revisions 68368 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r68368 | file | 2007-06-07 19:59:04 -0400 (Thu, 07 Jun 2007) | 2
- lines Do a DNS lookup immediately upon calling the dnsmgr
- function, don't wait until a refresh happens. (issue #9097
- reported by plack) ........
-
-2007-06-07 23:14 +0000 [r68354] Russell Bryant <russell@digium.com>
-
- * /, main/say.c: Merged revisions 68351 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r68351 | russell | 2007-06-07 18:13:33 -0500 (Thu, 07 Jun 2007) |
- 3 lines Fix a problem where saying a character wouldn't properly
- break out when the caller pressed '#' (issue #8113, reported by
- patbaker82, patch from jamesgolovich (hey, long time no see!) and
- patbaker82) ........
-
-2007-06-07 23:00 +0000 [r68326] Jason Parker <jparker@digium.com>
-
- * apps/app_voicemail.c: Fix incorrect French syntax of "old
- messages". Request for feedback was sent to asterisk-dev mailing
- list, with little response. Issue 9118, patch by junky.
-
-2007-06-07 22:14 +0000 [r68313] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_iax2.c: some improvements to the IAX2 full frame
- dropping logic recently added: - use inaddrcmp(), since we have
- it - output the type of frame and subclass being dropped, and the
- type/subclass that is already being processed (which caused the
- drop)
-
-2007-06-07 21:16 +0000 [r68280] Russell Bryant <russell@digium.com>
-
- * channels/chan_agent.c, apps/app_queue.c: Fix loading persistent
- queue members when using realtime configuration for queues. Also,
- remove an unneeded leading slash for the astdb family. (issue
- #9911, patch by atis)
-
-2007-06-07 20:25 +0000 [r68211-68249] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Fix an issue with newer phones which
- require packets be padded out to the correct length. Issue 9887,
- patch by DEA.
-
- * apps/app_voicemail.c, /: Merged revisions 68204 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r68204 | qwell | 2007-06-07 15:02:50 -0500 (Thu, 07 Jun 2007) | 4
- lines Don't try to save voicemail greetings unless the user
- presses '1' to accept/save. Issue 9904, patch by me. ........
-
-2007-06-07 19:47 +0000 [r68198] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Submitting a fix for Issue 8016. Added a
- check to make sure that greetings get stored properly. (Issue
- 8016, reported by edhorton, patched by alamantia with
- modification by me. Thanks to Jason Parker for the advice on
- this).
-
-2007-06-07 19:46 +0000 [r68196] Olle Johansson <oej@edvina.net>
-
- * channels/chan_features.c: Disable chan_features by default in
- menuselect
-
-2007-06-07 19:30 +0000 [r68192] Russell Bryant <russell@digium.com>
-
- * main/strcompat.c: Include stdarg.h for build issues on Solaris
- (issue #9381)
-
-2007-06-07 18:39 +0000 [r68071-68157] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c: Fix logic when doing a name based channel search
- for a structure when you want to start from a specific point in
- the channel list. (issue #9324 reported by slavon)
-
- * apps/app_dial.c, /: Merged revisions 68070 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r68070 | file | 2007-06-07 10:19:40 -0400 (Thu, 07 Jun 2007) | 2
- lines Allow the 'g' option to work if used with the 'S' option.
- (issue #9888 reported by gasparz) ........
-
-2007-06-07 10:00 +0000 [r67993-68030] Olle Johansson <oej@edvina.net>
-
- * res/res_jabber.c: Adding a few Todo's to res_jabber so we don't
- forget.
-
- * res/res_jabber.c: Ok, we found out that this is not about if you
- have any *active* clients using TLS, but if you have initialized
- TLS at all during the lifetime of the module. So if you reload to
- disable TLS, it won't help.
-
- * res/res_jabber.c: If you have a jabber client that uses TLS,
- refuse unload. Bad fix, but will prevent crashes while we are
- trying to find a workaround. Iksemel development seems to have
- stalled and we might have to stop using the TCP/TLS connections
- in that library and use our own, which would scale better from a
- poll/select perspective I guess. It would also make it easier to
- migrate to OpenSSL and stop Asterisk from depending on both
- OpenSSL and GnuTLS.
-
- * include/asterisk/jabber.h, res/res_jabber.c: Issue #9738 - Make
- sure we can unload res_jabber. Patch by phsultan - thanks! Due to
- a bug in the iksemel library, this will not work if you are using
- GTLS in the connection. That's being investigated. If you figure
- out a way to handle that without us having to patch iksemel, let
- us know in the bug report. Thanks.
-
-2007-06-07 00:10 +0000 [r67924-67941] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 67938 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r67938 | file | 2007-06-06 20:09:13 -0400 (Wed, 06 Jun 2007) | 2
- lines Only notify the devicestate system of a peer state change
- when the peer is built from the config file. (issue #9900
- reported by arkadia) ........
-
- * main/file.c: Properly handle cases where a stream can't be
- written to. (issue #9757 reported by junky)
-
-2007-06-06 22:08 +0000 [r67862-67872] Russell Bryant <russell@digium.com>
-
- * res/res_snmp.c: Disable reload functionality in res_snmp. It is
- not possible to initialize the snmp library more than once
- without completely unloading the module and loading it again.
- (issue #9571, reported by hristo, additional helpful debug
- information from festr, patch from me)
-
- * channels/chan_sip.c: Fix a crash when doing call pickups with SIP
- phones. The code unlocked the channel when it should not have.
- (issue #9652, reported by corruptor, fixed by me)
-
-2007-06-06 19:26 +0000 [r67804] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Fix for Issue 9810. There was a segfault
- under a specific set of circumstances: 1. VoiceMailMain was
- configured in the dialplan with an extension as its argument 2. A
- message was left for this mailbox 3. Tried to call VoiceMailMain
- but hung up before entering password. This was fixed by checking
- that a pointer was non-null prior to trying to dereference it.
- (Issue 9810, reported by xmarksthespot, patched by Corydon76 with
- modifications by me).
-
-2007-06-06 16:55 +0000 [r67716] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /, include/asterisk/linkedlists.h: Merged
- revisions 67715 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r67715 | russell | 2007-06-06 11:40:51 -0500 (Wed, 06 Jun 2007) |
- 5 lines We have some bug reports showing crashes due to a double
- free of a channel. Add a sanity check to ast_channel_free() to
- make sure we don't go on trying to free a channel that wasn't
- found in the channel list. (issue #8850, and others...) ........
-
-2007-06-06 13:30 +0000 [r67594-67650] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c, /: Merged revisions 67649 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r67649 | file | 2007-06-06 09:28:34 -0400 (Wed, 06 Jun 2007) | 2
- lines Reinvite the RTP back to the Asterisk machine when the
- timeout happens. (issue #9888 reported by gasparz) ........
-
- * main/translate.c: Fix plc_samples warning when registering a
- translator. (issue #9897 reported by xylome)
-
- * apps/app_directed_pickup.c: Include macroexten while searching
- for a channel to pick up in case they are in a macro. (issue
- #9491 reported by jamesb63)
-
- * res/res_agi.c: Make the new "agi debug off" CLI command work.
- (issue #9890 reported by eliel)
-
- * /, main/devicestate.c: Merged revisions 67593 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r67593 | file | 2007-06-06 08:18:36 -0400 (Wed, 06 Jun 2007) | 2
- lines Revert channel name splitting fix for Zap. The moral of the
- story is don't use - in your user/peer names. (issue #9668
- reported by stevedavies) ........
-
-2007-06-05 23:01 +0000 [r67558] Russell Bryant <russell@digium.com>
-
- * apps/app_meetme.c: Fix some crashes related to the use of the
- "meetme" CLI command. The code for this command was not locking
- the conference list at all. (issue #9351, reported by and patch
- submitted by Junk-Y, committed patch is different and by me)
-
-2007-06-05 21:30 +0000 [r67526] Steve Murphy <murf@digium.com>
-
- * pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/pbx_ael.c: this fixes bug
- 9883, wherein macros were not allowing the includes construct.
- fixed and tested, looks OK. Now includes can serve as an adjunct
- to catch.
-
-2007-06-05 20:53 +0000 [r67457-67492] Russell Bryant <russell@digium.com>
-
- * include/asterisk/linkedlists.h: This bug has been hanging over my
- head ever since I wrote this SLA code. Every time I tried to go
- debug it by adding some debug output, the behavior would change.
- It turns out I wasn't crazy. I had the following piece of code:
- if (remove) AST_LIST_REMOVE_CURRENT(...); Well,
- AST_LIST_REMOVE_CURRENT was not wrapped in braces, so my
- conditional statement didn't do much good at all. It always ran
- at least all of the macro minus the first statement, so I was
- seeing list entries magically disappear when they weren't
- supposed to. After many hours of debugging, I have come to this
- extremely irritating fix. :) (issues #9581, #9497)
-
- * channels/chan_zap.c: Suppress a bunch of debug output unless
- option_debug is on
-
-2007-06-05 18:32 +0000 [r67424] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Fix for bug number 9786, wherein voicemails
- saved to IMAP storage using extensions other than gsm were unable
- to be played over the phone. (Issue 9786, reporter:
- xmarksthespot, Patched by xmarksthe spot with revisions by me,
- reviewed by Russell Bryant).
-
-2007-06-05 18:18 +0000 [r67421] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Correctly update date/time on devices
- throughout the life of the device, instead of just at
- registration. Issue 9152, yet another patch by DEA.
-
-2007-06-05 18:17 +0000 [r67420] Steve Murphy <murf@digium.com>
-
- * pbx/pbx_ael.c: Added code to automatically add a default case to
- switches that don't have one. In some cases, rather than fall
- thru, it results in a goto with -1 result, which terminates the
- extension; a sort of dialplan seqfault, sort of. This was
- required to fix bug reported in 9881
-
-2007-06-05 17:07 +0000 [r67360-67372] Russell Bryant <russell@digium.com>
-
- * main/channel.c: Handle a failure in malloc() in
- ast_safe_string_alloc()
-
- * main/channel.c: Fix a problem that showed itself by causing Zap
- channel names to be completely bogus on my machine.
- ast_safe_string_alloc() was broken. It called vsnprintf() on a
- va_args list twice without re-initializing it. After the first
- usage, va_end() and va_start() must be called again.
-
-2007-06-05 16:14 +0000 [r67329-67334] Christian Richter <christian.richter@beronet.com>
-
- * /, channels/misdn/chan_misdn_config.h: Merged revisions 67307 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r67307 | crichter | 2007-06-05 17:42:03 +0200 (Di, 05 Jun 2007) |
- 1 line briding is a bool, fixed copy and paste issue. ........
-
- * channels/chan_misdn.c, /: Merged revisions 67306 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r67306 | crichter | 2007-06-05 17:39:43 +0200 (Di, 05
- Jun 2007) | 1 line simplified the EVENT_SETUP handling in the
- cb_events function a lot. Commented the different possibilities a
- bit and made functions of shared code. When the dialed extension
- does not exist in the extensions.conf we'll jump into the 'i'
- extension if this does exist, else we disconnect the call with
- the cause:1 = No Route to Destination. ........
-
-2007-06-05 15:51 +0000 [r67308] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c, main/loader.c, include/asterisk/module.h: When
- shutting down "gracefully", go through and run the unload()
- callbacks for all of the modules. "stop now" is considered a
- non-graceful shutdown and will not go through this process.
- (issue #9804, reported by chrisost, patch by me)
-
-2007-06-05 15:22 +0000 [r67304] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: Only muck with the thread structure if an
- idle one was found/created.
-
-2007-06-05 14:35 +0000 [r67270] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_iax2.c: ensure that a burst of full frames
- (AST_FRAME_DTMF being the prime example) will not be processed
- out of order... this is a brute force fix, but seems to be the
- safest fix for now (thanks to the Digium PQ department for
- finding this bug)
-
-2007-06-05 10:25 +0000 [r67210] Christian Richter <christian.richter@beronet.com>
-
- * channels/misdn_config.c, channels/chan_misdn.c, /,
- channels/misdn/chan_misdn_config.h: Merged revisions 67209 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r67209 | crichter | 2007-06-05 12:05:45 +0200 (Di, 05 Jun 2007) |
- 1 line added possibility to deactivate bridging per port ........
-
-2007-06-04 23:43 +0000 [r67162] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, funcs/func_math.c: Merged revisions 67161 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r67161 | tilghman | 2007-06-04 18:41:49 -0500 (Mon, 04 Jun 2007)
- | 2 lines According to MATH, 0+1181000386 = 1181000448. Oops.
- ........
-
-2007-06-04 23:31 +0000 [r67158] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Fix up a bunch of places where the iax2 pvt
- structure can disappear and the code did not account for it and
- crashes. (issues #9642, #9569, #9666, probably others ... based
- on the work by stevedavies and mihai, with additional changes
- from me)
-
-2007-06-04 23:26 +0000 [r67121-67156] Jason Parker <jparker@digium.com>
-
- * channels/chan_skinny.c: Fix for skinny keepalives. If there is no
- traffic from the phone for (keep_alive * 1100) ms (arbitrarily
- adding 10% for network issues, etc), unregister the device. Issue
- 8394, patch by DEA.
-
- * channels/chan_mgcp.c: Fixes for dtmf/dialing with mgcp (similar
- to the recent fix for chan_skinny) Issue 9855, patch by DEA.
-
-2007-06-04 22:28 +0000 [r67119] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Add comments for two functions that get
- called with the appropriate call locked, but perform operations
- that could result in the pvt structure getting destroyed before
- returning again, causing numerous seg faults all over the module.
- (inspired by issues #9642, #9569, and #9666, and the work done by
- stevedavies and mihai)
-
-2007-06-04 21:59 +0000 [r67073] Steve Murphy <murf@digium.com>
-
- * main/cdr.c: This typo has been here since 1.4 forked. It has been
- the source of heartburn to many a dialplan/CDR programmer.
-
-2007-06-04 21:47 +0000 [r67071] Russell Bryant <russell@digium.com>
-
- * main/rtp.c: Add a missing \n. (pointed out by jcmoore on IRC)
-
-2007-06-04 19:31 +0000 [r67064-67068] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Better handle SIP devices that say they have
- SDP content... but really don't. (issue #9398 reported by
- mthomasslo)
-
- * apps/app_dial.c: Initialize cidname variable to nothing since it
- may be used without having been touched. (issue #9661 reported by
- dimas)
-
- * res/res_features.c: Returning a value that indicates the parking
- of a call was a success when it really wasn't (because the
- parking slot selected was in use) is the wrong thing to do.
- (issue #9723 reported by mdu113)
-
-2007-06-04 17:11 +0000 [r67061] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * contrib/init.d/rc.debian.asterisk,
- contrib/init.d/rc.mandrake.asterisk, /,
- contrib/init.d/rc.redhat.asterisk,
- contrib/init.d/rc.gentoo.asterisk,
- contrib/init.d/rc.mandrake.zaptel,
- contrib/init.d/rc.slackware.asterisk: Merged revisions 67060 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r67060 | tilghman | 2007-06-04 12:10:30 -0500 (Mon, 04 Jun 2007)
- | 2 lines Add revision Id tags (by request of tzafrir) ........
-
-2007-06-04 16:02 +0000 [r67026] Russell Bryant <russell@digium.com>
-
- * configure, configure.ac: Change the configure script to build a
- test program against libcurl to make sure the results from
- curl-config can be used to compile successfully. This is intended
- to help prevent a situation where you are cross compiling, and
- the configure script finds the curl library installed on the
- host. (issue #9865, reported and patched by zandbelt)
-
-2007-06-04 15:50 +0000 [r67021] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * res/res_jabber.c: Issue 9739 - Malformed jid causes a crash
-
-2007-06-04 15:47 +0000 [r67018-67020] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c: Resolve a deadlock in chan_iax2. When
- handling an implicit ACK to a frame that was marked as the final
- transmission for a call, don't call iax2_destroy() for that call
- while the global frame queue is still locked. There is a very
- nice explanation of the deadlock in the report. (issue #9663,
- thorough report and patch from stevedavies, additional positive
- test reports from mihai and joff_oconnell)
-
- * include/asterisk/stringfields.h: Fix some compiler warnings in
- C++ modules. (issue #9866, reported by osk, patch by Corydon76)
-
-2007-06-01 21:45 +0000 [r66919] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * funcs/func_odbc.c: On some drivers, deallocating the statement
- handle isn't enough. We also have to clear the cursor (nice,
- Oracle)
-
-2007-06-01 21:31 +0000 [r66897-66917] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Removing extraneous debugging lines from
- revision 66897. Sorry :)
-
- * apps/app_voicemail.c: Submitting a fix for voicemail with IMAP
- storage. Attachments with format specified as gsm were duplicated
- (i.e. two attachments) were left. Thank you very much to
- xmarksthespot for submitting the patch that fixed this. (Issues
- 9787 and 8873, Reported by xmarksthespot and jerjer, patched by
- xmarksthespot)
-
-2007-06-01 19:41 +0000 [r66879-66881] Russell Bryant <russell@digium.com>
-
- * channels/chan_skinny.c: Changes to the way DTMF is handled in the
- core broke dialing in chan_skinny. This patch makes chan_skinny
- usable again. I did not end up testing this, but there are
- multiple positive test reports listed in the bug report. (issue
- #9596, reported by pj, testing by pj and mvanbaak, and the fix
- was written by DEA)
-
- * apps/app_page.c: List app_meetme as a module that app_page
- depends on.
-
-2007-05-31 23:03 +0000 [r66821] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * doc/asterisk.8: Issue 9850 - update preferred command line syntax
-
-2007-05-31 18:41 +0000 [r66775] Russell Bryant <russell@digium.com>
-
- * res/res_speech.c, include/asterisk/app.h,
- include/asterisk/speech.h: Change a couple of header files to not
- use "new", which is a reserved keyword in C++. (issue #9830,
- reported by osk)
-
-2007-05-31 17:15 +0000 [r66770] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, apps/app_macro.c: Merged revisions 66744 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r66744 | tilghman | 2007-05-31 10:58:45 -0500 (Thu, 31 May 2007)
- | 2 lines Issue 9818 - Fix for issue 8329 breaks pbx_realtime.
- Issue 8329 will remain unfixed for pbx_realtime, but only because
- we lack core API to do it. ........
-
-2007-05-31 16:14 +0000 [r66768] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 66764 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r66764 | file | 2007-05-31 12:12:39 -0400 (Thu, 31 May 2007) | 2
- lines It is now possible for this path of execution to have the
- frame pointer be NULL, therefore we need to check for it before
- trying to access it. (issue #9836 reported by barthpbx) ........
-
-2007-05-30 23:26 +0000 [r66671] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: Fixed seg-faults when recording greetings
- in voicemail with IMAP enabled. (Issue No. 9735, reported by
- xmarksthespot, patched by me)
-
-2007-05-30 17:28 +0000 [r66602-66639] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Silly me for having out of date source! Oh
- well... I'm still leaving my comment.
-
- * channels/chan_sip.c: When calling some peer/host that may not
- exist/reply back... don't keep the dialog in memory for all of
- eternity.
-
- * channels/chan_zap.c, channels/chan_features.c: Change how channel
- names are generated a bit. (issue #9825 reported by eldadran)
-
-2007-05-29 21:56 +0000 [r66538] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, funcs/func_strings.c: Merged revisions 66537 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r66537 | tilghman | 2007-05-29 16:49:35 -0500 (Tue, 29 May 2007)
- | 2 lines If the value of a variable passed to FIELDQTY is blank,
- then FIELDQTY should return 0, not 1. ........
-
-2007-05-29 19:32 +0000 [r66474-66503] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Properly handle 408 request timeout -
- according to the RFC, the dialog dies if a request in a dialog
- gets this response.
-
- * channels/chan_sip.c: Don't issue hangup on hangup on hangup on
- hangup (for jcmoore)
-
-2007-05-29 16:44 +0000 [r66437] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: Handle cases where a frame may have no data. (issue
- #9519 reported by dmb)
-
-2007-05-29 16:07 +0000 [r66404-66414] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c: Don't reset hangupcause if we already have
- one
-
- * channels/chan_sip.c: Tracking down hanging channels, killing them
- one by one. Issue #9235 and related
-
-2007-05-29 15:43 +0000 [r66398] Joshua Colp <jcolp@digium.com>
-
- * doc/datastores.txt: Update datastores documentation. (issue #9801
- reported by mnicholson)
-
-2007-05-29 09:41 +0000 [r66363] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 66349 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r66349 | oej | 2007-05-29 09:53:14 +0200 (Tue, 29 May 2007) | 2
- lines Issue #9802 - Change inuse counter on CANCEL ........
-
-2007-05-28 23:16 +0000 [r66312] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_zap.c: Make the usedistinctiveringdetection option
- work again. (issue #9823 reported by premeau)
-
-2007-05-27 04:12 +0000 [r66244] Jason Parker <jparker@digium.com>
-
- * channels/chan_zap.c: I don't know what this was trying to do, but
- it's clearly incorrect. Issues 9808 and 9809.
-
-2007-05-25 14:43 +0000 [r66160] Kevin P. Fleming <kpfleming@digium.com>
-
- * configure, configure.ac: have to check for OSP toolkit _after_
- checking for OpenSSL
-
-2007-05-25 14:41 +0000 [r66159] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
-
- * /, main/say.c: Merged revisions 66127 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r66127 | tilghman | 2007-05-25 08:46:35 -0500 (Fri, 25 May 2007)
- | 2 lines Issue 9791 - Fix pronunciation of seconds in Dutch
- ........
-
-2007-05-25 14:28 +0000 [r66157] Kevin P. Fleming <kpfleming@digium.com>
-
- * configure, configure.ac, channels/chan_gtalk.c, makeopts.in,
- res/res_jabber.c: handle the GNUTLS library properly in the
- configure script and build system don't build in OSP support
- unless we have found and are allowed to use SSL support
-
-2007-05-24 22:23 +0000 [r66076] Russell Bryant <russell@digium.com>
-
- * main/channel.c: if the string field init fails, clean up the
- stuff that was allocated already
-
-2007-05-24 22:16 +0000 [r66074] Joshua Colp <jcolp@digium.com>
-
- * main/slinfactory.c: Fix slinfactory logic when dealing with
- frames coming in that may already be in the signed linear format.
-
-2007-05-24 22:07 +0000 [r66068-66070] Russell Bryant <russell@digium.com>
-
- * main/channel.c: Check the result of ast_string_field_init() in
- ast_channel_alloc()
-
- * main/rtp.c: Make 1.4 build on my machine, too..
-
-2007-05-24 20:54 +0000 [r66029-66030] Jason Parker <jparker@digium.com>
-
- * configure: Rebuild configure script for previous ar fix.
-
- * configure.ac: Following moving strip to AC_PATH_TOOL, we need to
- do something similar for ar.
-
-2007-05-24 20:42 +0000 [r65978-66026] Russell Bryant <russell@digium.com>
-
- * configure, include/asterisk/autoconfig.h.in, configure.ac:
- Checking for the strip application needs to be done with
- AC_PATH_TOOL instead of AC_PATH_PROG to properly handle cross
- compilation environments.
-
- * Makefile: Clear CFLAGS before running make for menuselect. (issue
- #9784, reported by ovi, patch by me)
-
-2007-05-24 18:28 +0000 [r65965-65967] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_gtalk.c: oops, use #ifdef instead of #if
-
- * channels/chan_gtalk.c: don't reference GnuTLS headers and
- functions unless the configure script found it
-
- * main/rtp.c: don't use uninitialized variables
-
-2007-05-24 15:27 +0000 [r65902] Joshua Colp <jcolp@digium.com>
-
- * main/manager.c: Add the ability to blacklist certain commands
- from being executed using the Command AMI action. (issue #9240
- reported by junky)
-
-2007-05-24 15:26 +0000 [r65892-65901] Olle Johansson <oej@edvina.net>
-
- * channels/chan_gtalk.c: Issue 7672 - fix by zandbelt - Asterisk
- core dump since the GnuTLS interface did not support
- multithreading correctly.
-
- * channels/chan_gtalk.c: Issue 8193 - NAT issues with gtalk/STUN.
- Patch by phsultan. Thanks!
-
-2007-05-24 15:16 +0000 [r65877-65883] Jason Parker <jparker@digium.com>
-
- * .cleancount: Update cleancount for that last commit - just for
- good measure.
-
- * include/asterisk/translate.h, codecs/codec_speex.c,
- main/translate.c, codecs/codec_ilbc.c: Fix handling of
- zero-length frames when a codec is capable of native PLC. Issue
- 9183, patch by Mihai.
-
-2007-05-24 15:08 +0000 [r65866] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * funcs/func_math.c: merged qwell's func_math patch for issue 9507
-
-2007-05-24 15:08 +0000 [r65863] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c: I like it when the RTP stack compiles myself...
-
-2007-05-24 15:05 +0000 [r65857] Olle Johansson <oej@edvina.net>
-
- * channels/chan_gtalk.c: Issue 7686, fix by phsultan, NAT issues
- when calling from gtalk to SIP over nat.
-
-2007-05-24 15:04 +0000 [r65842-65853] Russell Bryant <russell@digium.com>
-
- * apps/app_festival.c: Ensure that frames are fully initialized.
- This will probably fix getting weird timestamp log messages in
- logs when using the Festival app. (issue #9781, patch by me)
-
- * main/rtp.c: Fix the calculation of the RTT for RTCP. The previous
- code would result in oscillating and incorrect data.
- Additionally, the RTT would sometimes report negative values due
- to incorrect calculations. (issue #9601, patch from davetroy)
-
-2007-05-24 14:48 +0000 [r65841] Olle Johansson <oej@edvina.net>
-
- * channels/chan_gtalk.c: Issue #8536 - Caller ID not set in CDR for
- jingle
-
-2007-05-24 14:42 +0000 [r65839] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 65837 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r65837 | file | 2007-05-24 10:40:38 -0400 (Thu, 24 May 2007) | 2
- lines Allow RFC2833 to be negotiated when an INVITE comes in
- without SDP and is not matched to a user or peer. (issue #9546
- reported by mcrawford) ........
-
-2007-05-24 14:38 +0000 [r65836] Olle Johansson <oej@edvina.net>
-
- * channels/chan_sip.c, res/res_jabber.c: Issue 8409 - phsultan -
- Fix "login" as component to jabber server. ...and, by accident,
- fix a bug in chan_sip for stopping a loop on retransmits of BYE
- requests.
-
-2007-05-24 09:37 +0000 [r65768] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /: Merged revisions 65767 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r65767 | crichter | 2007-05-24 11:19:58 +0200 (Do, 24
- Mai 2007) | 1 line we should only activate the generator in
- chan_misdn, when asterisk hask not yet taken the call
- (WAITING4DIGS state). Alerting audio will be generated fomr
- asterisk for example. ........
-
-2007-05-23 20:59 +0000 [r65677-65685] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_iax2.c: start the delayed PBX when receive voice or
- video full frames as well, and comment this delayed-PBX activity
-
- * /, channels/chan_sip.c: Merged revisions 65682 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r65682 | kpfleming | 2007-05-23 16:46:22 -0400 (Wed, 23 May 2007)
- | 2 lines ensure that variables are set on a newly created
- channel before we start a PBX on it ........
-
- * channels/chan_iax2.c: clear the 'delay PBX' flag when we are
- ready to start the PBX
-
- * channels/chan_iax2.c: don't start a PBX on a new incoming IAX2
- channel until we have some sort of response to our ACCEPT (ACK or
- anything else)
-
- * /, channels/chan_iax2.c: Merged revisions 65676 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r65676 | kpfleming | 2007-05-23 16:06:13 -0400 (Wed, 23 May 2007)
- | 2 lines if we are going to set variables on a newly created
- channel, it should be done *before* we start the PBX on it
- ........
-
-2007-05-23 13:07 +0000 [r65589] Russell Bryant <russell@digium.com>
-
- * channels/chan_zap.c, /: Merged revisions 65588 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r65588 | russell | 2007-05-23 08:06:17 -0500 (Wed, 23 May 2007) |
- 3 lines Revert revision 62417 as someone reported problems with
- it to Mark. This was related to issue #9588. ........
-
-2007-05-22 20:25 +0000 [r65541] Kevin P. Fleming <kpfleming@digium.com>
-
- * build_tools/make_version: when building a version string for a
- developer branch, include the base branch in the version string
-
-2007-05-22 18:40 +0000 [r65501] Russell Bryant <russell@digium.com>
-
- * apps/app_voicemail.c, channels/chan_zap.c: List res_smdi as a
- dependency for app_voicemail and chan_zap (Thanks to mnicholson
- for pointing it out)
-
-2007-05-22 15:04 +0000 [r65452] Joshua Colp <jcolp@digium.com>
-
- * apps/app_meetme.c: Remove a double const.
-
-2007-05-22 14:02 +0000 [r65408] BJ Weschke <bweschke@btwtech.com>
-
- * apps/app_followme.c: Fix a problem with flag recognition.
-
-2007-05-22 13:09 +0000 [r65394] Russell Bryant <russell@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 65389 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r65389 | russell | 2007-05-22 08:07:03 -0500 (Tue, 22 May 2007) |
- 4 lines Fix a memory leak that I just noticed in the device state
- handling in app_queue. On most device state changes, it would
- leak roughly 8 to 64 bytes (the length of the name of the
- device). ........
-
-2007-05-22 08:12 +0000 [r65342] Christian Richter <christian.richter@beronet.com>
-
- * channels/chan_misdn.c, /: Merged revisions 65328 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r65328 | crichter | 2007-05-22 09:46:39 +0200 (Di, 22
- Mai 2007) | 1 line we stop the tones only when we're in the
- pre-call phase, otherwise e.g. when in CONNECTED state we should
- not stop tones when we receive an Information Message ........
-
-2007-05-20 17:59 +0000 [r65250] Joshua Colp <jcolp@digium.com>
-
- * res/res_agi.c: res_agi needs to export two symbols
- (ast_agi_register and ast_agi_unregister) for usage by others.
- (issue #9755 reported by mnicholson)
-
-2007-05-18 22:26 +0000 [r65200-65201] Steve Murphy <murf@digium.com>
-
- * main/cdr.c: Ugh. The svnmerge didn't catch the shift from cdr.c
- to main/cdr.c, and neither did I. This is the remainder of the
- 9717 patch, the fix for the run-away FAIL status for a call
-
- * apps/app_dial.c, /, include/asterisk/cdr.h: Merged revisions
- 65172 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r65172 | murf | 2007-05-18 14:56:20 -0600 (Fri, 18 May 2007) | 1
- line This update will fix the situation that occurs as described
- by 9717, where when several targets are specified for a dial, if
- any one them reports FAIL, the whole call gets FAIL, even though
- others were ringing OK. I rearranged the priorities, so that a
- new disposition, NULL, is at the lowest level, and the
- disposition get init'd to NULL. Then, next up is FAIL, and next
- up is BUSY, then NOANSWER, then ANSWERED. All the related set
- routines will only do so if the disposition value to be set to is
- greater than what's already there. This gives the intended
- effect. So, if all the targets are busy, you'd get BUSY for the
- call disposition. If all get BUSY, but one, and that one rings is
- not answered, you get NOANSWER. If by some freak of nature, the
- NULL value doesn't get overridden, then the disp2str routine will
- report NOANSWER as before. ........
-
-2007-05-18 18:16 +0000 [r65041-65123] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 65122 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r65122 | oej | 2007-05-18 20:10:46 +0200 (Fri, 18 May 2007) | 2
- lines Not getting an ACK to a 200 OK in the initial invite is
- critical to the call. ........
-
- * /, channels/chan_sip.c: Merged revisions 65075 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r65075 | oej | 2007-05-18 17:12:09 +0200 (Fri, 18 May 2007) | 5
- lines Issue 9235 - part of the problem, maybe not all. Please
- retry with this patch (and no other patch) if you have problems
- with hanging SIP channels. Thank you. A special Thank You to
- WeBRainstorm that gave me access to his system. ........
-
- * channels/chan_sip.c: - Adding support for putting calls OFF hold
- with a re-invite with blank SDP. This was a bug found while doing
- tests at SIPit in Antwerp. - In order to not duplicate code, I
- restructured some of the code for putting calls on/off hold.
- Thanks DEA for reminding me. This fix has been asleep in the
- videocaps branch until now.
-
-2007-05-18 12:40 +0000 [r65039] Christian Richter <christian.richter@beronet.com>
-
- * /, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c: Merged
- revisions 65007 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r65007 | crichter | 2007-05-18 13:23:11 +0200 (Fr, 18 Mai 2007) |
- 1 line fixed a warning regarding Keypad encoding. encode the IE
- sending_complete at the right position. ........