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authorlmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b>2010-04-05 15:17:15 +0000
committerlmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b>2010-04-05 15:17:15 +0000
commit7eced4c00317ea627a7bcd92ad5c46406c19004c (patch)
treead4e7072ef78138c6ba46d882cf583047d85ea61
parentee6ae2e6c1775d3c323adecbe52095b536320638 (diff)
parent2883ed8b48771fa51656fab2f006e6b62d07b37e (diff)
Creating tag for the release of asterisk-1.6.0.27-rc1
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.0.27-rc1@256165 f38db490-d61c-443f-a65b-d21fe96a405b
-rw-r--r--.lastclean1
-rw-r--r--.version1
-rw-r--r--ChangeLog57401
-rw-r--r--doc/tex/localchannel.tex4
4 files changed, 2 insertions, 57405 deletions
diff --git a/.lastclean b/.lastclean
deleted file mode 100644
index 7facc8993..000000000
--- a/.lastclean
+++ /dev/null
@@ -1 +0,0 @@
-36
diff --git a/.version b/.version
deleted file mode 100644
index a0d26cba5..000000000
--- a/.version
+++ /dev/null
@@ -1 +0,0 @@
-1.6.0.27-rc1
diff --git a/ChangeLog b/ChangeLog
deleted file mode 100644
index e94b74876..000000000
--- a/ChangeLog
+++ /dev/null
@@ -1,57401 +0,0 @@
-2010-04-05 Leif Madsen <lmadsen@digium.com>
-
- * Asterisk 1.6.0.27-rc1 Released
-
-2010-04-02 23:47 +0000 [r256011-256016] Russell Bryant <russell@digium.com>
-
- * channels/chan_local.c, /: Merged revisions 256015 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r256015 | russell | 2010-04-02 18:46:45 -0500
- (Fri, 02 Apr 2010) | 16 lines Merged revisions 256014 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r256014 | russell | 2010-04-02 18:45:56 -0500 (Fri, 02 Apr 2010)
- | 9 lines Resolve a deadlock that occurs due to a pointless call
- to ast_bridged_channel() (closes issue #16840) Reported by:
- bzing2 Patches: patch.txt uploaded by bzing2 (license 902)
- issue_16840.rev1.diff uploaded by russell (license 2) Tested by:
- bzing2, russell ........ ................
-
- * main/channel.c, /: Merged revisions 256010 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r256010 | russell | 2010-04-02 18:30:58 -0500 (Fri, 02 Apr 2010)
- | 9 lines Merged revisions 256009 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r256009 | russell | 2010-04-02 18:30:15 -0500 (Fri, 02 Apr 2010)
- | 2 lines Remove extremely verbose debug message. ........
- ................
-
-2010-04-02 20:20 +0000 [r255953] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c, /: Merged revisions 255952 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r255952 |
- tilghman | 2010-04-02 15:19:01 -0500 (Fri, 02 Apr 2010) | 8 lines
- Pass the PID of the Asterisk process, not the PID of the canary.
- (closes issue #17065) Reported by: globalnetinc Patches:
- astcanary.patch uploaded by makoto (license 38) Tested by: frawd,
- globalnetinc ........
-
-2010-04-01 18:21 +0000 [r255674-255813] Tilghman Lesher <tlesher@digium.com>
-
- * /, include/asterisk/lock.h: Merged revisions 255796 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r255796 | tilghman | 2010-04-01 13:16:37 -0500 (Thu, 01 Apr 2010)
- | 7 lines Fix DEBUG_THREADS build on Darwin. (closes issue
- #16828) Reported by: oej Patches: 20100331__issue16828.diff.txt
- uploaded by tilghman (license 14) ........
-
- * apps/app_voicemail.c, /: Recorded merge of revisions 255592 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r255592 | tilghman | 2010-03-31 14:13:02 -0500
- (Wed, 31 Mar 2010) | 22 lines Recorded merge of revisions 255591
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r255591 | tilghman | 2010-03-31 14:09:46 -0500 (Wed, 31 Mar 2010)
- | 15 lines Ensure line terminators in email are consistent. Fixes
- an issue with certain Mail Transport Agents, where attachments
- are not interpreted correctly. (closes issue #16557) Reported by:
- jcovert Patches: 20100308__issue16557__1.4.diff.txt uploaded by
- tilghman (license 14) 20100308__issue16557__1.6.0.diff.txt
- uploaded by tilghman (license 14)
- 20100308__issue16557__trunk.diff.txt uploaded by tilghman
- (license 14) Tested by: ebroad, zktech Reviewboard:
- https://reviewboard.asterisk.org/r/544/ ........ ................
-
-2010-03-31 17:54 +0000 [r255507] Leif Madsen <lmadsen@digium.com>
-
- * apps/app_dial.c, configs/sip.conf.sample: Merged revisions 255504
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r255504 | lmadsen | 2010-03-31 12:48:09 -0500 (Wed, 31
- Mar 2010) | 5 lines Add documentation clarifying when 't' and 'T'
- can be used. (closes issue #17021) Reported by: kovzol Tested by:
- lmadsen, kovzol, davidw, ebroad ........
-
-2010-03-30 20:57 +0000 [r255324-255411] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_h323.c: Merged revisions 255410 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r255410 | russell | 2010-03-30 15:56:26 -0500
- (Tue, 30 Mar 2010) | 9 lines Merged revisions 255409 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r255409 | russell | 2010-03-30 15:56:00 -0500 (Tue, 30
- Mar 2010) | 2 lines Don't kill Asterisk if the H323 listener does
- not start. ........ ................
-
- * /, pbx/pbx_dundi.c: Merged revisions 255323 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r255323 | russell | 2010-03-30 11:07:49 -0500 (Tue, 30 Mar 2010)
- | 9 lines Merged revisions 255322 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r255322 | russell | 2010-03-30 11:06:06 -0500 (Tue, 30 Mar 2010)
- | 2 lines Don't make Asterisk not start if pbx_dundi fails to
- initialize. ........ ................
-
-2010-03-26 19:24 +0000 [r255054] Leif Madsen <lmadsen@digium.com>
-
- * /, configs/sip.conf.sample: Merged revisions 255021 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r255021 | lmadsen | 2010-03-26 14:07:38 -0500 (Fri, 26 Mar 2010)
- | 8 lines Update confusing documentation for tlsbindaddr. Update
- some confusing documentation for the tlsbindaddr option in
- sip.conf.sample. Point at a link instead which has better
- documentation. (closes issue #17054) Reported by: klaus3000
- ........
-
-2010-03-25 20:42 +0000 [r254803] Jason Parker <jparker@digium.com>
-
- * utils/Makefile, /: Merged revisions 254802 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r254802 | qwell | 2010-03-25 15:41:49 -0500 (Thu, 25 Mar 2010) |
- 9 lines Merged revisions 254800 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r254800 | qwell | 2010-03-25 15:41:15 -0500 (Thu, 25 Mar 2010) |
- 1 line Don't remove local copies of utils in uninstall. ........
- ................
-
-2010-03-25 20:09 +0000 [r254719] Russell Bryant <russell@digium.com>
-
- * channels/chan_usbradio.c, /: Merged revisions 254718 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r254718 | russell | 2010-03-25 15:08:40 -0500 (Thu, 25 Mar 2010)
- | 2 lines chan_usbradio depends on alsa. ........
-
-2010-03-25 19:59 +0000 [r254716] Jason Parker <jparker@digium.com>
-
- * main/astobj2.c, include/asterisk/astobj2.h: Fix DEBUG_THREADS
- issue with out-of-tree modules. Take 2, without ABI breakage this
- time. Review: https://reviewboard.asterisk.org/r/588/
-
-2010-03-25 17:45 +0000 [r254549-254554] Mark Michelson <mmichelson@digium.com>
-
- * include/asterisk/acl.h, /: Merged revisions 254553 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r254553 | mmichelson | 2010-03-25 12:42:36 -0500
- (Thu, 25 Mar 2010) | 11 lines Merged revisions 254552 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r254552 | mmichelson | 2010-03-25 12:33:35 -0500 (Thu, 25 Mar
- 2010) | 5 lines Add doxygen for acl.h Review:
- https://reviewboard.asterisk.org/r/528 ........ ................
-
- * channels/chan_sip.c: Undo unnecessary commit. Sean Bright beat me
- to the punch on this one.
-
- * channels/chan_sip.c: Fix potential use of uninitialized value.
-
-2010-03-25 17:19 +0000 [r254546] Sean Bright <sean@malleable.com>
-
- * channels/chan_sip.c: Initialize stream to avoid a compilation
- error.
-
-2010-03-25 17:05 +0000 [r254540] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: Fix crashes resulting from reading
- non-existent RTP streams. Specifically, when using the CHANNEL
- dialplan function, it was possible to crash Asterisk by trying to
- get the rtpdest of a stream type that is not in use by the
- channel. This commit fixes that issue.
-
-2010-03-25 17:02 +0000 [r254539] Leif Madsen <lmadsen@digium.com>
-
- * contrib/scripts/safe_asterisk, /: Make safe_asterisk work on
- dash/sh/bash etc. Merged from the change to trunk via issue
- #13111. For some reason the changes there were only done on
- trunk, and thus were available for 1.6.1 and 1.6.2 when they were
- branched. Because this change is available on both 1.6.1 and
- 1.6.2, it makes sense to allow it on the 1.6.0 branch as well.
- (closes issue #17094) Reported by: stuarth Much thanks to
- Tilghman and Sean Bright for the help on this merge. Merged
- revisions 135061 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r135061 |
- mvanbaak | 2008-08-01 07:17:33 -0500 (Fri, 01 Aug 2008) | 8 lines
- Make safe_asterisk work on dash/sh/bash etc. (closes issue
- #13111) Reported by: pabelanger Patches:
- 2008071901_issue13111_safe_asterisk.diff uploaded by mvanbaak
- (license 7) Tested by: mvanbaak, pabelanger ........
-
-2010-03-25 16:57 +0000 [r254538] Sean Bright <sean@malleable.com>
-
- * /: Unblock r135061
-
-2010-03-25 16:19 +0000 [r254466] Terry Wilson <twilson@digium.com>
-
- * /, main/file.c: Merged revisions 254453 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r254453 | twilson | 2010-03-25 11:03:51 -0500 (Thu, 25 Mar 2010)
- | 9 lines Merged revisions 254451 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r254451 | twilson | 2010-03-25 10:57:29 -0500 (Thu, 25 Mar 2010)
- | 2 lines Handle new SRCCHANGE control message here too ........
- ................
-
-2010-03-25 16:11 +0000 [r254455] Mark Michelson <mmichelson@digium.com>
-
- * main/rtp.c, /: Recorded merge of revisions 254454 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r254454 | mmichelson | 2010-03-25 11:04:48 -0500
- (Thu, 25 Mar 2010) | 50 lines Recorded merge of revisions 254452
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r254452 | mmichelson | 2010-03-25 10:59:56 -0500 (Thu, 25 Mar
- 2010) | 44 lines Several fixes regarding RFC2833 DTMF detection.
- Here is a copy and paste of the details from my request on
- reviewboard that dealt with these changes: Fix 1. The first
- change in place is to fix Mantis issue 15811, which deals with a
- situation where Asterisk will incorrectly interpret out of order
- RFC2833 frames as duplicate DTMF digits. For instance, we would
- receive a sequence like: seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3:
- DTMF 1 seqno 4: DTMF 1 seqno 6: DTMF 1 (end) seqno 5: DTMF 1
- seqno 7: DTMF 1 (end) seqno 8: DTMF 1 (end) Prior to this patch
- when we received the frame with seqno 5, we would interpret this
- as a new DTMF 1. With this patch, we will check the seqno of the
- incoming digit and not process the frame if the seqno is lower
- than the last recorded seqno. Note that we do not record the
- seqno of the dropped DTMF frame for future processing. While the
- above situation is what was designed to be fixed, the patch is
- written in such a way that the following would also be fixed too:
- seqno 9: DTMF 1 seqno 10: DTMF 1 (end) seqno 11: DTMF 1 (end)
- seqno 13: DTMF 2 seqno 12: DTMF 1 (end) seqno 14: DTMF 2 seqno
- 15: DTMF 2 (end) seqno 16: DTMF 2 (end) seqno 17: DTMF 2 (end) In
- this second situation, the beginning of the DTMF 2 arrives before
- the final end frame of the DTMF 1. With the patch, seqno 12 is no
- processed and thus we properly interpret the DTMF. Fix 2. The
- second change in place is to fix an issue like the following:
- seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: DTMF 1 (end) *packet
- lost* seqno 4: DTMF 1 (end) *packet lost* seqno 5: DTMF 1 (end)
- *packet lost* seqno 6: DTMF 2 When we receive seqno 6, we had
- code in place that was supposed to properly end the previously
- unended DTMF 1. The problem was that the code was essentially a
- no-op. The code would set up an end frame for the DTMF 1 but
- would immediately overwrite the frame with the begin for DTMF 2.
- I changed process_dtmf_rfc2833() so that instead of returning a
- single frame, it is given as an output parameter a list of
- frames. Each frame that needs to be returned is appended to this
- list. Fix 3. The final change is a minor one where an
- AST_CONTROL_SRCCHANGE frame could get lost. If we process a cisco
- DTMF or an RFC 3389 frame and no frame was returned, then we
- would return &ast_null_frame. The problem is that earlier in the
- function, we may have generated an AST_CONTROL_SRCCHANGE frame
- and put it in the list of frames we wish to return. This frame
- would be lost in such a case. The patch fixes this problem
- ........ ................
-
-2010-03-25 15:22 +0000 [r254449] Leif Madsen <lmadsen@digium.com>
-
- * /, res/res_agi.c: Merged revisions 254446 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r254446 |
- lmadsen | 2010-03-25 10:21:26 -0500 (Thu, 25 Mar 2010) | 9 lines
- handle_speechset has 4 arguments. Update code to reflect that
- handle_speechset has 4 arguments. (closes issue #17093) Reported
- by: gpatri Patches: res_agi.patch uploaded by gpatri (license
- 1014) Tested by: pabelanger, mmichelson ........
-
-2010-03-24 17:17 +0000 [r254061-254278] Jeff Peeler <jpeeler@digium.com>
-
- * res/res_monitor.c, /: Merged revisions 254277 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r254277 | jpeeler | 2010-03-24 12:15:05 -0500 (Wed, 24 Mar 2010)
- | 78 lines Merged revisions 254235 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r254235 | jpeeler | 2010-03-23 19:37:23 -0500 (Tue, 23 Mar 2010)
- | 72 lines Ensure that monitor recordings are written to the
- correct location (again) This is an extension to 248860. As such
- the dialplan test has been extended: ; non absolute path, not
- combined exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test)
- exten => 5040, n, dial(sip/5001) ; absolute path, not combined
- exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2) exten =>
- 5041, n, dial(sip/5001) ; no path, not combined exten => 5042, 1,
- monitor(wav,monitor_test3) exten => 5042, n, dial(sip/5001) ;
- combined: changemonitor from non absolute to no path (leaves
- tmp/jeff) exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test4,m)
- exten => 5043, n, changemonitor(monitor_test5) exten => 5043, n,
- dial(sip/5001) ; combined: changemonitor from no path to non
- absolute path exten => 5044, 1, monitor(wav,monitor_test6,m)
- exten => 5044, n, changemonitor(tmp/jeff/monitor_test7) ; this
- wasn't possible before exten => 5044, n, dial(sip/5001) ; non
- absolute path, combined exten => 5045, 1,
- monitor(wav,tmp/jeff/monitor_test8,m) exten => 5045, n,
- dial(sip/5001) ; absolute path, combined exten => 5046, 1,
- monitor(wav,/tmp/jeff/monitor_test9,m) exten => 5046, n,
- dial(sip/5001) ; no path, combined exten => 5047, 1,
- monitor(wav,monitor_test10,m) exten => 5047, n, dial(sip/5001) ;
- combined: changemonitor from non absolute to absolute (leaves
- tmp/jeff) exten => 5048, 1,
- monitor(wav,tmp/jeff/monitor_test11,m) exten => 5048, n,
- changemonitor(/tmp/jeff/monitor_test12) exten => 5048, n,
- dial(sip/5001) ; combined: changemonitor from absolute to non
- absolute (leaves /tmp/jeff) exten => 5049, 1,
- monitor(wav,/tmp/jeff/monitor_test13,m) exten => 5049, n,
- changemonitor(tmp/jeff/monitor_test14) exten => 5049, n,
- dial(sip/5001) ; combined: changemonitor from no path to absolute
- exten => 5050, 1, monitor(wav,monitor_test15,m) exten => 5050, n,
- changemonitor(/tmp/jeff/monitor_test16) exten => 5050, n,
- dial(sip/5001) ; combined: changemonitor from absolute to no path
- (leaves /tmp/jeff) exten => 5051, 1,
- monitor(wav,/tmp/jeff/monitor_test17,m) exten => 5051, n,
- changemonitor(monitor_test18) exten => 5051, n, dial(sip/5001) ;
- not combined: changemonitor from non absolute to no path (leaves
- tmp/jeff) exten => 5052, 1, monitor(wav,tmp/jeff/monitor_test19)
- exten => 5052, n, changemonitor(monitor_test20) exten => 5052, n,
- dial(sip/5001) ; not combined: changemonitor from no path to non
- absolute exten => 5053, 1, monitor(wav,monitor_test21) exten =>
- 5053, n, changemonitor(tmp/jeff/monitor_test22) exten => 5053, n,
- dial(sip/5001) ; not combined: changemonitor from non absolute to
- absolute (leaves tmp/jeff) exten => 5054, 1,
- monitor(wav,tmp/jeff/monitor_test23) exten => 5054, n,
- changemonitor(/tmp/jeff/monitor_test24) exten => 5054, n,
- dial(sip/5001) ; not combined: changemonitor from absolute to non
- absolute (leaves /tmp/jeff) exten => 5055, 1,
- monitor(wav,/tmp/jeff/monitor_test24) exten => 5055, n,
- changemonitor(tmp/jeff/monitor_test25) exten => 5055, n,
- dial(sip/5001) ; not combined: changemonitor from no path to
- absolute exten => 5056, 1, monitor(wav,monitor_test26) exten =>
- 5056, n, changemonitor(/tmp/jeff/monitor_test27) exten => 5056,
- n, dial(sip/5001) ; not combined: changemonitor from absolute to
- no path (leaves /tmp/jeff) exten => 5057, 1,
- monitor(wav,/tmp/jeff/monitor_test28) exten => 5057, n,
- changemonitor(monitor_test29) exten => 5057, n, dial(sip/5001)
- ........ ................
-
- * main/channel.c, /: Merged revisions 254050 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r254050 |
- jpeeler | 2010-03-23 16:17:23 -0500 (Tue, 23 Mar 2010) | 14 lines
- Exit native bridging early for greater timing accuracy with
- warnings This changes native bridging to break one millisecond
- early so that the more accurate timeval calculations done in the
- generic bridge can be performed using the bridge config.
- Currently the time between exiting native bridging slightly late
- can sometimes cause a large enough discrepancy for warnings to be
- missed. For the record, 1.4 does not attempt to native bridge at
- all when warnings are enabled. (closes issue #15815) Reported by:
- adomjan Review: https://reviewboard.asterisk.org/r/577/ ........
-
-2010-03-23 20:52 +0000 [r254044] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
-
- * tests/Makefile, /: Merged revisions 254001 via svnmerge from
- http://svn.digium.com/svn/asterisk/trunk ........ r254001 |
- tzafrir | 2010-03-23 21:19:52 +0200 (Tue, 23 Mar 2010) | 2 lines
- Change the name of the category 'TEST' to match the name of the
- subdir ........
-
-2010-03-22 19:57 +0000 [r253803] Matthew Nicholson <mnicholson@digium.com>
-
- * /, main/features.c: Merged revisions 253800 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r253800 | mnicholson | 2010-03-22 14:52:52 -0500 (Mon, 22 Mar
- 2010) | 11 lines Merged revisions 253799 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r253799 | mnicholson | 2010-03-22 14:50:00 -0500 (Mon, 22 Mar
- 2010) | 4 lines Unconditionally copy the caller's account code to
- the called party. (related to issue #16331) ........
- ................
-
-2010-03-20 18:29 +0000 [r253625-253630] Russell Bryant <russell@digium.com>
-
- * main/sched.c, main/manager.c, main/features.c,
- apps/app_waituntil.c, main/logger.c: Resolve 1.6.0 compilation
- issues on FreeBSD.
-
- * apps/app_dial.c, channels/chan_dahdi.c, main/tcptls.c, /,
- main/features.c, pbx/pbx_dundi.c, cdr/cdr_pgsql.c,
- main/stdtime/localtime.c, apps/app_followme.c: Merged revisions
- 253536-253538,253540 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r253536 |
- russell | 2010-03-20 06:33:30 -0500 (Sat, 20 Mar 2010) | 4 lines
- Use SHRT_MAX instead of MAXSHORT. These changes fix build issues
- I had with this module on FreeBSD. ........ r253537 | russell |
- 2010-03-20 06:39:39 -0500 (Sat, 20 Mar 2010) | 2 lines Resolve a
- compiler warning on FreeBSD. ........ r253538 | russell |
- 2010-03-20 06:43:08 -0500 (Sat, 20 Mar 2010) | 2 lines Resolve
- compiler warnings on FreeBSD. ........ r253540 | russell |
- 2010-03-20 07:03:07 -0500 (Sat, 20 Mar 2010) | 2 lines Resolve
- more compiler warnings on FreeBSD. ........
-
- * main/utils.c: Resolve compiler warnings on FreeBSD.
-
-2010-03-18 17:56 +0000 [r253259-253348] Leif Madsen <lmadsen@digium.com>
-
- * apps/app_userevent.c: Slightly different fix for UserEvent docs
- update. (issue #16961)
-
- * doc/tex/localchannel.tex: Merged revisions 253256 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r253256 | lmadsen | 2010-03-18 10:46:52 -0500 (Thu, 18 Mar 2010)
- | 9 lines Update to new Local channel documentation. Add same
- changes as commit to 1.4, but convert to TeX. (issue #16963)
- Reported by: kobaz Patches: localchannel-2.txt uploaded by kobaz
- (license 834) ........
-
-2010-03-17 16:25 +0000 [r253158] Terry Wilson <twilson@digium.com>
-
- * main/rtp.c, channels/chan_skinny.c, channels/chan_h323.c,
- channels/chan_mgcp.c, channels/chan_sip.c,
- include/asterisk/rtp.h: Revert API change in release branches
- This re-renames ast_rtp_update_source to ast_rtp_new_source
-
-2010-03-17 00:31 +0000 [r253031] Leif Madsen <lmadsen@digium.com>
-
- * /, configs/say.conf.sample: Merged revisions 253028 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r253028 | lmadsen | 2010-03-16 19:29:06 -0500
- (Tue, 16 Mar 2010) | 13 lines Merged revisions 253018 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r253018 | lmadsen | 2010-03-16 19:26:19 -0500 (Tue, 16 Mar 2010)
- | 6 lines Add french snipset to say.conf. Add the french snipset
- to say.conf. (Closes issue #15799) ........ ................
-
-2010-03-16 23:54 +0000 [r252979] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_stack.c, /: Merged revisions 252976 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r252976 |
- tilghman | 2010-03-16 18:49:35 -0500 (Tue, 16 Mar 2010) | 8 lines
- Mask out previous arguments on each nested invocation of Gosub.
- (closes issue #16758) Reported by: wdoekes Patches:
- 20100316__issue16758.diff.txt uploaded by tilghman (license 14)
- Review: https://reviewboard.asterisk.org/r/561/ ........
-
-2010-03-16 19:01 +0000 [r252768] Russell Bryant <russell@digium.com>
-
- * utils/Makefile, /: Merged revisions 252767 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r252767 | russell | 2010-03-16 14:01:04 -0500 (Tue, 16 Mar 2010)
- | 13 lines Merged revisions 252766 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r252766 | russell | 2010-03-16 14:00:43 -0500 (Tue, 16 Mar 2010)
- | 6 lines Don't treat warnings as errors for muted. muted
- supports OS X, but uses functions marked as deprecated in 10.6.
- However, the functions are still supported, so just ignore the
- warnings for now and allow the build to proceed. ........
- ................
-
-2010-03-16 18:49 +0000 [r252765] Leif Madsen <lmadsen@digium.com>
-
- * /, configs/extensions.ael.sample: Merged revisions 252762 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r252762 | lmadsen | 2010-03-16 13:48:22 -0500
- (Tue, 16 Mar 2010) | 15 lines Merged revisions 252761 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r252761 | lmadsen | 2010-03-16 13:46:20 -0500 (Tue, 16 Mar 2010)
- | 7 lines Additional extensions.ael global variable fixes. Fixing
- up a couple more overlapping global variable namespaces shared
- with extensions.conf.sample. Also noticed a few of the lines that
- were commented out didn't have the closing semi-colon so I added
- that as well. (issue #17035) ........ ................
-
-2010-03-15 21:59 +0000 [r252624] Sean Bright <sean@malleable.com>
-
- * /, apps/app_meetme.c: Merged revisions 252623 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r252623 |
- seanbright | 2010-03-15 17:55:44 -0400 (Mon, 15 Mar 2010) | 4
- lines Resolve a crash in SLATrunk when the specified trunk
- doesn't exist. Reported by philipp64 in #asterisk-dev. ........
-
-2010-03-15 21:53 +0000 [r252620] Tilghman Lesher <tlesher@digium.com>
-
- * contrib/init.d/org.asterisk.asterisk.plist, /: Merged revisions
- 252619 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r252619 | tilghman | 2010-03-15 16:51:55 -0500 (Mon, 15 Mar 2010)
- | 9 lines Merged revisions 252617 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r252617 | tilghman | 2010-03-15 16:43:14 -0500 (Mon, 15 Mar 2010)
- | 2 lines Uh, yeah. Umask. I'm stupid. ........ ................
-
-2010-03-15 20:54 +0000 [r252537] Leif Madsen <lmadsen@digium.com>
-
- * configs/extensions.ael.sample: Merged revisions 252534 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r252534 | lmadsen | 2010-03-15 15:52:32 -0500
- (Mon, 15 Mar 2010) | 15 lines Merged revisions 252533 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r252533 | lmadsen | 2010-03-15 15:48:56 -0500 (Mon, 15 Mar 2010)
- | 7 lines Update extensions.ael file to not overlap
- extensions.conf. Updated the extensions.ael file so the global
- variables don't overlap those that we have in extensions.conf
- (sample files). This way unexpected things won't happed hopefully
- if both pbx_ael and res_config are loaded. (closes issue #17035)
- Reported by: pprindeville ........ ................
-
-2010-03-15 01:37 +0000 [r252363] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c, Makefile,
- contrib/init.d/org.asterisk.asterisk.plist (added), /: Merged
- revisions 252362 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r252362 | tilghman | 2010-03-14 20:37:04 -0500 (Sun, 14 Mar 2010)
- | 11 lines Merged revisions 252361 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r252361 | tilghman | 2010-03-14 20:33:50 -0500 (Sun, 14 Mar 2010)
- | 4 lines Launch Asterisk on Mac OS X with launchd. Reviewboard:
- https://reviewboard.asterisk.org/r/551/ ........ ................
-
-2010-03-14 17:45 +0000 [r252315] Sean Bright <sean@malleable.com>
-
- * cdr/cdr_sqlite3_custom.c, /: Merged revisions 252314 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r252314 | seanbright | 2010-03-14 13:43:46 -0400 (Sun, 14 Mar
- 2010) | 8 lines Fix building CDR and CEL SQLite3 modules. They
- added a sqlite3_log() function which was conflicting with our
- function names. (closes issue #17017) Reported by: alephlg
- ........
-
-2010-03-13 00:30 +0000 [r252134-252176] Terry Wilson <twilson@digium.com>
-
- * main/rtp.c: Remove unused field
-
- * main/rtp.c, channels/chan_mgcp.c, main/channel.c,
- channels/chan_sip.c, channels/chan_skinny.c,
- include/asterisk/rtp.h, channels/chan_h323.c,
- configs/sip.conf.sample, include/asterisk/frame.h: Merged
- revisions 252089 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r252089 |
- twilson | 2010-03-12 16:04:51 -0600 (Fri, 12 Mar 2010) | 20 lines
- Only change the RTP ssrc when we see that it has changed This
- change basically reverts the change reviewed in
- https://reviewboard.asterisk.org/r/374/ and instead limits the
- updating of the RTP synchronization source to only those times
- when we detect that the other side of the conversation has
- changed the ssrc. The problem is that SRCUPDATE control frames
- are sent many times where we don't want a new ssrc, including
- whenever Asterisk has to send DTMF in a normal bridge. This is
- also not the first time that this mistake has been made. The
- initial implementation of the ast_rtp_new_source function also
- changed the ssrc--and then it was removed because of this same
- issue. Then, we put it back in again to fix a different issue.
- This patch attempts to only change the ssrc when we see that the
- other side of the conversation has changed the ssrc. It also
- renames some functions to make their purpose more clear. Review:
- https://reviewboard.asterisk.org/r/540/ ........
-
-2010-03-12 19:53 +0000 [r251995] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_dahdi.c: Forward declaring dahdi_pri was already
- done.
-
-2010-03-12 19:49 +0000 [r251992] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 251989 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r251989 | tilghman | 2010-03-12 13:43:23 -0600 (Fri, 12 Mar 2010)
- | 8 lines Don't override a user option with the global option.
- (closes issue #16849) Reported by: ip-rob Patches:
- 20100311__issue16849.diff.txt uploaded by tilghman (license 14)
- Tested by: ip-rob ........
-
-2010-03-12 19:42 +0000 [r251988] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 251987 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r251987 | rmudgett | 2010-03-12 13:40:16 -0600
- (Fri, 12 Mar 2010) | 9 lines Merged revisions 251986 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r251986 | rmudgett | 2010-03-12 13:33:22 -0600 (Fri, 12
- Mar 2010) | 1 line Make chan_dahdi wakeup_sub() prototype not
- conditional. ........ ................
-
-2010-03-11 21:08 +0000 [r251882-251885] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_exec.c: Merged revisions 251884 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r251884 |
- tilghman | 2010-03-11 15:07:07 -0600 (Thu, 11 Mar 2010) | 8 lines
- Because ExecIf needs to reprocess arguments, it's best if we
- don't remove quotes during parsing. (closes issue #16905)
- Reported by: ip-rob Patches: 20100303__issue16905.diff.txt
- uploaded by tilghman (license 14) Tested by: ip-rob ........
-
- * /, apps/app_system.c: Merged revisions 251877 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r251877 |
- tilghman | 2010-03-11 14:25:02 -0600 (Thu, 11 Mar 2010) | 8 lines
- If the argument to the system application is quoted, ensure we
- remove the quotes before trying to execute. (closes issue #16842)
- Reported by: ip-rob Patches: 20100310__issue16842.diff.txt
- uploaded by tilghman (license 14) Tested by: ip-rob ........
-
-2010-03-11 Leif Madsen <lmadsen@digium.com>
-
- * Asterisk 1.6.0.26 released
-
-2010-03-04 Leif Madsen <lmadsen@digium.com>
-
- * Asterisk 1.6.0.26-rc1 released
-
-2010-03-03 21:27 +0000 [r250612] Leif Madsen <lmadsen@digium.com>
-
- * doc/tex/localchannel.tex: Merged revisions 250609 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r250609 | lmadsen | 2010-03-03 16:22:55 -0500 (Wed, 03 Mar 2010)
- | 11 lines Update existing Local channel documentation. A
- complete re-write of the Local channel documentation has been
- performed, with the existing information from localchannel.txt
- and localchannel.tex merged in. (closes issue #16637) Reported
- by: kobaz Patches: localchannel.tex uploaded by lmadsen (license
- 10) localchannel.txt uploaded by lmadsen (license 10) Tested by:
- lmadsen, jsmith, mmichelson ........
-
-2010-03-03 19:07 +0000 [r250482] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 250481 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r250481 | jpeeler | 2010-03-03 13:06:06 -0600
- (Wed, 03 Mar 2010) | 22 lines Merged revisions 250480 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r250480 | jpeeler | 2010-03-03 13:04:11 -0600 (Wed, 03 Mar 2010)
- | 15 lines Make sure to clear red alarm after polarity reversal.
- From the issue: The automatic overnight line tests (or manual
- ones) used on UK (BT) lines causes a red alarm on a dahdi /
- TDM400P connected channel. This is because the line uses voltage
- tests (battery loss) and polarity reversal. The polarity reversal
- causes chan_dahdi to initiate v23 CallerID processing but during
- this the event DAHDI_EVENT_NOALARM is ignored so that the alarm
- is never cleared. (closes issue #14163) Reported by: jedi98
- Patches: chan_dahdi-1.4-inalarm.diff uploaded by jedi98 (license
- 653) Tested by: mattbrown, Chainsaw, mikeeccleston ........
- ................
-
-2010-03-03 18:06 +0000 [r250265-250398] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 250395 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r250395 | dvossel | 2010-03-03 12:03:19 -0600
- (Wed, 03 Mar 2010) | 22 lines Merged revisions 250394 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r250394 | dvossel | 2010-03-03 12:02:27 -0600 (Wed, 03 Mar 2010)
- | 16 lines fixes problem with duplicate TXREQ packets When
- Asterisk receives an IAX2 TXREQ packet, try_transfer() will call
- store_by_transfercallno() to link the chan_iax2_pvt struct into
- iax_transfercallno_pvts. If a duplicate TXREQ packet is received
- for the same call, the pvt struct will be linked into
- iax_transfercallno_pvts multiple times. This patch fixes this.
- Thanks rain for debugging this and providing a patch! (closes
- issue #16904) Reported by: rain Patches:
- iax2-double-txreq-fix.diff uploaded by rain (license 327) Tested
- by: rain, dvossel ........ ................
-
- * /, channels/chan_sip.c: Merged revisions 250246 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r250246 |
- dvossel | 2010-03-02 18:18:28 -0600 (Tue, 02 Mar 2010) | 2 lines
- fixes signed to unsigned int comparision issue for FaxMaxDatagram
- value. ........
-
-2010-03-02 21:11 +0000 [r250040-250054] Leif Madsen <lmadsen@digium.com>
-
- * doc/tex/imapstorage.tex, /: Merged revisions 250051 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r250051 | lmadsen | 2010-03-02 16:09:27 -0500 (Tue, 02 Mar 2010)
- | 8 lines Update IMAP documentation. Update the IMAP
- documentation to make it clear that storing voicemails in the
- same folder as a large number of emails could potentially cause
- significant slow downs when writing or retrieving voicemails.
- (issue #16704) Reported by: TimeHider Tested by: lmadsen,
- TimeHider ........
-
- * configs/cdr.conf.sample: Merged revisions 250045 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r250045 | lmadsen | 2010-03-02 15:52:19 -0500
- (Tue, 02 Mar 2010) | 15 lines Merged revisions 250043 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r250043 | lmadsen | 2010-03-02 15:51:35 -0500 (Tue, 02 Mar 2010)
- | 7 lines Update documentation to clarify purpose of unanswered
- option. (closes issue #16267) Reported by: elsto Patches:
- cdr.conf.sample.patch.txt uploaded by lmadsen (license 10) Tested
- by: davidw, elsto ........ ................
-
- * doc/tex/configuration.tex: Merged revisions 250037 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r250037 | lmadsen | 2010-03-02 15:36:10 -0500 (Tue, 02 Mar 2010)
- | 4 lines Update documentation to not imply we support overriding
- options. (closes issue #16855) Reported by: davidw ........
-
-2010-03-02 19:45 +0000 [r249948] Alec L Davis <sivad.a@paradise.net.nz>
-
- * apps/app_echo.c: revert ability to exit echo app caused a
- regression, as only supported VOICE, not VIDEO etc. (issue
- #16880)
-
-2010-03-02 19:20 +0000 [r249907] David Vossel <dvossel@digium.com>
-
- * channels/chan_oss.c, channels/misdn_config.c,
- include/asterisk/abstract_jb.h, configs/alsa.conf.sample,
- channels/chan_jingle.c, channels/chan_usbradio.c,
- channels/chan_dahdi.c, channels/chan_skinny.c,
- configs/mgcp.conf.sample, main/abstract_jb.c,
- channels/chan_h323.c, channels/chan_alsa.c,
- configs/sip.conf.sample, channels/chan_mgcp.c,
- channels/chan_unistim.c, configs/console.conf.sample,
- configs/chan_dahdi.conf.sample, channels/chan_local.c,
- configs/oss.conf.sample, channels/chan_sip.c, /,
- configs/usbradio.conf.sample, configs/misdn.conf.sample,
- channels/chan_gtalk.c, channels/chan_console.c: Merged revisions
- 249893 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r249893 |
- dvossel | 2010-03-02 13:08:38 -0600 (Tue, 02 Mar 2010) | 11 lines
- fixes adaptive jitterbuffer configuration When configuring the
- adaptive jitterbuffer, the target_extra value not only could not
- be set from the configuration, but was not even being set to its
- proper default. This value is required in order for the adaptive
- jitterbuffer to work correctly. To resolve this a config option
- has been added to expose this value to the conf files, and a
- default value is provided when no config specific value is
- present. ........
-
-2010-03-02 09:16 +0000 [r249846] Alec L Davis <sivad.a@paradise.net.nz>
-
- * apps/app_echo.c: fixes ability to exit echo app when called from
- a ISDN channel, null frames prevent '#' exit. Now only echo back
- VOICE and DTMF frames (closes issue #16880) Reported by:
- alecdavis Patches: based on echo_exit_1-6-1.diff.txt uploaded by
- alecdavis (license 585) Tested by: alecdavis
-
-2010-03-01 19:38 +0000 [r249673] Sean Bright <sean@malleable.com>
-
- * apps/app_voicemail.c, /: Merged revisions 249672 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r249672 | seanbright | 2010-03-01 14:36:30 -0500
- (Mon, 01 Mar 2010) | 18 lines Merged revisions 249671 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r249671 | seanbright | 2010-03-01 14:35:01 -0500 (Mon, 01 Mar
- 2010) | 11 lines Fix crash in app_voicemail related to message
- counting. We were passing a 'struct inprocess **' and treating it
- like a 'struct inprocess *' causing a segfault. (closes issue
- #16921) Reported by: whardier Patches: 20100301_issue16921.patch
- uploaded by seanbright (license 71) Tested by: whardier ........
- ................
-
-2010-03-01 17:13 +0000 [r249539] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_local.c, /: Merged revisions 249538 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r249538 | jpeeler | 2010-03-01 11:11:31 -0600
- (Mon, 01 Mar 2010) | 18 lines Merged revisions 249536 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r249536 | jpeeler | 2010-03-01 11:02:03 -0600 (Mon, 01 Mar 2010)
- | 11 lines Modify queued frames from local channels to not set
- the other side to up In this case, attended transfers were broken
- due to ast_feature_request_and_dial detecting the channel being
- set to up before the answer frame could be read and therefore
- failing to mark the channel as ready. This fix is a regression
- fix for 244785, which should continue to work properly as well.
- (closes issue #16816) Reported by: jamhed Tested by: jamhed,
- corruptor ........ ................
-
-2010-02-27 23:38 +0000 [r249364] Alec L Davis <sivad.a@paradise.net.nz>
-
- * channels/chan_dahdi.c: overlap receiving: automatically send CALL
- PROCEEDING when dialplan starts Following Q.931 5.2.4 When the
- user has determined that sufficient call information has been
- received the user shall stop T302 and send CALL PROCEEDING to the
- network. Previously timeouts were possible if the dialplan took a
- long time to issue any response back to the network. Verified
- that our local TELCO also does the same. (issue #16789) Reported
- by: alecdavis Patches: overlap_receiving_trunk.diff.txt uploaded
- by alecdavis (license 585) Tested by: alecdavis (closes issue
- #16789)
-
-2010-02-27 14:09 +0000 [r249236] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 249235 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r249235 | kpfleming | 2010-02-27 09:08:35 -0500
- (Sat, 27 Feb 2010) | 9 lines Merged revisions 249234 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r249234 | kpfleming | 2010-02-27 09:07:59 -0500 (Sat, 27
- Feb 2010) | 1 line add a reference to the now-published IAX2 RFC
- ........ ................
-
-2010-02-26 17:05 +0000 [r249102] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 249101 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r249101 | mmichelson | 2010-02-26 11:04:58 -0600 (Fri, 26 Feb
- 2010) | 14 lines Merged revisions 249100 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r249100 | mmichelson | 2010-02-26 11:04:29 -0600 (Fri, 26 Feb
- 2010) | 8 lines For T.38 reINVITEs treat a 606 the same as a 488.
- (closes issue #16792) Reported by: vrban Patches: t38_606.patch
- uploaded by vrban (license 756) ........ ................
-
-2010-02-25 23:11 +0000 [r248953] Jeff Peeler <jpeeler@digium.com>
-
- * res/res_monitor.c, /: Merged revisions 248952 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r248952 | jpeeler | 2010-02-25 17:09:54 -0600 (Thu, 25 Feb 2010)
- | 24 lines Merged revisions 248860 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r248860 | jpeeler | 2010-02-25 15:22:06 -0600 (Thu, 25 Feb 2010)
- | 18 lines Ensure that monitor recordings are written to the
- correct location (again) This is an extension to 248757. As such
- the dialplan test has been extended: exten => 5040, 1,
- monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n,
- dial(sip/5001) exten => 5041, 1,
- monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n,
- dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b)
- exten => 5042, n, dial(sip/5001) exten => 5043, 1,
- monitor(wav,tmp/jeff/monitor_test3,m) exten => 5043, n,
- changemonitor(monitor_test4) exten => 5043, n, dial(sip/5001)
- exten => 5044, 1, monitor(wav,monitor_test4,m) exten => 5044, n,
- changemonitor(tmp/jeff/monitor_test5) ; this looks to fail by
- design and emits a warning exten => 5044, n, dial(sip/5001)
- ........ ................
-
-2010-02-25 22:42 +0000 [r248947] Mark Michelson <mmichelson@digium.com>
-
- * /, main/acl.c: Merged revisions 248946 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r248946 |
- mmichelson | 2010-02-25 16:41:48 -0600 (Thu, 25 Feb 2010) | 5
- lines Fix incorrect ACL behavior when CIDR notation of "/0" is
- used. AST-2010-003 ........
-
-2010-02-25 21:24 +0000 [r248862] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c, /: Merged revisions 248861 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r248861 | tilghman | 2010-02-25 15:22:39 -0600 (Thu, 25 Feb 2010)
- | 22 lines Merged revisions 248859 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r248859 | tilghman | 2010-02-25 15:21:05 -0600 (Thu, 25 Feb 2010)
- | 15 lines Some platforms clear /var/run at boot, which makes
- connecting a remote console... difficult. Previously, we only
- created the default /var/run/asterisk directory at install time.
- While we could create it in the init script, that would not work
- for those who start asterisk manually from the command line. So
- the safest thing to do is to create it as part of the Asterisk
- boot process. This also changes the ownership of the directory,
- because the pid and ctl files are created after we setuid/setgid.
- (closes issue #16802) Reported by: Brian Patches:
- 20100224__issue16802.diff.txt uploaded by tilghman (license 14)
- Tested by: tzafrir ........ ................
-
-2010-02-25 18:46 +0000 [r248795] Jeff Peeler <jpeeler@digium.com>
-
- * res/res_monitor.c, /: Merged revisions 248793 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r248793 | jpeeler | 2010-02-25 12:37:56 -0600 (Thu, 25 Feb 2010)
- | 22 lines Merged revisions 248757 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r248757 | jpeeler | 2010-02-25 12:06:54 -0600 (Thu, 25 Feb 2010)
- | 15 lines Ensure that monitor recordings are written to the
- correct location. Recordings should be placed in the monitor
- directory when a non-absolute path is used. Exact dialplan used
- for testing: exten => 5040, 1,
- monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n,
- dial(sip/5001) exten => 5041, 1,
- monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n,
- dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b)
- exten => 5042, n, dial(sip/5001) ABE-2101 ........
- ................
-
-2010-02-24 21:23 +0000 [r248613] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/logger.c: Merged revisions 248584 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r248584 | tilghman | 2010-02-24 15:17:26 -0600 (Wed, 24 Feb 2010)
- | 14 lines Merged revisions 248582 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r248582 | tilghman | 2010-02-24 15:02:18 -0600 (Wed, 24 Feb 2010)
- | 7 lines Remove color code sequences from verbose messages that
- go to logfiles. (closes issue #16786) Reported by: dodo Patches:
- logger2.patch uploaded by dodo (license 989) Tested by: tilghman
- ........ ................
-
-2010-02-23 16:51 +0000 [r248400] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 248397 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r248397 | dvossel | 2010-02-23 10:34:39 -0600 (Tue, 23 Feb 2010)
- | 15 lines Merged revisions 248396 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r248396 | dvossel | 2010-02-23 10:26:05 -0600 (Tue, 23 Feb 2010)
- | 9 lines fixes invite with replaces deadlock (closes issue
- #16862) Reported by: pwalker Patches: replaces_deadlock_1.4
- uploaded by dvossel (license 671) Tested by: pwalker, dvossel
- ........ ................
-
-2010-02-19 19:04 +0000 [r247933-248008] Tilghman Lesher <tlesher@digium.com>
-
- * main/loader.c, /, channels/chan_console.c: Merged revisions
- 228798 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk (closes issue
- #16470) Reported by: kjotte ........ r228798 | tilghman |
- 2009-11-09 01:37:52 -0600 (Mon, 09 Nov 2009) | 14 lines Fix
- various problems detected with Valgrind. * chan_console accessed
- pvts after deallocation. * The module loader did not check
- usecount on shutdown, which led to chan_iax2 reading a timer that
- was already unloaded. (closes issue #16062) Reported by:
- alexanderheinz Patches: 20091109__issue16062.diff.txt uploaded by
- tilghman (license 14) Tested by: tilghman ........
-
- * main/ast_expr2f.c: Restore generated file from flex source
-
-2010-02-19 18:13 +0000 [r247919-247922] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_misdn.c, /: Merged revisions 247914 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r247914 | rmudgett | 2010-02-19 11:33:33 -0600
- (Fri, 19 Feb 2010) | 62 lines Merged revisions 247910 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r247910 | rmudgett | 2010-02-19 11:18:49 -0600
- (Fri, 19 Feb 2010) | 55 lines Merged revision 247904 from
- https://origsvn.digium.com/svn/asterisk/be/branches/C.2-...
- .......... r247904 | rmudgett | 2010-02-19 10:49:44 -0600 (Fri,
- 19 Feb 2010) | 49 lines Make chan_misdn DTMF processing
- consistent with other channel technologies. The processing of
- DTMF tones on the receiving side of an ISDN channel is
- inconsistent with the way it is handled in other channels,
- especially DAHDI analog. This causes DTMF tones sent from an ISDN
- phone to be doubled at the connected party. We are using the
- following 2 options of misdn.conf 1) astdtmf=yes 2) senddtmf=yes
- Option one is necessary because the asterisk DSP DTMF detection
- is better than mISDN's internal DSP. Not as many false positives.
- Option two is necessary to transmit DTMF tones end to end when
- mISDN channels are connected to SIP channels with out of band
- DTMF for example. The symptom is that DTMF tones sent by an ISDN
- phone are doubled on the way through asterisk when two mISDN
- channels are connected with a Local channel in between or if it
- is bridged to an analog channel. The doubling of DTMF tones is
- because DTMF is passed inband to asterisk by the mISDN channel
- and passed out of band once again after the release of the DTMF
- tone. Passing it inband is wrong. Neither an analog channel nor
- SIP channel passes DTMF inband if configured to inband DTMF.
- Analog and SIP channels filter out the DTMF tones because they
- use the voice frames returned by ast_dsp_process. But chan_misdn
- passes the unfiltered input voice frames instead. To overcome one
- aspect of the problem, the doubling of DTMF tones when two mISDN
- channels are directly bridged, someone made an 'optimization',
- where in that case the DTMF tone passed out-of-band to the peer
- channel is not translated to an inband tone at the transmit side.
- This optimization is bad because it does not work in general. For
- example, analog channels or mISDN channels when bridged through
- an intermediary local channel will generate DTMF tones from
- out-of-band information. Also, of course, it must not be done
- when there is no inband DTMF available. This patch fixes the
- issue. Now chan_misdn will filter the received inband DTMF signal
- the same as other channel types. Another change included: No need
- to build an extra translation path because ast_process_dsp does
- it if required. Patches: misdn-dtmf.patch JIRA ABE-2080
- ................ ................
-
- * main/ast_expr2f.c: Restore fwrite() line so ast_expr2f.c can
- compile.
-
-2010-02-18 23:15 +0000 [r247842] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_speech.c, /: Merged revisions 247841 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r247841 |
- tilghman | 2010-02-18 17:13:46 -0600 (Thu, 18 Feb 2010) | 7 lines
- Revert an errant part of a previous cleanup, to fix a memory
- corruption issue. (closes issue #16368) Reported by: thirionjwf
- Patches: res_speech.c.patch uploaded by thirionjwf (license 955)
- ........
-
-2010-02-18 22:45 +0000 [r247839] David Vossel <dvossel@digium.com>
-
- * channels/chan_sip.c: fixes dialog ref count crash isolated to the
- 1.6.0 branch (closes issue #16375) Reported by: kobaz (closes
- issue #16796) Reported by: kobaz
-
-2010-02-18 21:47 +0000 [r247789] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 247787 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r247787 |
- tilghman | 2010-02-18 15:42:53 -0600 (Thu, 18 Feb 2010) | 17
- lines If the peer record is from realtime, it could be set to 0,
- due to MySQL not representing NULL well in integer columns. NULL
- means the value is not specified for the column, which normally
- means the driver uses whatever is the default value. However, on
- MySQL, placing a NULL in either a float or integer column results
- in a retrieval of the 0 value. Hence, users get an errant error
- on load. This patch suppresses that error and makes the value as
- if it was not there. Note that this cannot be done in the
- realtime driver, because the lack of difference between NULL and
- 0 can only be intepreted correctly by the driver itself. If we
- did it in the realtime driver, then it would be effectively
- impossible to set any realtime field to 0, because it would act
- as if the field were unspecified and possibly take on a different
- value. (closes issue #16683) Reported by: wdoekes ........
-
-2010-02-18 19:45 +0000 [r247655] Matthew Nicholson <mnicholson@digium.com>
-
- * /, main/features.c: Merged revisions 247652 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r247652 | mnicholson | 2010-02-18 13:39:37 -0600 (Thu, 18 Feb
- 2010) | 13 lines Merged revisions 247651 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r247651 | mnicholson | 2010-02-18 13:38:09 -0600 (Thu, 18 Feb
- 2010) | 6 lines Copy the calling party's account code to the
- called party if they don't already have one. (closes issue
- #16331) Reported by: bluefox Tested by: mnicholson ........
- ................
-
-2010-02-18 16:56 +0000 [r247504-247510] Leif Madsen <lmadsen@digium.com>
-
- * README-SERIOUSLY.bestpractices.txt: Merged revisions 247509 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r247509 | lmadsen | 2010-02-18 11:54:43 -0500
- (Thu, 18 Feb 2010) | 9 lines Merged revisions 247508 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r247508 | lmadsen | 2010-02-18 11:53:44 -0500 (Thu, 18
- Feb 2010) | 1 line Add additional link to best practices document
- per jsmith. ........ ................
-
- * README-SERIOUSLY.bestpractices.txt (added): Merged revisions
- 247503 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r247503 | lmadsen | 2010-02-18 11:41:04 -0500 (Thu, 18 Feb 2010)
- | 18 lines Merged revisions 247502 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r247502 | lmadsen | 2010-02-18 11:38:17 -0500 (Thu, 18 Feb 2010)
- | 10 lines Add best practices documentation. (issue #16808)
- Reported by: lmadsen (issue #16810) Reported by: Nick_Lewis
- Tested by: lmadsen Review:
- https://reviewboard.asterisk.org/r/507/ ........ ................
-
-2010-02-18 04:20 +0000 [r247424] Russell Bryant <russell@digium.com>
-
- * Makefile, /, sounds/Makefile: Merged revisions 247423 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r247423 | russell | 2010-02-17 22:20:11 -0600
- (Wed, 17 Feb 2010) | 17 lines Merged revisions 247422 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r247422 | russell | 2010-02-17 22:19:01 -0600 (Wed, 17 Feb 2010)
- | 10 lines Tweak argument handling for wget in the sounds
- Makefile. 1) Fix the check to see if we are using wget to not be
- full of fail. The configure script populates this variable with
- the absolute path to wget if it is found, so it didn't work. 2)
- Allow some extra arguments to be passed in for wget. This is just
- a simple change to allow our Bamboo build script to tell wget to
- be quiet and not fill up our logs with download status output.
- ........ ................
-
-2010-02-17 21:35 +0000 [r246986-247338] Mark Michelson <mmichelson@digium.com>
-
- * /, main/utils.c, include/asterisk/strings.h: Merged revisions
- 247335 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r247335 |
- mmichelson | 2010-02-17 15:22:40 -0600 (Wed, 17 Feb 2010) | 20
- lines Fix two problems in ast_str functions found while writing a
- unit test. 1. The documentation for ast_str_set and
- ast_str_append state that the max_len parameter may be -1 in
- order to limit the size of the ast_str to its current allocated
- size. The problem was that the max_len parameter in all cases was
- a size_t, which is unsigned. Thus a -1 was interpreted as
- UINT_MAX instead of -1. Changing the max_len parameter to be
- ssize_t fixed this issue. 2. Once issue 1 was fixed, there was an
- off-by-one error in the case where we attempted to write a string
- larger than the current allotted size to a string when -1 was
- passed as the max_len parameter. When trying to write more than
- the allotted size, the ast_str's __AST_STR_USED was set to 1
- higher than it should have been. Thanks to Tilghman for quickly
- spotting the offending line of code. Oh, and the unit test that I
- referenced in the top line of this commit will be added to
- reviewboard shortly. Sit tight... ........
-
- * /, apps/app_queue.c: Merged revisions 247169 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r247169 | mmichelson | 2010-02-17 10:24:54 -0600 (Wed, 17 Feb
- 2010) | 9 lines Merged revisions 247168 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r247168 | mmichelson | 2010-02-17 10:24:17 -0600 (Wed, 17 Feb
- 2010) | 3 lines Make sure that when autofill is disabled that
- callers not in the front of the queue cannot place calls.
- ........ ................
-
- * /, include/asterisk/strings.h: Merged revisions 246985 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r246985 | mmichelson | 2010-02-16 15:15:38 -0600 (Tue,
- 16 Feb 2010) | 3 lines Add some clarifying documentation to the
- ast_str_set and ast_str_append functions. ........
-
-2010-02-16 21:07 +0000 [r246903-246984] David Vossel <dvossel@digium.com>
-
- * main/tcptls.c, /: Merged revisions 246980 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r246980 |
- dvossel | 2010-02-16 14:54:48 -0600 (Tue, 16 Feb 2010) | 8 lines
- warning message if openssl support is missing while attempting
- tls connection (closes issue #16673) Reported by: michaesc
- Patches: tls_error_msg.diff uploaded by dvossel (license 671)
- ........
-
- * main/channel.c, /: Merged revisions 246899 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r246899 |
- dvossel | 2010-02-16 11:07:41 -0600 (Tue, 16 Feb 2010) | 16 lines
- fixes sample rate conversion issue with Monitor application When
- using ast_seekstream with the read/write streams of a monitor,
- the number of samples we are seeking must be of the same rate as
- the stream or the jump calculation will be incorrect. This patch
- adds logic to correctly convert the number of samples to jump to
- the sample rate the read/write stream is using. For example, if
- the call is G722 (16khz) and the read/write stream is recording a
- 8khz wav, seeking 320 samples of 16khz audio is not the same as
- seeking 320 samples of 8khz audio when performing the
- ast_seekstream on the stream. ABE-2044 ........
-
-2010-02-15 23:44 +0000 [r246711] Tilghman Lesher <tlesher@digium.com>
-
- * Makefile, /: Merged revisions 246710 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r246710 | tilghman | 2010-02-15 17:43:28 -0600 (Mon, 15 Feb 2010)
- | 12 lines Merged revisions 246709 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r246709 | tilghman | 2010-02-15 17:42:33 -0600 (Mon, 15 Feb 2010)
- | 5 lines Make the menuselect instructions correct by allowing
- 'make menuselect' to actually solve dependency problems.
- (Previously, it would fail out again with the same message about
- running 'make menuselect', which was NOT at all helpful.)
- ........ ................
-
-2010-02-12 23:35 +0000 [r246549] David Vossel <dvossel@digium.com>
-
- * main/channel.c, /: Merged revisions 246546 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r246546 | dvossel | 2010-02-12 17:32:33 -0600 (Fri, 12 Feb 2010)
- | 21 lines Merged revisions 246545 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r246545 | dvossel | 2010-02-12 17:30:17 -0600 (Fri, 12 Feb 2010)
- | 16 lines lock channel during datastore removal On channel
- destruction the channel's datastores are removed and destroyed.
- Since there are public API calls to find and remove datastores on
- a channel, a lock should be held whenever datastores are removed
- and destroyed. This resolves a crash caused by a race condition
- in app_chanspy.c. (closes issue #16678) Reported by:
- tim_ringenbach Patches: datastore_destroy_race.diff uploaded by
- tim ringenbach (license 540) Tested by: dvossel ........
- ................
-
-2010-02-12 18:57 +0000 [r246462] Jason Parker <jparker@digium.com>
-
- * main/channel.c: Fix some silly formatting that made my head hurt.
-
-2010-02-10 21:27 +0000 [r246201-246205] Tilghman Lesher <tlesher@digium.com>
-
- * /, funcs/func_strings.c: Merged revisions 246204 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r246204 | tilghman | 2010-02-10 15:24:10 -0600 (Wed, 10 Feb 2010)
- | 2 lines Fussy compiler on another machine... ........
-
- * /, funcs/func_strings.c: Merged revisions 246200 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r246200 | tilghman | 2010-02-10 15:19:35 -0600 (Wed, 10 Feb 2010)
- | 2 lines Fix weird issue with unit tests on optimized build -
- turned out to be a signing issue. ........
-
-2010-02-10 17:56 +0000 [r246122] David Vossel <dvossel@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 246116 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r246116 | dvossel | 2010-02-10 11:49:34 -0600 (Wed, 10 Feb 2010)
- | 14 lines Merged revisions 246115 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r246115 | dvossel | 2010-02-10 11:44:20 -0600 (Wed, 10 Feb 2010)
- | 8 lines fixes random deadlock in app_queue with use_weight
- during reload (closes issue #16677) Reported by: tim_ringenbach
- Patches: app_queue_use_weight_deadlock.diff uploaded by tim
- ringenbach (license 540) ........ ................
-
-2010-02-10 16:53 +0000 [r246071] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_local.c, /: Merged revisions 246070 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r246070 | jpeeler | 2010-02-10 10:47:37 -0600 (Wed, 10 Feb 2010)
- | 22 lines Change channel state on local channels for
- busy,answer,ring. Previously local channels channel state never
- changed. This became problematic when the state of the other side
- of the local channel was lost, for example during a masquerade.
- Changing the state of the local channel allows for the scenario
- to be detected when the channel state is set to ringing, but the
- peer isn't ringing. The specific problem scenario is described in
- 164201. Although this was noted on one of the issues, here is the
- tested dialplan verified to work: exten =>
- 9700,1,Dial(Local/*9700@default&Local/0009700@default) exten =>
- *9700,1,Set(GLOBAL(TESTCHAN)=${CHANNEL:0:${MATH(${LEN(${CHANNEL})}-1):0:2}}1)
- exten => *9700,n,wait(3) ;3 works, 1 did not exten =>
- *9700,n,Dial(SIP/5001) exten => 0009700,1,Wait(1) ;1 works, 3 did
- not exten =>
- 0009700,n,ChannelRedirect(${TESTCHAN},parkedcalls,701,1) (closes
- issue #14992) Reported by: davidw ........
-
-2010-02-10 15:38 +0000 [r245946-246023] Tilghman Lesher <tlesher@digium.com>
-
- * /, funcs/func_strings.c: Merged revisions 246022 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r246022 | tilghman | 2010-02-10 09:36:57 -0600 (Wed, 10 Feb 2010)
- | 2 lines Enable warnings on atypical conditions for the FILTER
- function (suggested by mmichelson on the -dev list). ........
-
- * configs/extensions.conf.sample, /, funcs/func_strings.c: Merged
- revisions 245945 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r245945 | tilghman | 2010-02-10 08:06:12 -0600 (Wed, 10 Feb 2010)
- | 9 lines Merged revisions 245944 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r245944 | tilghman | 2010-02-10 07:37:13 -0600 (Wed, 10 Feb 2010)
- | 2 lines Include examples of FILTER usage in extension patterns
- where a "." may be a risk. ........ ................
-
-2010-02-09 23:14 +0000 [r245796] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 245793 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r245793 | dvossel | 2010-02-09 17:07:17 -0600
- (Tue, 09 Feb 2010) | 18 lines Merged revisions 245792 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r245792 | dvossel | 2010-02-09 16:55:38 -0600 (Tue, 09 Feb 2010)
- | 12 lines Fixes iaxs and iaxsl size off by one issue. 2^15 =
- 32768 which is the maximum allowed iax2 callnumber. Creating the
- iaxs and iaxsl array of size 32768 means the maximum callnumber
- is actually out of bounds. This causes a nasty crash. (closes
- issue #15997) Reported by: exarv Patches: iax_fix.diff uploaded
- by dvossel (license 671) ........ ................
-
-2010-02-09 18:09 +0000 [r245730] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_fax.c: Merged revisions 245729 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r245729 |
- tilghman | 2010-02-09 12:06:30 -0600 (Tue, 09 Feb 2010) | 8 lines
- Ensure frames are only freed once. (closes issue #16361) Reported
- by: vlad Patches: 20100208__issue16361.diff.txt uploaded by
- tilghman (license 14) Tested by: kenny, bloodoff, misaksen
- ........
-
-2010-02-09 16:25 +0000 [r245681] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, apps/app_fax.c: Merged revisions 245680 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r245680 |
- kpfleming | 2010-02-09 10:24:52 -0600 (Tue, 09 Feb 2010) | 8
- lines Don't offer MMR or JBIG transcoding during T.38
- negotiation. After further discussion with Steve Underwood, we
- should not (yet) be offering to receive MMR or JBIG transcoded
- streams from T.38 endpoints. A future spandsp release will
- support those features, and then they can be enabled during
- negotiation ........
-
-2010-02-08 23:51 +0000 [r245627] Jason Parker <jparker@digium.com>
-
- * apps/app_voicemail.c: Stop playing the message number multiple
- times. Also remove some accidentally duplicated code, which may
- have been causing a memleak. This was caused by a bad merge.
- (closes issue #16579) Reported by: kue Patches: 0016525.patch
- uploaded by hokie21 (license 987)
-
-2010-02-08 22:46 +0000 [r245579] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/Makefile, channels/Makefile: Merged revisions 245578 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r245578 | tilghman | 2010-02-08 16:31:40 -0600 (Mon, 08
- Feb 2010) | 12 lines Actually use _ASTLDFLAGS in the main/ and
- channels/ Makefiles. They were previously passed correctly, but
- they simply weren't used. This caused issues with various
- platforms whose builds needed to pass special linker flags via
- the configure script. (closes issue #16596) Reported by:
- pprindeville Patches: asterisk-1.6-astldflags.patch uploaded by
- pprindeville (license 347) Tested by: tilghman ........
-
-2010-02-08 20:42 +0000 [r245498] Jason Parker <jparker@digium.com>
-
- * main/ast_expr2.fl, /, main/ast_expr2f.c: Merged revisions 245497
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r245497 | qwell | 2010-02-08 14:41:05 -0600
- (Mon, 08 Feb 2010) | 11 lines Merged revisions 245496 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r245496 | qwell | 2010-02-08 14:39:50 -0600 (Mon, 08 Feb 2010) |
- 4 lines Remove reference of documentation in source directory.
- People don't always build Asterisk from source (distro packages,
- anybody?). ........ ................
-
-2010-02-05 19:26 +0000 [r245093] Jeff Peeler <jpeeler@digium.com>
-
- * contrib/firmware (removed), /, LICENSE: Merged revisions 245090
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r245090 | jpeeler | 2010-02-05 13:26:22 -0600
- (Fri, 05 Feb 2010) | 11 lines Merged revisions 245044 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r245044 | kpfleming | 2010-02-05 12:32:29 -0600 (Fri, 05 Feb
- 2010) | 5 lines Remove contrib/firmware directory as it is empty
- Remove explicit license for IAXy firmware as it is no longer
- included in the tree ........ ................
-
-2010-02-05 17:10 +0000 [r244928] Sean Bright <sean@malleable.com>
-
- * main/asterisk.c, /: Merged revisions 244927 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r244927 | seanbright | 2010-02-05 12:05:32 -0500 (Fri, 05 Feb
- 2010) | 9 lines Merged revisions 244926 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r244926 | seanbright | 2010-02-05 12:03:35 -0500 (Fri, 05 Feb
- 2010) | 1 line Update main copyright date. ........
- ................
-
-2010-02-03 18:40 +0000 [r244506] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 244505 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r244505 | tilghman | 2010-02-03 12:34:29 -0600 (Wed, 03 Feb 2010)
- | 8 lines The chanvar= setting should inherit the entire list of
- variables, not just the first one. (closes issue #16359) Reported
- by: raarts Patches: dahdi-setvars.diff uploaded by raarts
- (license 937) Tested by: raarts ........
-
-2010-02-02 22:32 +0000 [r244447] David Vossel <dvossel@digium.com>
-
- * main/udptl.c, /, channels/chan_sip.c, include/asterisk/udptl.h:
- Merged revisions 244443 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r244443 |
- dvossel | 2010-02-02 16:27:23 -0600 (Tue, 02 Feb 2010) | 18 lines
- fixes crash during T.38 negotiation caused by invalid or missing
- FaxMaxDatagram field AST-2010-001 (closes issue #16634) Reported
- by: krn (closes issue #16724) Reported by: barthpbx (closes issue
- #16517) Reported by: bklang (closes issue #16485) Reported by:
- elsto ........
-
-2010-02-02 Leif Madsen <lmadsen@digium.com>
-
- * Release Asterisk 1.6.0.22
-
- * AST-2010-001: An attacker attempting to negotiate T.38 over SIP can
- remotely crash Asterisk by modifying the FaxMaxDatagram field of
- the SDP to contain either a negative or exceptionally large value.
- The same crash occurs when the FaxMaxDatagram field is omitted from
- the SDP as well.
-
-2010-01-14 Leif Madsen <lmadsen@digium.com>
-
- * Release Asterisk 1.6.0.21
-
-2010-01-08 Leif Madsen <lmadsen@digium.com>
-
- * Release Asterisk 1.6.0.21-rc1
-
-2010-01-07 21:17 +0000 [r238494] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_oss.c, main/poll.c, channels/chan_usbradio.c,
- include/asterisk/utils.h, /, channels/chan_sip.c,
- channels/chan_alsa.c, channels/chan_console.c: Merged revisions
- 209400 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r209400 |
- kpfleming | 2009-07-28 08:49:46 -0500 (Tue, 28 Jul 2009) | 3
- lines Define side-effect-safe MIN and MAX macros and remove
- duplicate definitions from various files. (closes issue #16251)
- Reported by: asgaroth ........
-
-2010-01-07 20:22 +0000 [r238364-238441] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 238412 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r238412 | dvossel | 2010-01-07 14:15:27 -0600
- (Thu, 07 Jan 2010) | 16 lines Merged revisions 238411 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r238411 | dvossel | 2010-01-07 14:14:25 -0600 (Thu, 07 Jan 2010)
- | 10 lines fixes crash in "scheduled_destroy" in chan_iax A
- signed short was used to represent a callnumber. This is makes it
- possible to attempt to access the iaxs array with a negative
- index. (closes issue #16565) Reported by: jensvb ........
- ................
-
- * /, channels/chan_sip.c: Merged revisions 238405 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r238405 |
- dvossel | 2010-01-07 14:00:31 -0600 (Thu, 07 Jan 2010) | 8 lines
- Change in sip show channels display format allowing more digits
- for CID (closes issue #16459) Reported by: Rzadzins Patches:
- chan_sip_longer_cid.patch uploaded by Rzadzins (license 953)
- ........
-
- * /, apps/app_queue.c: Merged revisions 238361 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r238361 |
- dvossel | 2010-01-07 12:58:23 -0600 (Thu, 07 Jan 2010) | 8 lines
- cli 'queue show' formatting fix. queue name was truncated over 12
- characters (closes issue #16078) Reported by: RoadKill Patches:
- quequename_limit.patch uploaded by ppyy (license 906) Tested by:
- dvossel ........
-
-2010-01-06 21:48 +0000 [r238232] Tilghman Lesher <tlesher@digium.com>
-
- * /, funcs/func_cdr.c: Merged revisions 238231 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r238231 | tilghman | 2010-01-06 15:45:17 -0600 (Wed, 06 Jan 2010)
- | 11 lines Merged revisions 238230 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r238230 | tilghman | 2010-01-06 15:41:55 -0600 (Wed, 06 Jan 2010)
- | 4 lines Revise documentation on disposition values to the
- actual values used. (closes issue #16289) Reported by: wdoekes
- ........ ................
-
-2010-01-06 20:38 +0000 [r238135-238182] Jeff Peeler <jpeeler@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 238181 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r238181 |
- jpeeler | 2010-01-06 14:37:18 -0600 (Wed, 06 Jan 2010) | 8 lines
- Fix misreverting from 177158. (closes issue #15725) Reported by:
- shanermn Patches: v1-15725.patch uploaded by dimas (license 88)
- Tested by: shanermn ........
-
- * /, main/features.c: Merged revisions 238134 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r238134 |
- jpeeler | 2010-01-06 13:05:06 -0600 (Wed, 06 Jan 2010) | 10 lines
- Fix channel name comparison for bridge application. The channel
- name comparison was not comparing the whole string and therefore
- if one channel name was a substring of the other, the bridge
- would fail. (closes issue #16528) Reported by: telecos82 Patches:
- res_features_r236843.diff uploaded by telecos82 (license 687)
- ........
-
-2010-01-06 15:20 +0000 [r238011] Russell Bryant <russell@digium.com>
-
- * /, apps/app_mp3.c: Merged revisions 238010 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r238010 | russell | 2010-01-06 09:19:10 -0600 (Wed, 06 Jan 2010)
- | 14 lines Merged revisions 238009 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r238009 | russell | 2010-01-06 09:18:22 -0600 (Wed, 06 Jan 2010)
- | 7 lines Resolve a crash due to an ast_frame not being fully
- initialized. (closes issue #16531) Reported by: john8675309
- (closes SWP-615) ........ ................
-
-2010-01-06 06:51 +0000 [r237966] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_sip.c: Something clearly went wrong with a merge
- somewhere, because these are all duplicates (and therefore dead
- code).
-
-2010-01-05 23:10 +0000 [r237843-237923] David Vossel <dvossel@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 237920 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r237920 |
- dvossel | 2010-01-05 17:08:50 -0600 (Tue, 05 Jan 2010) | 16 lines
- fixes holdtime playback issue in app_queue When reporting hold
- time, the number of seconds should be mod 60. Otherwise audio
- playback could be something like "2 minutes 123 seconds" rather
- than "2 minutes 3 seconds". Also, the "minute" sound file is
- missing, so for the moment until that file can be created the
- "minutes" file is used instead. (closes issue #16168) Reported
- by: nickilo Patches: patch-unified-trunk-rev-222176 uploaded by
- nickilo (license ) Tested by: nickilo, wonderg ........
-
- * main/pbx.c, /: Merged revisions 237839 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r237839 |
- dvossel | 2010-01-05 13:29:47 -0600 (Tue, 05 Jan 2010) | 19 lines
- fixes subscriptions being lost after 'module reload' During a
- module reload if multiple extension configs are present, such as
- both extensions.conf and extensions.ael, watchers for one
- config's hints will be lost during the merging of the other
- config. This happens because hint watchers are only preserved for
- the current config being merged. The old context list is
- destroyed after the merging takes place, meaning any watchers
- that were not perserved will be removed. Now all hints are
- preserved during merging regardless of what config file is being
- merged. These hints are only restored if they are present within
- the new context list. (closes issue #16093) Reported by: jlaroff
- ........
-
-2010-01-05 17:19 +0000 [r237712] Russell Bryant <russell@digium.com>
-
- * /, main/utils.c: Merged revisions 237699 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r237699 | russell | 2010-01-05 11:16:01 -0600 (Tue, 05 Jan 2010)
- | 14 lines Merged revisions 237697 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r237697 | russell | 2010-01-05 11:13:28 -0600 (Tue, 05 Jan 2010)
- | 7 lines Change a NOTICE log message to DEBUG where it belongs.
- (closes issue #16479) Reported by: alexrecarey (closes SWP-577)
- ........ ................
-
-2010-01-04 21:51 +0000 [r237407-237575] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/say.c: Merged revisions 237574 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r237574 | tilghman | 2010-01-04 15:48:20 -0600 (Mon, 04 Jan 2010)
- | 13 lines Merged revisions 237573 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r237573 | tilghman | 2010-01-04 15:45:46 -0600 (Mon, 04 Jan 2010)
- | 6 lines Bounds checking for input string (closes issue #16407)
- Reported by: qwell Patches: 20100104__issue16407.diff.txt
- uploaded by tilghman (license 14) ........ ................
-
- * main/pbx.c, /: Merged revisions 237494 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r237494 | tilghman | 2010-01-04 14:59:01 -0600 (Mon, 04 Jan 2010)
- | 15 lines Merged revisions 237493 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r237493 | tilghman | 2010-01-04 14:57:35 -0600 (Mon, 04 Jan 2010)
- | 8 lines Regression in issue #15421 - Pattern matching (closes
- issue #16482) Reported by: wdoekes Patches:
- astsvn-16482-betterfix.diff uploaded by wdoekes (license 717)
- 20091223__issue16482.diff.txt uploaded by tilghman (license 14)
- Tested by: wdoekes, tilghman ........ ................
-
- * main/config.c, /: Merged revisions 237414 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r237414 |
- tilghman | 2010-01-04 13:03:20 -0600 (Mon, 04 Jan 2010) | 2 lines
- Oops, didn't compile (thanks, kpfleming) ........
-
- * main/config.c, /: Merged revisions 237410 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r237410 |
- tilghman | 2010-01-04 12:42:10 -0600 (Mon, 04 Jan 2010) | 7 lines
- Further reduce the encoded blank values back to blank in the
- realtime API. (closes issue #16533) Reported by: sergee Patches:
- 200100104__issue16533.diff.txt uploaded by tilghman (license 14)
- Tested by: sergee ........
-
- * main/pbx.c, /, res/res_agi.c, include/asterisk/channel.h: Merged
- revisions 237406 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r237406 | tilghman | 2010-01-04 12:28:28 -0600 (Mon, 04 Jan 2010)
- | 23 lines Merged revisions 237405 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r237405 | tilghman | 2010-01-04 12:19:00 -0600 (Mon, 04 Jan 2010)
- | 16 lines Add a flag to disable the Background behavior, for AGI
- users. This is in a section of code that relates to two other
- issues, namely issue #14011 and issue #14940), one of which was
- the behavior of Background when called with a context argument
- that matched the current context. This fix broke FreePBX,
- however, in a post-Dial situation. Needless to say, this is an
- extremely difficult collision of several different issues. While
- the use of an exception flag is ugly, fixing all of the issues
- linked is rather difficult (although if someone would like to
- propose a better solution, we're happy to entertain that
- suggestion). (closes issue #16434) Reported by: rickead2000
- Patches: 20091217__issue16434.diff.txt uploaded by tilghman
- (license 14) 20091222__issue16434__1.6.1.diff.txt uploaded by
- tilghman (license 14) Tested by: rickead2000 ........
- ................
-
-2010-01-04 16:26 +0000 [r237324] Jeff Peeler <jpeeler@digium.com>
-
- * /, res/res_agi.c: Merged revisions 237323 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r237323 |
- jpeeler | 2010-01-04 10:24:51 -0600 (Mon, 04 Jan 2010) | 5 lines
- Fix timeout for AGI command speech recognize. (closes issue
- #16297) Reported by: semond ........
-
-2010-01-04 16:21 +0000 [r237320] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_local.c, /: Merged revisions 237319 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r237319 | tilghman | 2010-01-04 10:20:03 -0600
- (Mon, 04 Jan 2010) | 10 lines Merged revisions 237318 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r237318 | tilghman | 2010-01-04 10:18:59 -0600 (Mon, 04 Jan 2010)
- | 3 lines It's also possible for the Local channel to directly
- execute an Application. Reviewboard:
- https://reviewboard.asterisk.org/r/452/ ........ ................
-
-2010-01-02 09:56 +0000 [r237137] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 237136 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r237136 | oej | 2010-01-02 10:54:22 +0100 (Lör, 02 Jan 2010) | 10
- lines Merged revisions 237135 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r237135 | oej | 2010-01-02 10:52:30 +0100 (Lör, 02 Jan 2010) | 2
- lines Release memory of the contact acl before unloading module
- ........ ................
-
-2009-12-30 22:00 +0000 [r236983] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_local.c, /: Merged revisions 236982 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r236982 | tilghman | 2009-12-30 15:59:18 -0600
- (Wed, 30 Dec 2009) | 16 lines Merged revisions 236981 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r236981 | tilghman | 2009-12-30 15:57:10 -0600 (Wed, 30 Dec 2009)
- | 9 lines Don't queue frames to channels that have no means to
- process them. (closes issue #15609) Reported by: aragon Patches:
- 20091230__issue16521__1.4__chan_local_only.diff.txt uploaded by
- tilghman (license 14) Tested by: aragon Review:
- https://reviewboard.asterisk.org/r/452/ ........ ................
-
-2009-12-30 21:12 +0000 [r236903] Jeff Peeler <jpeeler@digium.com>
-
- * /, utils/ael_main.c: Merged revisions 236902 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r236902 |
- jpeeler | 2009-12-30 15:09:28 -0600 (Wed, 30 Dec 2009) | 2 lines
- One more LOW_MEMORY compile fix. ........
-
-2009-12-30 17:55 +0000 [r236805-236849] Tilghman Lesher <tlesher@digium.com>
-
- * /, cdr/cdr_adaptive_odbc.c: Merged revisions 236847 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r236847 | tilghman | 2009-12-30 11:53:29 -0600 (Wed, 30 Dec 2009)
- | 4 lines When the field is blank, don't warn about the field
- being unable to be coerced, just skip the column. (closes
- http://lists.digium.com/pipermail/asterisk-dev/2009-December/041362.html)
- Reported by Nic Colledge on the -dev list, fixed by me. ........
-
- * /, channels/chan_sip.c: Merged revisions 236802 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r236802 |
- tilghman | 2009-12-29 17:05:45 -0600 (Tue, 29 Dec 2009) | 7 lines
- Shut down the SIP session timers more gracefully, in order to
- prevent a possible crash. (closes issue #16452) Reported by:
- corruptor Patches: 20091221__issue16452.diff.txt uploaded by
- tilghman (license 14) Tested by: corruptor ........
-
-2009-12-28 22:10 +0000 [r236714] Jason Parker <jparker@digium.com>
-
- * main/ast_expr2.c, /, main/ast_expr2.y: Merged revisions 236713
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r236713 | qwell | 2009-12-28 16:09:40 -0600 (Mon, 28 Dec
- 2009) | 8 lines Allow "REMAINDER" to function properly in
- expressions. (closes issue #16427) Reported by: wdoekes Patches:
- ast16-reminder-remainder.patch uploaded by wdoekes (license 717)
- Tested by: wdoekes ........
-
-2009-12-28 17:39 +0000 [r236668] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 236667 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r236667 | tilghman | 2009-12-28 11:37:46 -0600 (Mon, 28 Dec 2009)
- | 4 lines Use recommended option, not deprecated option. (closes
- issue #16515) Reported by: ManChicken ........
-
-2009-12-28 15:31 +0000 [r236511-236633] Sean Bright <sean@malleable.com>
-
- * include/asterisk/threadstorage.h, /, configure,
- include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
- 236613 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r236613 | seanbright | 2009-12-28 10:22:54 -0500 (Mon, 28 Dec
- 2009) | 14 lines Merged revisions 236585 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r236585 | seanbright | 2009-12-28 10:12:08 -0500 (Mon, 28 Dec
- 2009) | 7 lines Try a test compile to see if PTHREAD_ONCE_INIT
- requires extra braces. There was conditional code (based on build
- platform) to optioinally wrap PTHREAD_ONCE_INIT in braces that
- was removed since it is fixed in newer versions of
- Solaris/OpenSolaris, but I am still running into it on Solaris 10
- x86 so add a configure-time check for it. ........
- ................
-
- * /, apps/app_meetme.c: Merged revisions 236510 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r236510 | seanbright | 2009-12-28 07:44:58 -0500 (Mon, 28 Dec
- 2009) | 19 lines Merged revisions 236509 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r236509 | seanbright | 2009-12-28 07:43:36 -0500 (Mon, 28 Dec
- 2009) | 12 lines Avoid a crash with large numbers of MeetMe
- conferences. Similar to changes made to Queue(), when we have
- large numbers of conferences in meetme.conf (1000s) and we use
- alloca()/strdupa(), we can blow out the stack and crash, so
- instead just use a single fixed buffer. (closes issue #16509)
- Reported by: Kashif Raza Patches: 20091223_16509.patch uploaded
- by seanbright (license 71) Tested by: seanbright ........
- ................
-
-2009-12-27 18:22 +0000 [r236435] Tilghman Lesher <tlesher@digium.com>
-
- * contrib/init.d/rc.debian.asterisk, /: Merged revisions 236434 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r236434 | tilghman | 2009-12-27 12:20:53 -0600
- (Sun, 27 Dec 2009) | 9 lines Merged revisions 236433 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r236433 | tilghman | 2009-12-27 12:19:38 -0600 (Sun, 27
- Dec 2009) | 2 lines Turn on colors in the daemon, since there's
- many requests for it on Ubuntu. ........ ................
-
-2009-12-26 15:29 +0000 [r236359] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, sounds/Makefile: Merged revisions 236358 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r236358 | kpfleming | 2009-12-26 09:27:44 -0600 (Sat, 26 Dec
- 2009) | 9 lines Merged revisions 236357 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r236357 | kpfleming | 2009-12-26 09:26:17 -0600 (Sat, 26 Dec
- 2009) | 1 line update to latest releases with zero uid/gid
- ........ ................
-
-2009-12-23 18:26 +0000 [r236187-236301] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_stack.c, /: Merged revisions 236300 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r236300 |
- tilghman | 2009-12-23 12:25:27 -0600 (Wed, 23 Dec 2009) | 7 lines
- AGI may be invoked from outside the dialplan (closes issue
- #16510) Reported by: atis Patches: 20091223__issue16510.diff.txt
- uploaded by tilghman (license 14) Tested by: atis ........
-
- * /, res/res_agi.c: Merged revisions 236186 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r236186 | tilghman | 2009-12-22 21:07:48 -0600 (Tue, 22 Dec 2009)
- | 11 lines Merged revisions 236184 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r236184 | tilghman | 2009-12-22 20:55:24 -0600 (Tue, 22 Dec 2009)
- | 4 lines If EXEC only gets a single argument, don't crash when
- the second is used. (closes issue #16504) Reported by: bklang
- ........ ................
-
-2009-12-22 17:10 +0000 [r236066] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 236063 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r236063 | dvossel | 2009-12-22 11:00:08 -0600 (Tue, 22 Dec 2009)
- | 18 lines Merged revisions 236062 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r236062 | dvossel | 2009-12-22 10:58:19 -0600 (Tue, 22 Dec 2009)
- | 11 lines fixes issue with p->method incorrectly set to ACK It
- is possible for a second ACK to come in for a retransmitted
- message. If an ack does not match an unacked message in our
- queue, restore the previous p->method as this ACK is completely
- ignored. (closes issue #16295) Reported by: omolenkamp Patches:
- issue16295_v2.diff uploaded by dvossel (license 671) ........
- ................
-
-2009-12-21 19:54 +0000 [r235942] Jeff Peeler <jpeeler@digium.com>
-
- * res/res_monitor.c, /: Merged revisions 235941 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r235941 | jpeeler | 2009-12-21 13:54:20 -0600 (Mon, 21 Dec 2009)
- | 20 lines Merged revisions 235940 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r235940 | jpeeler | 2009-12-21 13:43:41 -0600 (Mon, 21 Dec 2009)
- | 13 lines Change Monitor to not assume file to write to does not
- contain pathing. 227944 changed the fname_base argument to always
- append the configured monitor path. This change was necessary to
- properly compare files for uniqueness. If a full path is given
- though, nothing needs to be appended and that is handled
- correctly now. (closes issue #16377) (closes issue #16376)
- Reported by: bcnit Patches: res_monitor.c-issue16376-1.patch
- uploaded by dant (license 670) ........ ................
-
-2009-12-21 17:11 +0000 [r235824] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/features.c: Merged revisions 235822 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r235822 | tilghman | 2009-12-21 11:00:46 -0600 (Mon, 21 Dec 2009)
- | 15 lines Merged revisions 235821 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r235821 | tilghman | 2009-12-21 10:45:03 -0600 (Mon, 21 Dec 2009)
- | 8 lines Send parking lot announcement to the channel which
- parked the call, not the park-ee. (closes issue #16234) Reported
- by: yeshuawatso Patches: 20091210__issue16234.diff.txt uploaded
- by tilghman (license 14) 20091221__issue16234__1.4.diff.txt
- uploaded by tilghman (license 14) Tested by: yeshuawatso ........
- ................
-
-2009-12-18 22:58 +0000 [r235662] Jeff Peeler <jpeeler@digium.com>
-
- * main/channel.c, /, include/asterisk/cdr.h: Merged revisions
- 235660 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r235660 | jpeeler | 2009-12-18 16:51:37 -0600 (Fri, 18 Dec 2009)
- | 55 lines Merged revisions 235635 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r235635 | jpeeler | 2009-12-18 16:29:51 -0600 (Fri, 18 Dec 2009)
- | 48 lines Correct CDR dispositions for BUSY/FAILED This patch is
- simple in that it reorders the disposition defines so that the
- fix for issue 12946 works properly (the default CDR disposition
- was changed to AST_CDR_NOANSWER). Also, the
- AST_CDR_FLAG_ORIGINATED flag was set in ast_call to ensure all
- CDR records are written. The side effects of CDR changes are
- scary, so I'm documenting the test cases performed to attempt to
- catch any regressions. The following tests were all performed
- using 1.4 rev 195881 vs head (235571) + patch: A calls B C calls
- B (busy) Hangup C Hangup A (Both SIP and features) A calls B A
- blind transfers to C Hangup C (Both SIP and features) A calls B A
- attended transfers to C Hangup C A calls B A attended transfers
- to C (SIP) C blind transfers to A (features) Hangup A All of the
- test scenario CDRs matched. The following tests were performed
- just with the patch to ensure proper operation (with
- unanswered=yes): exten =>s,1,Answer exten =>s,n,ResetCDR(w) exten
- =>s,n,ResetCDR(w) exten =>s,1,ResetCDR(w) exten =>s,n,ResetCDR(w)
- (closes issue #16180) Reported by: aatef Patches: bug16180.patch
- uploaded by jpeeler (license 325) ........ ................
-
-2009-12-18 22:42 +0000 [r235574-235657] Tilghman Lesher <tlesher@digium.com>
-
- * /, configure, configure.ac: Merged revisions 235656 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r235656 | tilghman | 2009-12-18 16:40:46 -0600
- (Fri, 18 Dec 2009) | 9 lines Merged revisions 235652 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r235652 | tilghman | 2009-12-18 16:39:30 -0600 (Fri, 18
- Dec 2009) | 2 lines Revise verbiage, per #asterisk-dev discussion
- ........ ................
-
- * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
- Merged revisions 235573 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r235573 | tilghman | 2009-12-18 15:19:43 -0600 (Fri, 18 Dec 2009)
- | 9 lines Merged revisions 235572 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r235572 | tilghman | 2009-12-18 15:18:16 -0600 (Fri, 18 Dec 2009)
- | 2 lines Point to the typical missing package, not the cryptic
- "termcap support". ........ ................
-
-2009-12-17 Leif Madsen <lmadsen@digium.com>
-
- * Release Asterisk 1.6.0.20
-
-2009-12-09 Leif Madsen <lmadsen@digium.com>
-
- * Release Asterisk 1.6.0.20-rc1
-
-2009-12-09 20:00 +0000 [r233841-233881] Russell Bryant <russell@digium.com>
-
- * main/loader.c, /: Fix breakage of the "module load <module>" CLI
- command.
-
- * main/loader.c, formats/format_ilbc.c, formats/format_vox.c,
- include/asterisk/module.h, formats/format_pcm.c,
- formats/format_h263.c, formats/format_g723.c,
- formats/format_h264.c, formats/format_jpeg.c,
- formats/format_g726.c, formats/format_gsm.c,
- formats/format_g729.c, main/editline/makelist.in,
- formats/format_sln.c, formats/format_wav.c,
- formats/format_ogg_vorbis.c, UPGRADE.txt, UPGRADE-1.6.txt,
- formats/format_wav_gsm.c, formats/format_sln16.c: Set a module
- load priority for format modules. A recent change to
- app_voicemail made it such that the module now assumes that all
- format modules are available while processing voicemail
- configuration. However, when autoloading modules, it was possible
- that app_voicemail was loaded before the format modules. Since
- format modules don't depend on anything, set a module load
- priority on them to ensure that they get loaded first when
- autoloading. This version of the patch is specific to Asterisk
- 1.4 and 1.6.0. These versions did not already support module load
- priority in the module API. This adds a trivial version of this
- which is just a module flag to include it in a pass before
- loading "everything". Thanks to mmichelson for the review!
- (closes issue #16412) Reported by: jiddings Tested by: russell
- Review: https://reviewboard.asterisk.org/r/445/
-
-2009-12-08 18:28 +0000 [r233729] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 233718 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r233718 | tilghman | 2009-12-08 12:22:44 -0600 (Tue, 08 Dec 2009)
- | 8 lines Find another ref leak and change how we manage module
- references. (closes issue #16388) Reported by: parisioa Patches:
- 20091208__issue16388.diff.txt uploaded by tilghman (license 14)
- Tested by: parisioa, tilghman Review:
- https://reviewboard.asterisk.org/r/442/ ........
-
-2009-12-07 23:57 +0000 [r233617] Atis Lezdins <atis@iq-labs.net>
-
- * contrib/valgrind.supp, /: Merged revisions 233577 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r233577 | atis | 2009-12-08 01:10:13 +0200 (Tue, 08 Dec 2009) | 8
- lines Fix compatibility with valgrind 3.3 and older. (noticed in
- issue #16388) Reported by: parisioa Patches: valgrind.supp
- uloaded by atis (license 242) Tested by: atis, parisioa ........
-
-2009-12-07 23:30 +0000 [r233475-233614] David Vossel <dvossel@digium.com>
-
- * /, main/utils.c: Merged revisions 233611 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r233611 |
- dvossel | 2009-12-07 17:28:51 -0600 (Mon, 07 Dec 2009) | 4 lines
- fixes incorrect logic in ast_uri_encode issue #16299 ........
-
- * /, channels/chan_sip.c: Merged revisions 233472 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r233472 | dvossel | 2009-12-07 12:08:46 -0600 (Mon, 07 Dec 2009)
- | 15 lines Merged revisions 233471 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r233471 | dvossel | 2009-12-07 12:07:38 -0600 (Mon, 07 Dec 2009)
- | 9 lines fixes missing Contact header angle brackets (closes
- issue #16298) Reported by: mgernoth Patches:
- reg_parse_issue_1.4.diff uploaded by dvossel (license 671) Tested
- by: dvossel ........ ................
-
-2009-12-07 16:16 +0000 [r233397] Matthew Nicholson <mnicholson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 233394 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r233394 |
- mnicholson | 2009-12-07 10:14:42 -0600 (Mon, 07 Dec 2009) | 8
- lines Do not reject SDP packets describing only non audio
- streams. (closes issue #16387) Reported by: zalex1953 Patches:
- media-level-c-fix1.diff uploaded by mnicholson (license 96)
- Tested by: mnicholson, zalex1953 ........
-
-2009-12-04 21:56 +0000 [r233284] David Vossel <dvossel@digium.com>
-
- * configs/iax.conf.sample, /: Merged revisions 233280 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r233280 | dvossel | 2009-12-04 15:54:44 -0600
- (Fri, 04 Dec 2009) | 14 lines Merged revisions 233279 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r233279 | dvossel | 2009-12-04 15:54:01 -0600 (Fri, 04 Dec 2009)
- | 7 lines clarify requirecalltoken option in iax.sample.conf
- (closes issue #16223) Reported by: bklang Patches:
- clarify-iax-requirecalltoken.patch uploaded by bklang (license
- 919) ........ ................
-
-2009-12-04 20:29 +0000 [r233236] Matthias Nick <mnick@digium.com>
-
- * /, pbx/pbx_config.c: Merged revisions 233093 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r233093 |
- mnick | 2009-12-04 11:15:47 -0600 (Fri, 04 Dec 2009) | 8 lines
- Parse global variables or expressions in hint extensions Parse
- global variables or expressions in hint extensions. Like: exten
- => 400,hint,DAHDI/i2/${GLOBAL(var)} (closes issue #16166)
- Reported by: rmudgett Tested by: mnick, rmudgett ........
-
-2009-12-04 20:11 +0000 [r233230] Russell Bryant <russell@digium.com>
-
- * /: unblock a rev.
-
-2009-12-04 17:39 +0000 [r233167] David Vossel <dvossel@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 233121 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r233121 | dvossel | 2009-12-04 11:22:31 -0600
- (Fri, 04 Dec 2009) | 12 lines Merged revisions 233116 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r233116 | dvossel | 2009-12-04 11:21:34 -0600 (Fri, 04 Dec 2009)
- | 6 lines document and rename strip_control() in app_voicemail
- (closes issue #16291) Reported by: wdoekes ........
- ................
-
-2009-12-04 17:20 +0000 [r233112] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /: Merged revisions 233100 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r233100 | russell | 2009-12-04 11:18:22 -0600 (Fri, 04 Dec 2009)
- | 14 lines Merged revisions 233092 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r233092 | russell | 2009-12-04 11:12:47 -0600 (Fri, 04 Dec 2009)
- | 7 lines Only do frame payload check for HOLD frames. This code
- was added for helping to debug the source of invalid HOLD frames.
- However, a side effect of this is that it will incorrectly report
- errors for frames that have an integer payload. Make the check
- for this block specific to the HOLD frame case. ........
- ................
-
-2009-12-04 15:46 +0000 [r233047] Matthias Nick <mnick@digium.com>
-
- * main/dsp.c, /: Merged revisions 233046 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r233046 | mnick | 2009-12-04 09:38:33 -0600 (Fri, 04 Dec 2009) |
- 17 lines Merged revisions 233014 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r233014 | mnick | 2009-12-04 09:17:03 -0600 (Fri, 04 Dec 2009) |
- 11 lines Warning message gets displayed only once Added
- additional field 'int display_inband_dtmf_warning', which when
- set to '1' displays the warning ('Inband DTMF is not supported on
- codec %s. Use RFC2833'), and when set to '0' doesn't display the
- warning. Otherwise you would get hundreds of warnings every
- second. (closes issue #15769) Reported by: falves11 Patches:
- patch_15769_14.txt uploaded by mnick (license 874) Tested by:
- mnick, falves11 ........ ................
-
-2009-12-03 21:03 +0000 [r232864] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 232854 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r232854 | tilghman | 2009-12-03 14:47:07 -0600
- (Thu, 03 Dec 2009) | 15 lines Merged revisions 232820 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r232820 | tilghman | 2009-12-03 14:10:19 -0600 (Thu, 03 Dec 2009)
- | 8 lines Deprecate "cz" in favor of "cs". Also, change the use
- of language codes so that language registers as a prefix, rather
- than an exact match. (closes issue #16272) Reported by: patrol-cz
- Patches: 20091203__issue16272.diff.txt uploaded by tilghman
- (license 14) ........ ................
-
-2009-12-03 14:47 +0000 [r232811] David Ruggles <thedavidfactor@gmail.com>
-
- * apps/app_externalivr.c: Merged revisions 232587 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r232587 |
- diruggles | 2009-12-02 17:17:22 -0500 (Wed, 02 Dec 2009) | 12
- lines Prevent double closing of FDs by EIVR This caused a problem
- when asterisk was under heavy load and running both AGI and EIVR
- applications. EIVR would close an FD at which point it would be
- considered freed and be used by a new AGI instance the second
- close would then close the FD now in use by AGI. (closes issue
- #16305) Reported by: diLLec Tested by: thedavidfactor, diLLec
- Review: https://reviewboard.asterisk.org/r/436/ ........
-
-2009-12-03 00:32 +0000 [r232699] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 232660-232661 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r232660 | tilghman | 2009-12-02 18:08:55 -0600 (Wed, 02
- Dec 2009) | 19 lines Fix multiple issues with musiconhold, which
- led to classes not getting destroyed properly. * Classes are now
- tracked past removal from the core container, and module removal
- is actively prevented until all references are freed. * A hanging
- reference stored in the channel has been removed. This could have
- caused a mismatch and the music state not properly cleared, if
- two or more reloads occurred between MOH being stopped and MOH
- being restarted. * In certain circumstances, duplicate classes
- were possible. * A race existed at reload time between a process
- being killed and the thread responsible for reading from the
- related pipe respawning that process. * Several reference counts
- have also been corrected. At least one could have caused deleted
- classes to stick around forever, consuming resources. This
- originally manifested as MOH external processes that were not
- killed at reload time. (closes issue #16279, closes issue #16207)
- Reported by: parisioa, dcabot Patches:
- 20091202__issue16279__2.diff.txt uploaded by tilghman (license
- 14) Tested by: parisioa, tilghman ........ r232661 | tilghman |
- 2009-12-02 18:09:36 -0600 (Wed, 02 Dec 2009) | 2 lines Remove
- debugging line ........
-
-2009-12-02 22:03 +0000 [r232577-232583] Jeff Peeler <jpeeler@digium.com>
-
- * main/manager.c, /: Merged revisions 232582 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r232582 | jpeeler | 2009-12-02 16:02:43 -0600 (Wed, 02 Dec 2009)
- | 14 lines Merged revisions 232581 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r232581 | jpeeler | 2009-12-02 15:57:42 -0600 (Wed, 02 Dec 2009)
- | 7 lines Send ack (response/message) after receiving manager
- action userevent (closes issue #16264) Reported by: dimas
- Patches: event-ack.patch uploaded by dimas (license 88) ........
- ................
-
- * main/manager.c, /: Merged revisions 232576 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r232576 |
- jpeeler | 2009-12-02 15:32:50 -0600 (Wed, 02 Dec 2009) | 8 lines
- Make manager response to "Action: events" finish with empty line
- (closes issue #16275) Reported by: vnovy Patches: manager.c.diff
- uploaded by vnovy (license 922) ........
-
-2009-12-02 17:08 +0000 [r232357] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_amd.c: Merged revisions 232356 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r232356 | file | 2009-12-02 13:06:54 -0400 (Wed, 02 Dec 2009) |
- 12 lines Merged revisions 232355 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r232355 | file | 2009-12-02 13:04:52 -0400 (Wed, 02 Dec 2009) | 5
- lines Fix a bug where if you hung up very quickly after calling
- AMD it would overwrite the AMDSTATUS of HANGUP with TOOLONG.
- (closes issue #16239) Reported by: CGMChris ........
- ................
-
-2009-12-02 17:03 +0000 [r232354] David Vossel <dvossel@digium.com>
-
- * /, main/acl.c: Merged revisions 232351 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r232351 | dvossel | 2009-12-02 11:00:15 -0600 (Wed, 02 Dec 2009)
- | 12 lines Merged revisions 232350 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r232350 | dvossel | 2009-12-02 10:59:18 -0600 (Wed, 02 Dec 2009)
- | 6 lines ast_outaddrfor doesn't do htons() on port, looks odd in
- strace. (closes issue #16290) Reported by: wdoekes ........
- ................
-
-2009-12-02 16:41 +0000 [r232346] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 232345 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r232345 |
- file | 2009-12-02 12:40:14 -0400 (Wed, 02 Dec 2009) | 7 lines Add
- support for handling the 415 Unsupported media type response like
- we do for a 488 Not acceptable here response. (closes issue
- #16186) Reported by: atis Patches: sip_t38_response_415.patch
- uploaded by atis (license 242) ........
-
-2009-12-02 15:45 +0000 [r232272] David Vossel <dvossel@digium.com>
-
- * funcs/func_groupcount.c, /: Merged revisions 232269 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r232269 | dvossel | 2009-12-02 09:42:54 -0600
- (Wed, 02 Dec 2009) | 15 lines Merged revisions 232268 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r232268 | dvossel | 2009-12-02 09:41:36 -0600 (Wed, 02 Dec 2009)
- | 9 lines fixes segfault in func_groupcount closes issue #16337)
- Reported by: Parantido Patches: issue_16337.diff uploaded by
- dvossel (license 671) Tested by: Parantido, dvossel ........
- ................
-
-2009-12-02 00:49 +0000 [r232092] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 232091 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r232091 | jpeeler | 2009-12-01 18:45:18 -0600
- (Tue, 01 Dec 2009) | 17 lines Merged revisions 232090 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r232090 | jpeeler | 2009-12-01 18:42:58 -0600 (Tue, 01 Dec 2009)
- | 10 lines Do not modify the gain settings on data calls. (The
- digital flag actually represents a data call.) (closes issue
- #15972) Reported by: udosw Patches: transcap_digital_fix.diff.txt
- uploaded by alecdavis (license 585) Tested by: alecdavis ........
- ................
-
-2009-12-01 23:39 +0000 [r232009-232013] Russell Bryant <russell@digium.com>
-
- * /, funcs/func_lock.c: Merged revisions 232012 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r232012 |
- russell | 2009-12-01 17:38:34 -0600 (Tue, 01 Dec 2009) | 2 lines
- Fix a build error on FreeBSD. ........
-
- * /, main/file.c: Merged revisions 232008 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r232008 | russell | 2009-12-01 17:27:53 -0600 (Tue, 01 Dec 2009)
- | 9 lines Merged revisions 232007 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r232007 | russell | 2009-12-01 17:25:36 -0600 (Tue, 01 Dec 2009)
- | 2 lines Fix a warning pointed out by buildbot. ........
- ................
-
-2009-12-01 21:57 +0000 [r231928] Jeff Peeler <jpeeler@digium.com>
-
- * main/channel.c, /: Merged revisions 231927 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r231927 | jpeeler | 2009-12-01 15:54:21 -0600 (Tue, 01 Dec 2009)
- | 19 lines Merged revisions 231911 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r231911 | jpeeler | 2009-12-01 15:29:31 -0600 (Tue, 01 Dec 2009)
- | 12 lines Fix crash with invalid frame data The crash was
- happening as a result of a frame containing an invalid data
- pointer, but was set with data length of zero. The few times the
- issue was reproduced it _seemed_ that the frame was queued
- properly, that is the data pointer was set to NULL. I never could
- reproduce the crash so as a last resort the crash has been fixed,
- but a check in __ast_read has been added to give as much
- information about the source of problematic frames in the future.
- (closes issue #16058) Reported by: atis ........ ................
-
-2009-12-01 21:22 +0000 [r231879] David Vossel <dvossel@digium.com>
-
- * main/pbx.c, /: Merged revisions 231867 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r231867 | dvossel | 2009-12-01 15:20:19 -0600 (Tue, 01 Dec 2009)
- | 9 lines Merged revisions 231853 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r231853 | dvossel | 2009-12-01 15:14:31 -0600 (Tue, 01 Dec 2009)
- | 3 lines WaitExten m option with no parameters generates frame
- with zero datalen but non-null data ptr ........ ................
-
-2009-12-01 15:51 +0000 [r231744] Matthew Nicholson <mnicholson@digium.com>
-
- * /, main/file.c: Merged revisions 231741 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r231741 | mnicholson | 2009-12-01 09:47:36 -0600 (Tue, 01 Dec
- 2009) | 9 lines Merged revisions 231740 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r231740 | mnicholson | 2009-12-01 09:34:57 -0600 (Tue, 01 Dec
- 2009) | 2 lines Ignore unknown formats in ast_format_str_reduce()
- and return an error if no know formats are found. ........
- ................
-
-2009-11-30 21:52 +0000 [r231693] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/udptl.c, /, channels/chan_sip.c, include/asterisk/udptl.h:
- Merged revisions 231692 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r231692 |
- kpfleming | 2009-11-30 15:47:42 -0600 (Mon, 30 Nov 2009) | 22
- lines Another round of UDPTL stack fixes/improvements: 1) Allow
- users of UDPTL stack to associate a character-string tag with a
- UDPTL session, so that log/error/debug messages generated by the
- UDPTL stack can be 'connected' to the endpoint that caused them
- to be generated. 2) Improve comments (and process) of calculating
- the far end's maximum IFP size when redundancy mode is in use for
- error correction. 3) When an IFP larger than the calculated 'far
- max IFP' size is presented for writing, truncate it rather than
- putting in the buffer and allowing the buffer to overflow; this
- will cause the ends to retrain to a lower bit rate that produces
- IFPs of an appropriate size if possible, and if not possible, the
- FAX transfer will fail completely. In these cases, it is due to
- the one endpoint supplying a T38FaxMaxDatagram value that is
- improperly calculated and is too low to be of use; we have
- configuration options available to override this behavior. 4)
- Eliminate use of T38FaxMaxDatagram value in udptl.conf; it is no
- longer needed. ........
-
-2009-11-30 21:37 +0000 [r231691] Matthew Nicholson <mnicholson@digium.com>
-
- * apps/app_voicemail.c, include/asterisk/file.h, /, main/file.c,
- main/app.c: Merged revisions 231688 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r231688 | mnicholson | 2009-11-30 15:31:55 -0600 (Mon, 30 Nov
- 2009) | 15 lines Merged revisions 231614 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r231614 | mnicholson | 2009-11-30 15:11:44 -0600 (Mon, 30 Nov
- 2009) | 8 lines Remove duplicate entries from voicemail format
- lists. This prevents app_voicemail from entering an infinite loop
- when the same format is specified twice in the format list.
- (closes issue #15625) Reported by: Shagg63 Tested by: mnicholson
- Review: https://reviewboard.asterisk.org/r/429/ ........
- ................
-
-2009-11-30 20:45 +0000 [r231603] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 231602 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r231602 |
- file | 2009-11-30 16:44:30 -0400 (Mon, 30 Nov 2009) | 5 lines
- When receiving SDP that matches the version of the last one do
- not treat it as a fatal error. (closes issue #16238) Reported by:
- seandarcy ........
-
-2009-11-30 18:58 +0000 [r231517-231560] David Vossel <dvossel@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 231556 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r231556 |
- dvossel | 2009-11-30 12:55:07 -0600 (Mon, 30 Nov 2009) | 11 lines
- app_queue crashes randomly, often during call-transfers This
- patch adds a ref to the queue_ent object's parent call_queue in
- queue_exec() so the call_queue won't be destroyed while the the
- queue_ent still holds a pointer to it. (closes issue 0015686)
- Tested by: dvossel, aragon ........
-
- * main/rtp.c, /: Merged revisions 231491 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r231491 | dvossel | 2009-11-30 11:28:28 -0600 (Mon, 30 Nov 2009)
- | 17 lines Merged revisions 231441 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r231441 | dvossel | 2009-11-30 11:14:08 -0600 (Mon, 30 Nov 2009)
- | 11 lines fixes crash caused by RTP comfort noise payload
- greater than 24 bytes AST-2009-010 (closes issue #16242) Reported
- by: amorsen Patches: issue16242.diff uploaded by oej (license
- 306) Tested by: amorsen, oej, dvossel ........ ................
-
-2009-11-25 22:34 +0000 [r231300] Tilghman Lesher <tlesher@digium.com>
-
- * main/channel.c, /: Merged revisions 231299 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r231299 | tilghman | 2009-11-25 16:33:02 -0600 (Wed, 25 Nov 2009)
- | 9 lines Merged revisions 231298 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r231298 | tilghman | 2009-11-25 16:31:57 -0600 (Wed, 25 Nov 2009)
- | 2 lines After a frame duplication failure, unlock the channel
- before returning. ........ ................
-
-2009-11-25 15:48 +0000 [r231192] Matthew Nicholson <mnicholson@digium.com>
-
- * /, pbx/pbx_lua.c: Merged revisions 231189 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r231189 |
- mnicholson | 2009-11-25 09:42:48 -0600 (Wed, 25 Nov 2009) | 4
- lines Load pbx_lua with global symbols to allow linking with
- other lua libraries. Found by Maxim Litnitskiy. ........
-
-2009-11-24 18:53 +0000 [r231096] Jeff Peeler <jpeeler@digium.com>
-
- * /, main/features.c: Merged revisions 231095 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r231095 |
- jpeeler | 2009-11-24 12:50:36 -0600 (Tue, 24 Nov 2009) | 11 lines
- Fix erroneous hangup extension execution ast_spawn_extension
- behaves differently from 1.4 in that hangups and extensions that
- do not exist do not return an error, whereas in 1.6 it does. This
- is now taken into account so that the AST_FLAG_BRIDGE_HANGUP_RUN
- flag gets set properly. (closes issue #16106) Reported by:
- ajohnson Tested by: ajohnson ........
-
-2009-11-23 15:46 +0000 [r230882] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
- 230881 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r230881 |
- file | 2009-11-23 09:45:45 -0600 (Mon, 23 Nov 2009) | 7 lines
- Change fax detection in chan_sip so it behaves as one would
- expect. Internally the way T.38 is negotiated has changed and the
- option no longer reflects a behavior that is valid. It will now
- look for a CNG tone on received calls and if present send the
- call to the 'fax' extension. It is then up to the application or
- channel to request the switch over to T.38. ........
-
-2009-11-23 15:35 +0000 [r230782-230878] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 230877 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r230877 | kpfleming | 2009-11-23 09:34:16 -0600 (Mon, 23 Nov
- 2009) | 9 lines Merged revisions 230839 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r230839 | kpfleming | 2009-11-23 09:09:24 -0600 (Mon, 23 Nov
- 2009) | 1 line Correct fix for issue #16268... the reporter's
- original patch was very close to correct. ........
- ................
-
- * /, channels/chan_sip.c: Merged revisions 230773 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r230773 | kpfleming | 2009-11-23 08:15:48 -0600 (Mon, 23 Nov
- 2009) | 12 lines Merged revisions 230772 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r230772 | kpfleming | 2009-11-23 08:13:56 -0600 (Mon, 23 Nov
- 2009) | 5 lines Ensure that SDP parsing does not ignore the last
- line of the SDP. (closes issue #16268) Reported by: sgimeno
- ........ ................
-
-2009-11-20 22:37 +0000 [r230729] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 230726 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r230726 | dvossel | 2009-11-20 16:35:54 -0600 (Fri, 20 Nov 2009)
- | 7 lines fixes iax2 show cache locking error, thanks alecdavis!
- (closes issue #16094) Reported by: alecdavis Patches:
- bug16094.diff.txt uploaded by alecdavis (license 585) Tested by:
- alecdavis, dvossel ........
-
-2009-11-20 21:09 +0000 [r230631] Matthew Nicholson <mnicholson@digium.com>
-
- * /, main/features.c: Merged revisions 230628 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r230628 | mnicholson | 2009-11-20 15:01:10 -0600 (Fri, 20 Nov
- 2009) | 15 lines Merged revisions 230627 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r230627 | mnicholson | 2009-11-20 14:53:06 -0600 (Fri, 20 Nov
- 2009) | 8 lines Copy the peer CDR's userfield to the bridge CDR
- if it exists. This is necessary for the recordagentcalls option
- in chan_agent to store the recorded file name in the bridge CDR.
- (closes issue #14590) Reported by: msetim Patches:
- queue_agent_userfield.patch uploaded by Laureano (license 265)
- Tested by: Laureano, mnicholson ........ ................
-
-2009-11-20 17:37 +0000 [r230512-230587] David Vossel <dvossel@digium.com>
-
- * /, include/asterisk/audiohook.h, main/audiohook.c: Merged
- revisions 230583 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r230583 |
- dvossel | 2009-11-20 11:26:20 -0600 (Fri, 20 Nov 2009) | 6 lines
- audiohook signal trigger on every status change (issue #14618)
- Review: https://reviewboard.asterisk.org/r/434/ ........
-
- * apps/app_mixmonitor.c, /: Merged revisions 230509 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r230509 | dvossel | 2009-11-19 15:26:21 -0600
- (Thu, 19 Nov 2009) | 17 lines Merged revisions 230508 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r230508 | dvossel | 2009-11-19 15:22:46 -0600 (Thu, 19 Nov 2009)
- | 10 lines fixes MixMonitor thread not exiting when
- StopMixMonitor is used (closes issue #16152) Reported by: AlexMS
- Patches: stopmixmonitor_1.4.diff uploaded by dvossel (license
- 671) Tested by: dvossel, AlexMS Review:
- https://reviewboard.asterisk.org/r/424/ ........ ................
-
-2009-11-30 Leif Madsen <lmadsen@digium.com>
-
- * Release Asterisk 1.6.0.19
-
- * AST-2009-010
-
- * SDP parser regression fix (issue #16268, issue #16238)
-
-2009-11-18 Leif Madsen <lmadsen@digium.com>
-
- * Release Asterisk 1.6.0.18
-
-2009-11-13 Leif Madsen <lmadsen@digium.com>
-
- * Release Asterisk 1.6.0.18-rc3
-
-2009-11-13 15:56 +0000 [r229913] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 229912 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r229912 |
- file | 2009-11-13 09:56:16 -0600 (Fri, 13 Nov 2009) | 2 lines Fix
- T.38 negotiation regression introduced with the SDP parser
- changes. ........
-
-2009-11-12 16:49 +0000 [r229673] David Vossel <dvossel@digium.com>
-
- * funcs/func_audiohookinherit.c, /: Merged revisions 229670 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r229670 | dvossel | 2009-11-12 10:44:39 -0600
- (Thu, 12 Nov 2009) | 12 lines Merged revisions 229669 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r229669 | dvossel | 2009-11-12 10:41:49 -0600 (Thu, 12 Nov 2009)
- | 6 lines fixes merging error, datastore was being freed in the
- wrong function. (closes issue #16219) Reported by: aragon
- ........ ................
-
-2009-11-11 19:51 +0000 [r229475-229500] David Brooks <dbrooks@digium.com>
-
- * main/pbx.c: Merged revisions 229499 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r229499 | dbrooks | 2009-11-11 13:48:18 -0600 (Wed, 11 Nov 2009)
- | 15 lines Merged revisions 229498 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r229498 | dbrooks | 2009-11-11 13:46:19 -0600 (Wed, 11 Nov 2009)
- | 8 lines Solaris doesn't like NULL going to ast_log Solaris will
- crash if NULL is passed to ast_log. This simple patch simply uses
- S_OR to get around this. (closes issue #15392) Reported by:
- yrashk ........ ................
-
- * /, apps/app_softhangup.c: Merged revisions 229460 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r229460 | dbrooks | 2009-11-11 12:13:56 -0600 (Wed, 11 Nov 2009)
- | 7 lines Flags not initialized in app_softhangup.c, causing
- undefined behavior Trivial patch [kobaz] to initialize an
- ast_flags = {0} (closes issue #16129) Reported by: kobaz ........
-
-2009-11-10 22:17 +0000 [r229363] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c, /: Merged revisions 229361 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r229361 | tilghman | 2009-11-10 16:14:22 -0600 (Tue, 10 Nov 2009)
- | 19 lines Merged revisions 229360 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r229360 | tilghman | 2009-11-10 16:09:16 -0600 (Tue, 10 Nov 2009)
- | 12 lines If two pattern classes start with the same digit and
- have the same number of characters, they will compare equal. The
- example given in the issue report is that of [234] and [246],
- which have these characteristics, yet they are clearly not
- equivalent. The code still uses these two characteristics, yet
- when the two scores compare equal, an additional check will be
- done to compare all characters within the class to verify
- equality. (closes issue #15421) Reported by: jsmith Patches:
- 20091109__issue15421__2.diff.txt uploaded by tilghman (license
- 14) Tested by: jsmith, thedavidfactor ........ ................
-
-2009-11-10 22:03 +0000 [r229357] David Ruggles <thedavidfactor@gmail.com>
-
- * doc/externalivr.txt: Merged revisions 229356 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r229356 | diruggles | 2009-11-10 17:01:50 -0500 (Tue, 10 Nov
- 2009) | 16 lines Merged revisions 229355 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r229355 | diruggles | 2009-11-10 16:45:15 -0500 (Tue, 10 Nov
- 2009) | 9 lines Fix ExternalIVR Documentation Remove
- documentation for event that doesn't function (closes issue
- #16220) Reported by: thedavidfactor Patches:
- externalivr.txt.20091110.1622.patch uploaded by thedavidfactor
- (license 903) ........ ................
-
-2009-11-10 21:31 +0000 [r229352] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_stack.c, /: Merged revisions 229351 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r229351 |
- tilghman | 2009-11-10 15:22:50 -0600 (Tue, 10 Nov 2009) | 7 lines
- When GOSUB is invoked within an AGI, it may not exit correctly.
- (closes issue #16216) Reported by: atis Patches:
- 20091110__atis_work.diff.txt uploaded by tilghman (license 14)
- Tested by: atis ........
-
-2009-11-10 20:07 +0000 [r229283] Joshua Colp <jcolp@digium.com>
-
- * /, codecs/codec_g726.c: Merged revisions 229282 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r229282 | file | 2009-11-10 16:06:13 -0400 (Tue, 10 Nov 2009) |
- 15 lines Merged revisions 229281 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r229281 | file | 2009-11-10 16:03:14 -0400 (Tue, 10 Nov 2009) | 8
- lines Remove broken support for direct transcoding between G.726
- RFC3551 and G.726 AAL2. On some systems the translation core
- would actually consider g726aal2 -> g726 -> signed linear to be a
- quicker path then g726aal2 -> signed linear which exposed this
- problem. (closes issue #15504) Reported by: globalnetinc ........
- ................
-
-2009-11-10 17:54 +0000 [r229234] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 229168 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r229168 | dvossel | 2009-11-10 11:16:49 -0600
- (Tue, 10 Nov 2009) | 15 lines Merged revisions 229167 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r229167 | dvossel | 2009-11-10 11:15:57 -0600 (Tue, 10 Nov 2009)
- | 9 lines don't crash on log message in solaris AST-2009-006
- (closes issue #16206) Reported by: bklang Tested by: bklang
- ........ ................
-
-2009-11-10 17:37 +0000 [r229229] David Ruggles <thedavidfactor@gmail.com>
-
- * doc/externalivr.txt: Merged revisions 229228 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r229228 | diruggles | 2009-11-10 12:33:47 -0500 (Tue, 10 Nov
- 2009) | 18 lines Merged revisions 229191 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r229191 | diruggles | 2009-11-10 12:23:59 -0500 (Tue, 10 Nov
- 2009) | 11 lines Document ExternalIVR event tag collision
- ExternalIVR uses the D tag for two different event types. This
- documents that behavior and how to differentiate between the two
- cases. Also includes a minor spelling fix and clarification
- (closes issue #16211) Reported by: thedavidfactor Patches:
- externalivr.txt.20091109.1507.patch uploaded by thedavidfactor
- (license 903) ........ ................
-
-2009-11-10 15:41 +0000 [r229100] Matthew Nicholson <mnicholson@digium.com>
-
- * channels/chan_sip.c: Reverted revision 202006. (closes issue
- #16175) Reported by: paul-tg
-
-2009-11-10 11:19 +0000 [r229057] Gavin Henry <ghenry@suretecsystems.com>
-
- * contrib/scripts/asterisk.ldap-schema, /: Merged revisions 229050
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r229050 | ghenry | 2009-11-10 11:16:10 +0000 (Tue, 10
- Nov 2009) | 20 lines Schema file additions * Added
- AsteriskDialplan, AsteriskAccount and AsteriskMailbox
- objectClasses to allow standalone dialplan, account and mailbox
- entries (STRUCTURAL) * Added new Fields: - AstAccountLanguage,
- AstAccountTransport, AstAccountPromiscRedir, -
- AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
- - AstAccountVideoSupport, AstAccountIgnoreSDPVersion * Removed
- redundant IPaddr (there's already IPAddress) - Gives more
- configuration Flags for SIP-Users available (tested) - Allows to
- create Asterisk Attributes in defined Asterisk ObjectClasses
- without extensibleObject (which really should be the last
- resort); gives also additional possibilities for LDAP-filter
- (closes issue #15874) Reported by: Medozas Patches:
- asterisk.ldap-schema.patch uploaded by Medozas (license 41)
- Tested by: Medozas, suretec ........
-
-2009-11-09 Leif Madsen <lmadsen@digium.com>
-
- * Release Asterisk 1.6.0.18-rc2
-
-2009-11-09 15:39 +0000 [r228898] Leif Madsen <lmadsen@digium.com>
-
- * main/channel.c: Merged revisions 228897 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r228897 | lmadsen | 2009-11-09 09:38:38 -0600 (Mon, 09 Nov 2009)
- | 14 lines Merged revisions 228896 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r228896 | lmadsen | 2009-11-09 09:37:43 -0600 (Mon, 09 Nov 2009)
- | 6 lines Update WARNING message. Update a WARNING message to
- give a suggested fix when encountered. (closes issue #16198)
- Reported by: atis Tested by: atis ........ ................
-
-2009-11-09 14:58 +0000 [r228861] Matthew Nicholson <mnicholson@digium.com>
-
- * /, include/asterisk/lock.h: Merged revisions 228858 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r228858 | mnicholson | 2009-11-09 08:37:07 -0600
- (Mon, 09 Nov 2009) | 15 lines Merged revisions 228827 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r228827 | mnicholson | 2009-11-09 08:16:03 -0600 (Mon, 09 Nov
- 2009) | 8 lines Perform limited bounds checking when destroying
- ast_mutex_t structures to make sure we don't try to use negative
- indices. (closes issue #15588) Reported by: zerohalo Patches:
- 20090820__issue15588.diff.txt uploaded by tilghman (license 14)
- Tested by: zerohalo ........ ................
-
-2009-11-06 22:38 +0000 [r228696] David Vossel <dvossel@digium.com>
-
- * main/channel.c, /: Merged revisions 228693 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r228693 | dvossel | 2009-11-06 16:35:44 -0600 (Fri, 06 Nov 2009)
- | 16 lines Merged revisions 228692 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r228692 | dvossel | 2009-11-06 16:33:27 -0600 (Fri, 06 Nov 2009)
- | 9 lines fixes audiohook write crash occuring in chan_spy
- whisper mode. After writing to the audiohook list in ast_write(),
- frames were being freed incorrectly. Under certain conditions
- this resulted in a double free crash. (closes issue #16133)
- Reported by: wetwired ........ ................
-
-2009-11-06 20:42 +0000 [r228651] Matthew Nicholson <mnicholson@digium.com>
-
- * funcs/func_base64.c, /, main/utils.c: Merged revisions 228620 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r228620 | mnicholson | 2009-11-06 13:47:11 -0600
- (Fri, 06 Nov 2009) | 15 lines Merged revisions 228378 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r228378 | mnicholson | 2009-11-06 10:26:59 -0600 (Fri, 06 Nov
- 2009) | 8 lines Properly handle '=' while decoding base64
- messages and null terminate strings returned from BASE64_DECODE.
- (closes issue #15271) Reported by: chappell Patches:
- base64_fix.patch uploaded by chappell (license 8) Tested by:
- kobaz ........ ................
-
-2009-11-06 18:40 +0000 [r228549] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 228548 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r228548 | file | 2009-11-06 14:37:59 -0400 (Fri, 06 Nov 2009) |
- 11 lines Merged revisions 228547 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r228547 | file | 2009-11-06 14:32:58 -0400 (Fri, 06 Nov 2009) | 4
- lines Don't overwrite caller ID name on a trunk with the
- configured fullname when using users.conf (issue ABE-1989)
- ........ ................
-
-2009-11-06 Leif Madsen <lmadsen@digium.com>
-
- * Release Asterisk 1.6.0.18-rc1
-
-2009-11-06 17:53 +0000 [r228479-228500] Joshua Colp <jcolp@digium.com>
-
- * /, doc/tex/localchannel.tex: Merged revisions 228499 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r228499 | file | 2009-11-06 13:52:00 -0400 (Fri, 06 Nov 2009) | 2
- lines Fix the localchannel.tex file. ........
-
- * channels/chan_sip.c: Fix a logic flaw I introduced when I was
- testing stuff out.
-
-2009-11-06 17:10 +0000 [r228423] David Vossel <dvossel@digium.com>
-
- * /, codecs/codec_ilbc.c: Merged revisions 228420 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r228420 | dvossel | 2009-11-06 11:09:01 -0600 (Fri, 06 Nov 2009)
- | 19 lines Merged revisions 228418 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r228418 | dvossel | 2009-11-06 11:07:13 -0600 (Fri, 06 Nov 2009)
- | 13 lines fixes segfault in iLBC For reasons not yet known, it
- appears possible for an ast_frame to have a datalen greater than
- zero while the actual data is NULL during Packet Loss
- Concealment. Most codecs don't support PLC so this doesn't affect
- them. This patch catches the malformed frame and prevents the
- crash from occuring. Additional efforts to determine why it is
- possible for a frame to look like this are still being
- investigated. (issue #16979) ........ ................
-
-2009-11-06 16:56 +0000 [r228411-228415] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_sip.c: Fix a crash caused by freeing a dialog
- directly instead of using dialog_unref. (closes issue #16097)
- Reported by: steinwej Patches: no_RTP.diff uploaded by steinwej
- (license 841)
-
- * /, main/abstract_jb.c: Merged revisions 228410 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r228410 | file | 2009-11-06 12:42:23 -0400 (Fri, 06 Nov 2009) |
- 14 lines Merged revisions 228409 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r228409 | file | 2009-11-06 12:41:20 -0400 (Fri, 06 Nov 2009) | 7
- lines Fix a bug caused by a partially invalid frame (from the
- jitterbuffer) passing through the Asterisk core. (closes issue
- #15560) Reported by: jvandal (closes issue #15709) Reported by:
- covici ........ ................
-
-2009-11-06 15:44 +0000 [r228271-228342] David Vossel <dvossel@digium.com>
-
- * /, main/astfd.c: Merged revisions 228339 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r228339 | dvossel | 2009-11-06 09:42:46 -0600 (Fri, 06 Nov 2009)
- | 12 lines Merged revisions 228338 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r228338 | dvossel | 2009-11-06 09:41:41 -0600 (Fri, 06 Nov 2009)
- | 5 lines fixes crash in astfd.c (closes issue #15981) Reported
- by: slavon ........ ................
-
- * funcs/func_audiohookinherit.c, /: Merged revisions 228268 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r228268 | dvossel | 2009-11-06 09:04:24 -0600 (Fri, 06
- Nov 2009) | 9 lines fixes memory leak in func_audiohookinherit.c
- (closes issue #15394) Reported by: boroda Patches:
- bug15394_memoryleak_diff2.txt uploaded by dbrooks (license 790)
- Tested by: dbrooks, boroda ........
-
-2009-11-05 21:24 +0000 [r228190] Jeff Peeler <jpeeler@digium.com>
-
- * apps/app_chanspy.c, /: Merged revisions 228189 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r228189 |
- jpeeler | 2009-11-05 15:23:06 -0600 (Thu, 05 Nov 2009) | 11 lines
- Fix the fix for chanspy option o In 224178, I assumed the
- uploaded patch was correct as it had received positive feedback.
- The flags were being checked in the incorrect location. Upon
- testing the fix this time it was also found that the flags from
- the dialplan weren't being copied to the
- chanspy_translation_helper. (closes issue #16167) Reported by:
- marhbere ........
-
-2009-11-05 19:39 +0000 [r228146] David Brooks <dbrooks@digium.com>
-
- * channels/chan_misdn.c, /: Merged revisions 228145 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r228145 | dbrooks | 2009-11-05 13:34:50 -0600
- (Thu, 05 Nov 2009) | 16 lines Merged revisions 228078 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r228078 | dbrooks | 2009-11-05 12:59:41 -0600 (Thu, 05 Nov 2009)
- | 9 lines chan_misdn Asterisk 1.4.27-rc2 crash Crash related to
- chan_misdn connection. Patch submitted by gknispel_proformatique,
- tested by francesco_r. "I have many crash since i have upgraded
- to Asterisk 1.4.27-rc2. Attached a full bt." This patch zeros out
- an ast_frame. (closes issue #16041) Reported by: francesco_r
- ........ ................
-
-2009-11-05 19:17 +0000 [r228081] Jason Parker <jparker@digium.com>
-
- * channels/chan_vpb.cc, /: Merged revisions 228080 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r228080 | qwell | 2009-11-05 13:16:29 -0600
- (Thu, 05 Nov 2009) | 15 lines Merged revisions 228079 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r228079 | qwell | 2009-11-05 13:14:25 -0600 (Thu, 05 Nov 2009) |
- 8 lines Fix crash on VPB exception when no hardware is present.
- (closes issue #14970) Reported by: tzafrir Patches:
- vpb_exception.diff uploaded by tzafrir (license 46) Tested by:
- markwaters ........ ................
-
-2009-11-04 23:52 +0000 [r227946] Jeff Peeler <jpeeler@digium.com>
-
- * res/res_monitor.c, /: Merged revisions 227945 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r227945 | jpeeler | 2009-11-04 17:50:59 -0600 (Wed, 04 Nov 2009)
- | 21 lines Merged revisions 227944 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r227944 | jpeeler | 2009-11-04 17:47:08 -0600 (Wed, 04 Nov 2009)
- | 14 lines Fix incorrect filename comparsion after monitor file
- change The logic to detect if a requested file is indeed a
- different file from the current file was incorrect. The main
- issue being confusion of the use of filename_base which was
- previously set without pathing information and then compared to
- another full path. Robust file comparison logic has been added to
- properly check if two files are the same even if symlinks are
- used. (closes issue #15313) Reported by: caspy Patches:
- 20091103__issue15313__1.4.diff.txt uploaded by jpeeler (license
- 325) but mostly tilghman's work ........ ................
-
-2009-11-04 21:15 +0000 [r227763-227833] Matthew Nicholson <mnicholson@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 227829 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r227829 | mnicholson | 2009-11-04 15:03:33 -0600 (Wed, 04 Nov
- 2009) | 17 lines Merged revisions 227827 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r227827 | mnicholson | 2009-11-04 14:52:27 -0600 (Wed, 04 Nov
- 2009) | 10 lines This patch modifies the Dial application to
- monitor the calling channel for hangups while playing back
- announcements. (closes issue #16005) Reported by: falves11
- Patches: dial-announce-hangup-fix1.diff uploaded by mnicholson
- (license 96) Tested by: mnicholson, falves11 Review:
- https://reviewboard.asterisk.org/r/407/ ........ ................
-
- * channels/chan_sip.c: Modify the SDP parsing code to parse session
- and media level items separately. With the new code, media level
- proprieties should no longer be confused with session level
- proprieties. This change also reorganizes some of the SDP parsing
- code which should make it easier to manage in the future. (closes
- issue #14994) Reported by: frawd
-
-2009-11-04 19:27 +0000 [r227717-227743] Joshua Colp <jcolp@digium.com>
-
- * /, static-http/prototype.js: Merged revisions 227739 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r227739 | file | 2009-11-04 15:26:19 -0400 (Wed,
- 04 Nov 2009) | 12 lines Merged revisions 227735 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r227735 | file | 2009-11-04 15:25:37 -0400 (Wed, 04 Nov 2009) | 5
- lines Fix a security issue where it may be possible for someone
- to execute a cross-site AJAX request exploit. (AST-2009-009)
- ........ ................
-
- * /, channels/chan_sip.c: Merged revisions 227712 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r227712 | file | 2009-11-04 15:20:46 -0400 (Wed, 04 Nov 2009) |
- 12 lines Merged revisions 227700 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r227700 | file | 2009-11-04 15:17:39 -0400 (Wed, 04 Nov 2009) | 5
- lines Fix a security issue where sending a REGISTER with a
- differing username in the From URI and Authorization header would
- reveal whether it was valid or not. (AST-2009-008) ........
- ................
-
-2009-11-03 20:00 +0000 [r227373] Jason Parker <jparker@digium.com>
-
- * Makefile, /, main/Makefile: Merged revisions 227372 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r227372 | qwell | 2009-11-03 13:59:46 -0600 (Tue, 03 Nov 2009) |
- 9 lines Fix some build issues on Solaris. (closes issue #14517)
- (SWP-109) Reported by: asgaroth Patches: bug_14517.diff uploaded
- by snuffy (license 35) Tested by: asgaroth, snuffy, dougm, qwell
- ........
-
-2009-11-03 19:49 +0000 [r227362-227369] Leif Madsen <lmadsen@digium.com>
-
- * apps/app_controlplayback.c, /: Merged revisions 227368 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r227368 | lmadsen | 2009-11-03 13:48:53 -0600 (Tue, 03
- Nov 2009) | 8 lines Change warning message to debug message.
- app_controlplayback outputs a warning, when in fact it is normal.
- (closes issue #16071) Reported by: atis Patches:
- controlplayback_warning.patch uploaded by atis (license 242)
- ........
-
- * configs/extensions.conf.sample, /: Merged revisions 227361 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r227361 | lmadsen | 2009-11-03 13:25:18 -0600 (Tue, 03
- Nov 2009) | 11 lines Additional fixes to the
- extensions.conf.sample file. Update the extensions.conf.sample
- [stdexten] context so that we use the variable instead of
- requiring it to be passed explicitly. Also updated uses of the
- [stdexten] context throughout. (closes issue #15858) Reported by:
- pprindeville Patches: stdexten-context-update.txt uploaded by
- lmadsen (license 10) Tested by: pprindeville ........
-
-2009-11-03 18:05 +0000 [r227278] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_dahdi.c: Merged revisions 227275 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r227275 | rmudgett | 2009-11-03 11:55:47 -0600 (Tue, 03 Nov 2009)
- | 4 lines Make sure the outgoing flag is cleared if a new channel
- fails to get created for outgoing calls. This is the relevant
- portion of asterisk/trunk -r226648 ........
-
-2009-11-03 15:37 +0000 [r227168] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 227167 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r227167 | file | 2009-11-03 11:37:08 -0400 (Tue, 03 Nov 2009) |
- 12 lines Merged revisions 227166 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r227166 | file | 2009-11-03 11:36:16 -0400 (Tue, 03 Nov 2009) | 5
- lines Fix a bug where an RPID header could be generated with a
- blank username in the URI. (closes issue #15909) Reported by:
- kobaz ........ ................
-
-2009-11-03 15:24 +0000 [r227163] Leif Madsen <lmadsen@digium.com>
-
- * configs/extensions.conf.sample, /: Merged revisions 227162 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r227162 | lmadsen | 2009-11-03 09:19:47 -0600 (Tue, 03
- Nov 2009) | 7 lines Update extensions.conf.sample file to fix
- incorrect extensions. (closes issue #15857) Reported by:
- pprindeville Patches: stdexten.patch#2 uploaded by pprindeville
- (license 347) Tested by: pprindeville ........
-
-2009-11-03 11:21 +0000 [r227102] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 227091 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r227091 | oej | 2009-11-03 12:11:15 +0100 (Tis, 03 Nov 2009) | 15
- lines Merged revisions 227088 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r227088 | oej | 2009-11-03 11:29:59 +0100 (Tis, 03 Nov 2009) | 7
- lines Use proper response code when violating Contact ACL's.
- https://reviewboard.asterisk.org/r/415/ Thanks kpfleming for a
- quick review. (EDVX-003) ........ ................
-
-2009-11-02 21:05 +0000 [r226975-226976] David Brooks <dbrooks@digium.com>
-
- * channels/chan_sip.c: SIP channel name uniqueness SIP channel
- names were supposed to be unique by way of a name suffix derived
- from the pointer to the channel's private data. Uniqueness was
- preserved on 32-bit systems, but not on 64-bit systems. This
- patch, as suggested by kpfleming, replaces this suffix with a
- simple incremented unsigned int. (closes issue #15152) Reported
- by: palbrecht Review: https://reviewboard.asterisk.org/r/420/
-
- * /: SIP channel name uniqueness SIP channel names were supposed to
- be unique by way of a name suffix derived from the pointer to the
- channel's private data. Uniqueness was preserved on 32-bit
- systems, but not on 64-bit systems. This patch, as suggested by
- kpfleming, replaces this suffix with a simple incremented
- unsigned int. (closes issue #15152) Reported by: palbrecht
- Review: https://reviewboard.asterisk.org/r/420/
-
-2009-11-02 18:09 +0000 [r226891] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 226890 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r226890 | file | 2009-11-02 14:08:54 -0400 (Mon, 02 Nov 2009) |
- 18 lines Merged revisions 226889 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r226889 | file | 2009-11-02 14:08:11 -0400 (Mon, 02 Nov 2009) |
- 11 lines Fix a bug where the recorded privacy introduction file
- would not get removed if the caller hung up while the called
- party had not yet answered. This was fixed by introducing an
- argument to the 'n' option which, when enabled, removes the
- introduction file under all scenarios. This was done to preserve
- the behavior that has existed for quite some time. (closes issue
- #14674) Reported by: ulogic Patches: bug14674.patch uploaded by
- jpeeler (license 325) ........ ................
-
-2009-11-02 17:16 +0000 [r226813] Tilghman Lesher <tlesher@digium.com>
-
- * /, contrib/init.d/rc.redhat.asterisk: Merged revisions 226812 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r226812 | tilghman | 2009-11-02 11:15:31 -0600
- (Mon, 02 Nov 2009) | 15 lines Merged revisions 226811 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r226811 | tilghman | 2009-11-02 11:14:20 -0600 (Mon, 02 Nov 2009)
- | 8 lines Don't allow two separate instances of safe_asterisk
- when restarting from the init script. (closes issue #14562)
- Reported by: davidw Patches: Initially
- 20091022__issue14562.diff.txt uploaded by tilghman (license 14)
- Modified to 20091030__Issue14562_diff.txt uploaded by davidw
- (license 780) Tested by: davidw ........ ................
-
-2009-10-29 18:14 +0000 [r226533] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_local.c, /, doc/tex/localchannel.tex: Merged
- revisions 226532 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r226532 | file | 2009-10-29 15:13:42 -0300 (Thu, 29 Oct 2009) |
- 13 lines Merged revisions 226531 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r226531 | file | 2009-10-29 15:11:26 -0300 (Thu, 29 Oct 2009) | 6
- lines Add an option to enabling passing music on hold start and
- stop requests through instead of acting on them in chan_local.
- (closes issue #14709) Reported by: dimas ........
- ................
-
-2009-10-28 20:17 +0000 [r226381-226387] Leif Madsen <lmadsen@digium.com>
-
- * configs/sip.conf.sample: Merged revisions 226384 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r226384 | lmadsen | 2009-10-28 15:11:07 -0500
- (Wed, 28 Oct 2009) | 17 lines Merged revisions 226382 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r226382 | lmadsen | 2009-10-28 15:06:13 -0500 (Wed, 28 Oct 2009)
- | 9 lines Update documentation in sip.conf.sample. Update the
- documentation in sip.conf.sample in order to make it more clear
- that directmedia/canreinvite do not cause Asterisk to ignore
- reINVITEs. It is only used to stop Asterisk from generating a
- reINVITE, but does not stop it from accepting them if necessary.
- (closes issue #15644) Reported by: lmadsen ........
- ................
-
- * /, doc/tex/channelvariables.tex: Merged revisions 226378 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r226378 | lmadsen | 2009-10-28 14:50:00 -0500
- (Wed, 28 Oct 2009) | 15 lines Merged revisions 226377 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r226377 | lmadsen | 2009-10-28 14:48:29 -0500 (Wed, 28 Oct 2009)
- | 7 lines Update CALLINGSUBADDR channel variable documentation.
- (closes issue #15734) Reported by: alecdavis Patches:
- channelvariables.tex.diff.txt uploaded by alecdavis (license 585)
- Tested by: alecdavis ........ ................
-
-2009-10-28 18:05 +0000 [r226167-226306] Tilghman Lesher <tlesher@digium.com>
-
- * /, include/asterisk/linkedlists.h: Merged revisions 226305 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r226305 | tilghman | 2009-10-28 13:04:05 -0500
- (Wed, 28 Oct 2009) | 9 lines Merged revisions 226304 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r226304 | tilghman | 2009-10-28 13:02:25 -0500 (Wed, 28
- Oct 2009) | 2 lines Fix documentation (pointed out by
- TheDavidFactor on #-dev) ........ ................
-
- * main/manager.c, /: Merged revisions 226159 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r226159 | tilghman | 2009-10-27 15:22:07 -0500 (Tue, 27 Oct 2009)
- | 14 lines Merged revisions 226138 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r226138 | tilghman | 2009-10-27 15:16:49 -0500 (Tue, 27 Oct 2009)
- | 7 lines Manager output is not always NULL-terminated, so force
- a NULL at the end of the filestream. (closes issue #15495)
- Reported by: pdf Patches: 20090916__issue15495.diff.txt uploaded
- by tilghman (license 14) Tested by: pdf ........ ................
-
-2009-10-26 23:13 +0000 [r226019] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
-
- * /, configure, configure.ac: detect ARM Linux EABI OSARCH as
- linux-gnu instead of linux-gnueabi * Set OSARCH to linux-gnu even
- if host_os is linux-gnueabi * When checking if we are Linux,
- check OSARCH rather than host_os The newer ARM ABI ("EABI") shows
- the OS name 'linux-gnueabi' rather than 'linux-gnu' . This patch
- sets OSARCH to be 'linux-gnu' even in such a case. OSARCH is
- tested for the value of 'linux-gnu' in one or two places in the
- tree. This patch also fixes the check libcap to check for $OSARCH
- rather than $host_os . See also:
- http://wiki.debian.org/ArmEabiPort Merged revisions 225957 via
- svnmerge from http://svn.digium.com/svn/asterisk/branches/1.4
- Merged revisions 226018 via svnmerge from
- http://svn.digium.com/svn/asterisk/trunk
-
-2009-10-26 15:46 +0000 [r225869] Kevin P. Fleming <kpfleming@digium.com>
-
- * apps/app_fax.c: Backport audio handling loop fixes from trunk
- version of app_fax. This backport resolves some issues handling
- audio frames during FAX processing, and ensures that the FAX
- application doesn't accidentally get notified of a T.38
- switchover at the end of a successful FAX. (issue #16127)
-
-2009-10-23 14:05 +0000 [r225583] Kevin P. Fleming <kpfleming@digium.com>
-
- * Makefile, /: Merged revisions 225582 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r225582 | kpfleming | 2009-10-23 09:02:42 -0500 (Fri, 23 Oct
- 2009) | 17 lines Merged revisions 225581 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r225581 | kpfleming | 2009-10-23 09:00:01 -0500 (Fri, 23 Oct
- 2009) | 10 lines Don't force menuselect.makeopts to be rebuilt on
- every build. For some reason the menuselect.makeopts file was
- listed as PHONY in the Makefile, resulting in 'make' needing to
- rebuild it for every build. This then resulted in the embedded
- module rules being rebuilt on every build, which can be slow and
- is unnecessary. This patch fixes the problem by properly allowing
- 'make' to know when the menuselect.makeopts file needs to be
- rebuilt (defining the proper dependencies). ........
- ................
-
-2009-10-22 21:53 +0000 [r225486] Leif Madsen <lmadsen@digium.com>
-
- * doc/valgrind.txt, contrib/valgrind.supp (added): Merged revisions
- 225485 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r225485 | lmadsen | 2009-10-22 16:52:30 -0500 (Thu, 22 Oct 2009)
- | 19 lines Merged revisions 225484 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r225484 | lmadsen | 2009-10-22 16:51:52 -0500 (Thu, 22 Oct 2009)
- | 11 lines Clean valgrind output by suppressing false errors.
- Update valgrind.txt documentation and add valgrind.supp file in
- order to allow those who are creating valgrind output to have
- less false errors in the logfile. (closes issue #16007) Reported
- by: atis Patches: valgrind.txt.diff uploaded by atis (license
- 242) asterisk2.supp uploaded by atis (license 242) Tested by:
- atis, amorsen ........ ................
-
-2009-10-22 17:13 +0000 [r225361] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c, /, apps/app_meetme.c, include/asterisk/channel.h:
- Merged revisions 225360 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r225360 | tilghman | 2009-10-22 12:11:23 -0500 (Thu, 22 Oct 2009)
- | 11 lines Merged revisions 225105 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r225105 | tilghman | 2009-10-21 11:02:12 -0500 (Wed, 21 Oct 2009)
- | 4 lines Fix documentation for ast_softhangup() and correct the
- misuse thereof. (closes issue #16103) Reported by: majorbloodnok
- ........ ................
-
-2009-10-21 22:10 +0000 [r225310-225311] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 225307 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r225307 | dvossel | 2009-10-21 16:58:46 -0500
- (Wed, 21 Oct 2009) | 20 lines Merged revisions 225243 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r225243 | dvossel | 2009-10-21 15:58:08 -0500 (Wed, 21 Oct 2009)
- | 13 lines IAX2: VNAK loop caused by signaling frames with no
- destination call number It is possible for the PBX thread to
- queue up signaling frames before a destination call number is
- received. This can result in signaling frames being sent out with
- no destination call number. Since recent versions of Asterisk
- require accurate destination callnumbers for all Full Frames,
- this can cause a VNAK loop to occur. To resolve this no signaling
- frames are sent until a destination callnumber is received, and
- destination call numbers are now only required for iax_pvt
- matching when the frame is an ACK. Review:
- https://reviewboard.asterisk.org/r/413/ ........ ................
-
- * configs/iax.conf.sample, /, channels/chan_sip.c,
- configs/sip.conf.sample, channels/chan_iax2.c: Merged revisions
- 225033 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r225033 | dvossel | 2009-10-21 09:39:10 -0500 (Wed, 21 Oct 2009)
- | 27 lines Merged revisions 225032 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r225032 | dvossel | 2009-10-21 09:37:04 -0500 (Wed, 21 Oct 2009)
- | 20 lines IAX/SIP shrinkcallerid option The shrinking of caller
- id removes '(', ' ', ')', non-trailing '.', and '-' from the
- string. This means values such as 555.5555 and test-test result
- in 555555 and testtest. There are instances, such as Skype
- integration, where a specific value is passed via caller id that
- must be preserved unmodified. This patch makes the shrinking of
- caller id optional in chan_sip and chan_iax in order to support
- such cases. By default this option is on to preserve previous
- expected behavior. (closes issue #15940) Reported by: dimas
- Patches: v2-15940.patch uploaded by dimas (license 88)
- 15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671)
- Tested by: dvossel Review:
- https://reviewboard.asterisk.org/r/408/ ........ ................
-
-2009-10-21 03:15 +0000 [r224933] Russell Bryant <russell@digium.com>
-
- * include/asterisk/translate.h, main/dsp.c, main/frame.c, /,
- main/translate.c, include/asterisk/dsp.h, codecs/codec_dahdi.c,
- include/asterisk/frame.h: Merged revisions 224932 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r224932 | russell | 2009-10-20 22:09:04 -0500
- (Tue, 20 Oct 2009) | 12 lines Merged revisions 224931 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r224931 | russell | 2009-10-20 21:59:54 -0500 (Tue, 20 Oct 2009)
- | 5 lines Isolate frames returned from a DSP instance or codec
- translator. The reasoning for these changes are the same as what
- I wrote in the commit message for rev 222878. ........
- ................
-
-2009-10-20 22:10 +0000 [r224857] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/audiohook.c: Merged revisions 224856 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r224856 | tilghman | 2009-10-20 17:09:07 -0500 (Tue, 20 Oct 2009)
- | 12 lines Merged revisions 224855 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r224855 | tilghman | 2009-10-20 17:07:11 -0500 (Tue, 20 Oct 2009)
- | 5 lines Pay attention to the return value of the manipulate
- function. While this looks like an optimization, it prevents a
- crash from occurring when used with certain audiohook callbacks
- (diagnosed with SVN trunk, backported to 1.4 to keep the source
- consistent across versions). ........ ................
-
-2009-10-20 17:48 +0000 [r224775] Joshua Colp <jcolp@digium.com>
-
- * /, main/features.c: Merged revisions 224774 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r224774 | file | 2009-10-20 14:47:34 -0300 (Tue, 20 Oct 2009) |
- 12 lines Merged revisions 224773 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r224773 | file | 2009-10-20 14:46:37 -0300 (Tue, 20 Oct 2009) | 5
- lines Add support for relaying early media in the features
- attended transfer option. (closes issue #14828) Reported by:
- licedey ........ ................
-
-2009-10-19 23:50 +0000 [r224672] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/rtp.c, /: Merged revisions 224671 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r224671 | kpfleming | 2009-10-19 18:47:39 -0500 (Mon, 19 Oct
- 2009) | 14 lines Merged revisions 224670 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r224670 | kpfleming | 2009-10-19 18:44:07 -0500 (Mon, 19 Oct
- 2009) | 7 lines Correct timestamp calculations when RTP sample
- rates over 8kHz are used. While testing some endpoints that
- support 16kHz and 32kHz sample rates, some log messages were
- generated due to calc_rxstamp() computing timestamps in a way
- that produced odd results, so this patch sanitizes the result of
- the computations. ........ ................
-
-2009-10-19 19:50 +0000 [r224568] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 224567 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r224567 | file | 2009-10-19 16:49:09 -0300 (Mon, 19 Oct 2009) |
- 12 lines Merged revisions 224565 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r224565 | file | 2009-10-19 16:47:50 -0300 (Mon, 19 Oct 2009) | 5
- lines Do not attempt early media bridging (ie: direct RTP setup)
- if options are enabled that should prevent it. (closes issue
- #14763) Reported by: cupotka ........ ................
-
-2009-10-19 00:12 +0000 [r224449] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 224448 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r224448 | tilghman | 2009-10-18 19:05:56 -0500 (Sun, 18 Oct 2009)
- | 3 lines Allow ODBC storage to be queried with multiple
- mailboxes. This corrects an issue reported on the -users list.
- ........
-
-2009-10-17 02:03 +0000 [r224332-224337] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c: fix typo, sorry
-
- * channels/chan_dahdi.c, /: Merged revisions 224331 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r224331 | jpeeler | 2009-10-16 20:36:08 -0500
- (Fri, 16 Oct 2009) | 20 lines Merged revisions 224330 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r224330 | jpeeler | 2009-10-16 20:32:47 -0500 (Fri, 16 Oct 2009)
- | 13 lines Fix stale caller id data from being reported in AMI
- NewChannel event The problem here is that chan_dahdi is designed
- in such a way to set certain values in the dahdi_pvt only once.
- One of those such values is the configured caller id data in
- chan_dahdi.conf. For PRI, the configured caller id data could be
- overwritten during a call. Instead of saving the data and
- restoring, it was decided that for all non-analog channels it was
- simply best to not set the configured caller id in the first
- place and also clear it at the end of the call. (closes issue
- #15883) Reported by: jsmith ........ ................
-
-2009-10-16 20:48 +0000 [r224262] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 224261 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r224261 | rmudgett | 2009-10-16 15:40:57 -0500
- (Fri, 16 Oct 2009) | 25 lines Merged revisions 224260 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r224260 | rmudgett | 2009-10-16 15:25:23 -0500 (Fri, 16 Oct 2009)
- | 18 lines Never released PRI channels when using Busy() or
- Congestion() dialplan apps. When the Busy() or Congestion()
- application is used towards ISDN (an ISDN progress is sent), the
- responding ISDN Disconnect or Release may contain the ISDN cause
- user busy or one of the congestion causes. In chan_dahdi.c these
- causes will only set the needbusy or needcongestion flags and not
- activate the softhangup procedure. Unfortunately only the latter
- can interrupt the endless wait loop of Busy()/Congestion().
- Result: PRI channels staying in state busy for the rest of
- asterisk life or until the other end times out and forces the
- call to clear. (in issue 0014292) Reported by: tomaso Patches:
- disc_rel_userbusy.patch uploaded by tomaso (license 564) (This
- patch is unrelated to the issue.) ........ ................
-
-2009-10-15 15:57 +0000 [r224179] Jeff Peeler <jpeeler@digium.com>
-
- * apps/app_chanspy.c, /: Merged revisions 224178 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r224178 |
- jpeeler | 2009-10-15 10:57:14 -0500 (Thu, 15 Oct 2009) | 11 lines
- Readd removed ability to allow listening to one side of the call
- in app_chanspy (Option o) (closes issue #15675) Reported by:
- john8675309 Patches: issue15675patchtrunk.txt uploaded by dbrooks
- (license 790) Tested by: jgutierrez on users list:
- http://lists.digium.com/pipermail/asterisk-users/2009-October/239155.html
- ........
-
-2009-10-12 23:50 +0000 [r223833] Jeff Peeler <jpeeler@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 223832 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r223832 | jpeeler | 2009-10-12 18:48:09 -0500 (Mon, 12 Oct 2009)
- | 15 lines Merged revisions 223804 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r223804 | jpeeler | 2009-10-12 18:12:50 -0500 (Mon, 12 Oct 2009)
- | 8 lines Ensure ringing continues for branched calls after
- progress is received While waiting for an answer, don't send
- progress for branched calls for which ringing was sent. (closes
- issue #15028) Reported by: fnordian ........ ................
-
-2009-10-12 21:07 +0000 [r223759] David Vossel <dvossel@digium.com>
-
- * configs/iax.conf.sample, /: Merged revisions 223756 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r223756 | dvossel | 2009-10-12 15:58:27 -0500 (Mon, 12 Oct 2009)
- | 5 lines Clarifies trunkmaxsize, trunkfreq, and trunkmtu iax2
- options SWP-151 ........
-
-2009-10-12 14:28 +0000 [r223653] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, channels/chan_sip.c, apps/app_fax.c: Merged revisions 223652
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r223652 | kpfleming | 2009-10-12 09:25:29 -0500 (Mon, 12
- Oct 2009) | 13 lines Remove automatic switching from T.38 to
- voice mode in chan_sip. chan_sip has some code to automatically
- switch from T.38 mode to voice mode when a voice frame is written
- to the channel while it is in T.38 mode; this was intended to
- handle the situation when a FAX transmission has ended and the
- channel is not yet hung up, but is causing problems at the
- beginning of FAX sessions as well when there are still voice
- frames 'in flight' at the time the T.38 negotiation completes.
- This patch removes the automatic switchover, and changes app_fax
- to explicitly switch off T.38 mode when the FAX transmission
- process ends. (closes issue #16025) Reported by: jamicque
- ........
-
-2009-10-11 17:27 +0000 [r223488] Russell Bryant <russell@digium.com>
-
- * main/autoservice.c, /: Merged revisions 223487 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r223487 | russell | 2009-10-11 12:25:42 -0500 (Sun, 11 Oct 2009)
- | 17 lines Merged revisions 223485-223486 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r223485 | russell | 2009-10-11 12:22:52 -0500 (Sun, 11 Oct 2009)
- | 6 lines Don't use data outside of its scope. The purpose of
- this code was to have a hangup frame put on the list of deferred
- frames. However, the code that read the hangup frame was outside
- of the scope of where the hangup frame was declared. ........
- r223486 | russell | 2009-10-11 12:25:06 -0500 (Sun, 11 Oct 2009)
- | 2 lines Remove some unnecessary code. ........ ................
-
-2009-10-09 23:08 +0000 [r223404] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c, channels/chan_h323.c: Fix interpretation
- of PRIREDIRECTIONREASON set by chan_sip. This commit is the
- simplest way to solve a problem that has already been solved in
- trunk with the "COLP/CONP and Redirecting party information into
- Asterisk" commit. In trunk the redirection reason is translated
- into a generic redirect reason. I would have had to do the same
- fix except chan_sip never reads PRIREDIRECTREASON. So both
- chan_dahdi and chan_h323 have been modified to interpret the one
- different redirect reason of "no-answer" properly and set the
- ISDN reason code 2 of "no reply". (closes issue #15033) Reported
- by: steinwej
-
-2009-10-09 20:59 +0000 [r223331] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, apps/app_fax.c: Merged revisions 223330 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r223330 |
- kpfleming | 2009-10-09 15:58:44 -0500 (Fri, 09 Oct 2009) | 10
- lines Initiate T.38 switchover when acting as called party,
- regardless of FAX direction. SendFAX() and ReceiveFAX() can be
- given options to indicate whether they should act as the calling
- or called party; this mode should be used to decide whether to
- initiate a switchover to T.38, not the direction that the FAX
- transfer will take place. (closes issue #16039) Reported by:
- jamicque ........
-
-2009-10-09 18:36 +0000 [r223276] Matthew Nicholson <mnicholson@digium.com>
-
- * main/channel.c, /: Merged revisions 223273 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r223273 | mnicholson | 2009-10-09 13:34:08 -0500 (Fri, 09 Oct
- 2009) | 14 lines Merged revisions 223225 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r223225 | mnicholson | 2009-10-09 13:20:11 -0500 (Fri, 09 Oct
- 2009) | 8 lines Signal timeouts by returning AST_CONTROL_RINGING
- when originating calls. (closes issue #15104) Reported by:
- nblasgen Patches: manager-timeout1.diff uploaded by mnicholson
- (license 96) Tested by: nblasgen, mnicholson ........
- ................
-
-2009-10-09 18:20 +0000 [r223226] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 223215 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r223215 | mmichelson | 2009-10-09 13:17:34 -0500 (Fri, 09 Oct
- 2009) | 9 lines Recorded merge of revisions 223213 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r223213 | mmichelson | 2009-10-09 13:17:12 -0500 (Fri,
- 09 Oct 2009) | 3 lines Fix potential memory leak in app_dial.c
- ........ ................
-
-2009-10-09 17:57 +0000 [r223210] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 223206 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r223206 | dvossel | 2009-10-09 12:53:37 -0500 (Fri, 09 Oct 2009)
- | 16 lines Merged revisions 223205 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r223205 | dvossel | 2009-10-09 12:52:35 -0500 (Fri, 09 Oct 2009)
- | 10 lines fixes sip registration using authuser in user.conf
- (closes issue #14954) Reported by: tornblad Tested by:
- mmichelson, tornblad, dvossel ........ ................
-
-2009-10-09 17:27 +0000 [r223172] Matthew Nicholson <mnicholson@digium.com>
-
- * cdr/cdr_sqlite3_custom.c, /: Merged revisions 223136 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r223136 | mnicholson | 2009-10-09 12:14:38 -0500 (Fri, 09 Oct
- 2009) | 8 lines Don't close the sqlite database when reloading.
- Only close the database when unloading. (closes issue #15953)
- Reported by: frawd Patches: sqlite3_rev220097.diff uploaded by
- frawd (license 610) Tested by: frawd ........
-
-2009-10-09 17:11 +0000 [r223091-223135] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 223132 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r223132 |
- dvossel | 2009-10-09 11:54:02 -0500 (Fri, 09 Oct 2009) | 9 lines
- 'auth=' did not parse md5 secret correctly (closes issue #15949)
- Reported by: ebroad Patches: authparsefix.patch uploaded by
- ebroad (license 878) 15949_trunk.diff uploaded by dvossel
- (license 671) Tested by: ebroad ........
-
- * /, channels/chan_sip.c: Merged revisions 223088 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r223088 |
- dvossel | 2009-10-09 10:49:30 -0500 (Fri, 09 Oct 2009) | 14 lines
- p->peerauth is always empty in transmit_register() When using
- callbackextension or specifing the peer name in a registration
- string, the peer's specific auth settings set by the "auth="
- strings within the peer definition are not used by the
- registration. Thanks to ebroad for reporting the issue and
- providing the patch. (closes issue #15955) Reported by: ebroad
- Patches: regauthfix.patch uploaded by ebroad (license 878)
- ........
-
-2009-10-08 19:54 +0000 [r222881] Russell Bryant <russell@digium.com>
-
- * include/asterisk/file.h, main/frame.c, /, main/file.c,
- include/asterisk/frame.h: Merged revisions 222880 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r222880 | russell | 2009-10-08 14:52:03 -0500
- (Thu, 08 Oct 2009) | 51 lines Merged revisions 222878 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r222878 | russell | 2009-10-08 14:45:47 -0500 (Thu, 08 Oct 2009)
- | 44 lines Make filestream frame handling safer by isolating
- frames before returning them. This patch is related to a number
- of issues on the bug tracker that show crashes related to freeing
- frames that came from a filestream. A number of fixes have been
- made over time while trying to figure out these problems, but
- there re still people seeing the crash. (Note that some of these
- bug reports include information about other problems. I am
- specifically addressing the filestream frame crash here.) I'm
- still not clear on what the exact problem is. However, what is
- _very_ clear is that we have seen quite a few problems over time
- related to unexpected behavior when we try to use embedded frames
- as an optimization. In some cases, this optimization doesn't
- really provide much due to improvements made in other areas. In
- this case, the patch modifies filestream handling such that the
- embedded frame will not be returned. ast_frisolate() is used to
- ensure that we end up with a completely mallocd frame. In
- reality, though, we will not actually have to malloc every time.
- For filestreams, the frame will almost always be allocated and
- freed in the same thread. That means that the thread local frame
- cache will be used. So, going this route doesn't hurt. With this
- patch in place, some people have reported success in not seeing
- the crash anymore. (SWP-150) (AST-208) (ABE-1834) (issue #15609)
- Reported by: aragon Patches: filestream_frisolate-1.4.diff2.txt
- uploaded by russell (license 2) Tested by: aragon, russell
- (closes issue #15817) Reported by: zerohalo Tested by: zerohalo
- (closes issue #15845) Reported by: marhbere Review:
- https://reviewboard.asterisk.org/r/386/ ........ ................
-
-2009-10-08 19:42 +0000 [r222876] David Vossel <dvossel@digium.com>
-
- * main/netsock.c, /, include/asterisk/netsock.h: Merged revisions
- 222873 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r222873 |
- dvossel | 2009-10-08 14:35:30 -0500 (Thu, 08 Oct 2009) | 6 lines
- fixes an ast_netsock_list memory leak. ABE-1998 Review:
- https://reviewboard.asterisk.org/r/395/ ........
-
-2009-10-08 16:47 +0000 [r222693-222800] Richard Mudgett <rmudgett@digium.com>
-
- * channels/misdn_config.c, /: Merged revisions 222799 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r222799 | rmudgett | 2009-10-08 11:44:33 -0500
- (Thu, 08 Oct 2009) | 19 lines Merged revisions 222797 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r222797 | rmudgett | 2009-10-08 11:33:06 -0500 (Thu, 08 Oct 2009)
- | 12 lines Fix memory leak if chan_misdn config parameter is
- repeated. Memory leak when the same config option is set more
- than once in an misdn.conf section. Why must this be considered?
- Templates! Defining a template with default port options and
- later adding to or overriding some of them. Patches:
- memleak-misdn.patch JIRA ABE-1998 ........ ................
-
- * channels/chan_misdn.c, /: Merged revisions 222692 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r222692 | rmudgett | 2009-10-07 16:56:36 -0500
- (Wed, 07 Oct 2009) | 21 lines Merged revisions 222691 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r222691 | rmudgett | 2009-10-07 16:51:24 -0500 (Wed, 07 Oct 2009)
- | 14 lines chan_misdn.c:process_ast_dsp() memory leak misdn.conf:
- astdtmf must be set to "yes". With "no", buffer loss does not
- occur. The translated frame "f2" when passing through
- ast_dsp_process() is not freed whenever it is not used further in
- process_ast_dsp(). Then in the end it is never ever freed.
- Patches: translate.patch JIRA ABE-1993 ........ ................
-
-2009-10-07 18:35 +0000 [r222605] Sean Bright <sean@malleable.com>
-
- * main/pbx.c: Properly initialize ast_devstate_aggregate so we
- don't crash sporadically. This looks like it was just missed
- during a merge. (closes issue #15841) Reported by: amorsen
- Patches:
- ast_devstate_aggregate_init-in-ast_extension_state2.patch
- uploaded by amorsen (license 676) Tested by: amorsen (closes
- issue #15852) Reported by: amorsen Tested by: amorsen, farisraouf
-
-2009-10-07 17:47 +0000 [r222546] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 222543 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r222543 | dvossel | 2009-10-07 12:44:52 -0500 (Wed, 07 Oct 2009)
- | 14 lines Merged revisions 222542 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r222542 | dvossel | 2009-10-07 12:41:21 -0500 (Wed, 07 Oct 2009)
- | 8 lines crash on transfer handle_invite_replaces() attempts to
- uplock a pvt's owner channel without first verifing that it
- exists. (issue #16027) ........ ................
-
-2009-10-07 17:32 +0000 [r222541] Tilghman Lesher <tlesher@digium.com>
-
- * res/ael/pval.c: Small typo (thanks, jpeeler)
-
-2009-10-06 23:57 +0000 [r222302-222464] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 222463 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r222463 | jpeeler | 2009-10-06 18:56:01 -0500
- (Tue, 06 Oct 2009) | 14 lines Merged revisions 222462 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r222462 | jpeeler | 2009-10-06 18:51:19 -0500 (Tue, 06 Oct 2009)
- | 8 lines Add missing unlock(s) in dahdi_read (two cases in
- trunk) (closes issue #15683) Reported by: alecdavis ........
- ................
-
- * channels/chan_dahdi.c: Fix potential crash when entire span
- request is received. The variable index used in this scenario for
- accessing the dahdi_pvts was wrong and was most likely copied
- from the several other places it is used correctly. (closes issue
- #15998) Reported by: tsearle Patches: dahdi_reset_crash.patch
- uploaded by tsearle (license 373) Modified:
- branches/1.4/channels/chan_dahdi.c
-
- * channels/chan_dahdi.c, /: Merged revisions 222351 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r222351 | jpeeler | 2009-10-06 15:35:19 -0500 (Tue, 06 Oct 2009)
- | 9 lines Fix 222298 (crash during destruction of second channel
- when variable set with setvar). I mistakenly reasoned that setvar
- would be used on all channels. Since it can be set per channel,
- give each dahdi channel a copy of the variable. (related to
- #15899) ........
-
- * channels/chan_dahdi.c, /: Merged revisions 222298 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r222298 | jpeeler | 2009-10-06 14:24:59 -0500 (Tue, 06 Oct 2009)
- | 9 lines Fix crash during destruction of second channel when
- variable set with setvar. The setvar line in chan_dahdi.conf is
- shared among all the channels, so make sure to only free the
- resources only when the last channel is destroyed. (closes issue
- #15899) Reported by: tzafrir ........
-
-2009-10-06 19:19 +0000 [r222279] Tilghman Lesher <tlesher@digium.com>
-
- * res/ael/pval.c, /: Recorded merge of revisions 222273 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r222273 | tilghman | 2009-10-06 14:17:11 -0500 (Tue, 06
- Oct 2009) | 5 lines When we call a gosub routine, the variables
- should be scoped to avoid contaminating the caller. This affected
- the ~~EXTEN~~ hack, where a subroutine might have changed the
- value before it was used in the caller. Patch by myself, tested
- by ebroad on #asterisk ........
-
-2009-11-04 Leif Madsen <lmadsen@digium.com>
-
- * Release Asterisk 1.6.0.17
-
- * AST-2009-008 and AST-2009-009
-
-2009-10-06 Leif Madsen <lmadsen@digium.com>
-
- * Release Asterisk 1.6.0.16-rc2
-
-2009-10-06 01:33 +0000 [r222111-222185] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/astobj2.c, /, funcs/func_dialgroup.c,
- include/asterisk/astobj2.h, res/res_phoneprov.c,
- channels/chan_console.c, res/res_musiconhold.c, apps/app_queue.c,
- channels/chan_iax2.c: Merged revisions 222176 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r222176 | kpfleming | 2009-10-05 20:24:24 -0500 (Mon, 05 Oct
- 2009) | 27 lines Recorded merge of revisions 222152 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05
- Oct 2009) | 20 lines Fix ao2_iterator API to hold references to
- containers being iterated. See Mantis issue for details of what
- prompted this change. Additional notes: This patch changes the
- ao2_iterator API in two ways: F_AO2I_DONTLOCK has become an enum
- instead of a macro, with a name that fits our naming policy;
- also, it is now necessary to call ao2_iterator_destroy() on any
- iterator that has been created. Currently this only releases the
- reference to the container being iterated, but in the future this
- could also release other resources used by the iterator, if the
- iterator implementation changes to use additional resources.
- (closes issue #15987) Reported by: kpfleming Review:
- https://reviewboard.asterisk.org/r/383/ ........ ................
-
- * main/udptl.c, /, channels/chan_sip.c, configs/udptl.conf.sample,
- UPGRADE.txt, configs/sip.conf.sample: Recorded merge of revisions
- 222110 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r222110 |
- kpfleming | 2009-10-05 14:45:00 -0500 (Mon, 05 Oct 2009) | 25
- lines Allow non-compliant T.38 endpoints to be supportable via
- configuration option. Many T.38 endpoints incorrectly send the
- maximum IFP frame size they can accept as the T38FaxMaxDatagram
- value in their SDP, when in fact this value is supposed to be the
- maximum UDPTL payload size (datagram size) they can accept. If
- the value they supply is small enough (a commonly supplied value
- is '72'), T.38 UDPTL transmissions will likely fail completely
- because the UDPTL packets will not have enough room for a primary
- IFP frame and the redundancy used for error correction. If this
- occurs, the Asterisk UDPTL stack will emit log messages warning
- that data loss may occur, and that the value may need to be
- overridden. This patch extends the 't38pt_udptl' configuration
- option in sip.conf to allow the administrator to override the
- value supplied by the remote endpoint and supply a value that
- allows T.38 FAX transmissions to be successful with that
- endpoint. In addition, in any SIP call where the override takes
- effect, a debug message will be printed to that effect. This
- patch also removes the T38FaxMaxDatagram configuration option
- from udptl.conf.sample, since it has not actually had any effect
- for a number of releases. In addition, this patch cleans up the
- T.38 documentation in sip.conf.sample (which incorrectly
- documented that T.38 support was passthrough only). (issue
- #15586) Reported by: globalnetinc ........
-
-2009-10-02 17:37 +0000 [r222038] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 222030 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r222030 | dvossel | 2009-10-02 12:34:07 -0500
- (Fri, 02 Oct 2009) | 9 lines Merged revisions 222026 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r222026 | dvossel | 2009-10-02 12:32:13 -0500 (Fri, 02
- Oct 2009) | 3 lines Removes unnecessary unlock, clarifies a
- memcpy. ........ ................
-
-2009-10-02 17:00 +0000 [r221972] Tilghman Lesher <tlesher@digium.com>
-
- * main/astobj2.c, /: Merged revisions 221971 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r221971 | tilghman | 2009-10-02 11:59:57 -0500 (Fri, 02 Oct 2009)
- | 9 lines Merged revisions 221970 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r221970 | tilghman | 2009-10-02 11:58:03 -0500 (Fri, 02 Oct 2009)
- | 2 lines Ensure the result of the hash function is positive.
- Negative array offsets suck. ........ ................
-
-2009-10-02 13:04 +0000 [r221963] Sean Bright <sean@malleable.com>
-
- * funcs/func_strings.c: Revert XML docs that ended up in the 1.6.0
- and 1.6.1 branches during a merge.
-
-2009-10-02 03:05 +0000 [r221921] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/logger.c: Merged revisions 221920 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r221920 |
- tilghman | 2009-10-01 22:04:34 -0500 (Thu, 01 Oct 2009) | 4 lines
- Initialize a variable that we check immediately upon startup.
- (closes issue #15973) Reported by: atis ........
-
-2009-10-02 01:20 +0000 [r221853] Richard Mudgett <rmudgett@digium.com>
-
- * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h, /:
- Merged revisions 221844 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r221844 | rmudgett | 2009-10-01 20:09:31 -0500 (Thu, 01 Oct 2009)
- | 33 lines Merged revisions 221769 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r221769 | rmudgett | 2009-10-01 18:18:28 -0500 (Thu, 01 Oct 2009)
- | 26 lines Occasionally losing use of B channels in chan_misdn. I
- have not been able to reproduce the problem of losing channels.
- However, I have seen in the code a reentrancy problem that might
- give these symptoms. The reentrancy patch does several things: 1)
- Guards B channel and B channel structure allocation. 2) Makes the
- B channel structure find routines more precise in locating
- records. 3) Never leave a B channel allocated if we received
- cause 44. The last item may cause temporary outgoing call
- problems, but they should clear when the line becomes idle.
- (closes issue #15490) Reported by: slutec18 Patches:
- issue15490_channel_alloc_reentrancy.patch uploaded by rmudgett
- (license 664) Tested by: rmudgett, slutec18 (closes issue #15458)
- Reported by: FabienToune Patches:
- issue15458_channel_alloc_reentrancy.patch uploaded by rmudgett
- (license 664) Tested by: FabienToune, rmudgett, slutec18 ........
- ................
-
-2009-10-02 00:03 +0000 [r221778] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c, main/rtp.c, /, main/say.c: Merged revisions
- 221777 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r221777 | tilghman | 2009-10-01 18:59:15 -0500 (Thu, 01 Oct 2009)
- | 9 lines Merged revisions 221776 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r221776 | tilghman | 2009-10-01 18:53:12 -0500 (Thu, 01 Oct 2009)
- | 2 lines Fix a bunch of off-by-one errors ........
- ................
-
-2009-10-01 21:04 +0000 [r221745] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 221697 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r221697 |
- dvossel | 2009-10-01 14:33:33 -0500 (Thu, 01 Oct 2009) | 9 lines
- outbound tls connections were not defaulting to port 5061 (closes
- issue #15854) Reported by: dvossel Patches:
- sip_port_config_trunk.diff uploaded by dvossel (license 671)
- Tested by: dvossel ........
-
-2009-10-01 20:34 +0000 [r221742] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 221705 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r221705 |
- tilghman | 2009-10-01 15:09:46 -0500 (Thu, 01 Oct 2009) | 2 lines
- Revision 220906 (a merge from 1.4) was not merged correctly,
- causing a problem with non-dynamic peers. ........
-
-2009-10-01 20:19 +0000 [r221712] David Vossel <dvossel@digium.com>
-
- * channels/chan_sip.c: Fixes issue with non dynamic hosts not being
- set for peers
-
-2009-10-01 17:09 +0000 [r221662] Matthew Nicholson <mnicholson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 221554,221589 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r221554 | oej | 2009-10-01 02:00:04 -0500 (Thu,
- 01 Oct 2009) | 3 lines Simplify code for porturi, use TRUE/FALSE
- constructs when it's just TRUE or FALSE. ................ r221589
- | mnicholson | 2009-10-01 10:26:20 -0500 (Thu, 01 Oct 2009) | 9
- lines Merged revisions 221588 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r221588 | mnicholson | 2009-10-01 10:24:00 -0500 (Thu, 01 Oct
- 2009) | 2 lines Use unsigned ints for portinuri flags. ........
- ................
-
-2009-10-01 16:18 +0000 [r221598] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/udptl.c, /, configs/udptl.conf.sample, UPGRADE.txt: Merged
- revisions 221592 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r221592 |
- kpfleming | 2009-10-01 11:16:09 -0500 (Thu, 01 Oct 2009) | 12
- lines Remove ability to control T.38 FAX error correction from
- udptl.conf. chan_sip has had the ability to control T.38 FAX
- error correction mode on a per-peer (or global) basis for a
- couple of releases now, which is where it should have been all
- along. This patch removes the ability to configure it in
- udptl.conf, but issues a warning if the user tries to do, telling
- them to look at sip.conf.sample for how to configure it now. For
- any SIP peers that are T.38 enabled in sip.conf, there is already
- a default for FEC error correction even if the user does not
- specify any mode, so this change will not turn off error
- correction by default, it will have the same default value that
- has been in the udptl.conf sample file. ........
-
-2009-09-30 23:08 +0000 [r221486] Matthew Nicholson <mnicholson@digium.com>
-
- * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
- 221432 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r221432 | mnicholson | 2009-09-30 15:40:20 -0500 (Wed, 30 Sep
- 2009) | 17 lines Merged revisions 221360 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r221360 | mnicholson | 2009-09-30 14:36:06 -0500 (Wed, 30 Sep
- 2009) | 10 lines Fix SRV lookup and Request-URI generation in
- chan_sip. This patch adds a new field "portinuri" to the sip
- dialog struct and the sip peer struct. That field is used during
- RURI generation to determine if the port should be included in
- the RURI. It is also used in some places to determine if an SRV
- lookup should occur. (closes issue #14418) Reported by: klaus3000
- Tested by: klaus3000, mnicholson Review:
- https://reviewboard.asterisk.org/r/369/ ........ ................
-
-2009-09-30 20:02 +0000 [r221369] Matthias Nick <mnick@digium.com>
-
- * configs/cdr_custom.conf.sample, /, funcs/func_strings.c: Merged
- revisions 221368 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r221368 | mnick | 2009-09-30 14:42:36 -0500 (Wed, 30 Sep 2009) |
- 23 lines Merged revisions 221153,221157,221303 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r221153 | mnick | 2009-09-30 10:37:39 -0500 (Wed, 30 Sep 2009) |
- 2 lines check bounds - prevents for buffer overflow ........
- r221157 | mnick | 2009-09-30 10:41:46 -0500 (Wed, 30 Sep 2009) |
- 8 lines added a new dialplan function 'CSV_QUOTE' and changed the
- cdr_custom.sample.conf (closes issue #15471) Reported by: dkerr
- Patches: csv_quote_14.txt uploaded by mnick (license ) Tested by:
- mnick ........ r221303 | mnick | 2009-09-30 14:02:00 -0500 (Wed,
- 30 Sep 2009) | 2 lines changed the prototype definition of
- csv_quote ........ ................
-
-2009-09-30 18:50 +0000 [r221301] Terry Wilson <twilson@digium.com>
-
- * main/rtp.c, /, channels/chan_sip.c, include/asterisk/rtp.h,
- configs/sip.conf.sample: Merged revisions 221266 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r221266 | twilson | 2009-09-30 12:52:30 -0500
- (Wed, 30 Sep 2009) | 32 lines Merged revisions 221086 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009)
- | 25 lines Change the SSRC by default when our media stream
- changes Be default, change SSRC when doing an audio stream
- changes Asterisk doesn't honor marker bit when reinvited to
- already-bridged RTP streams,resulting in far-end stack discarding
- packets with "old" timestamps that areactually part of a new
- stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is
- a reinvite, unless the 'constantssrc' is set to true in sip.conf.
- The original issue reported to Digium support detailed the
- following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based
- Application Server Call comes in fromITSP, Asterisk dials the app
- server which sends a re-invite back toAsterisk--not to negotiate
- to send media directly to the ITSP, but to indicatethat it's
- changing the stream it's sending to Asterisk. The app
- servergenerates a new SSRC, sequence numbers, timestamps, and
- sets the marker bit on the new stream. Asterisk passes through
- the teimstamp of the new stream, butdoes not reset the SSRC,
- sequence numbers, or set the marker bit. When the timestamp on
- the new stream is older than the timestamp on the originalstream,
- the ITSP (which doesn't know there has been any change) discards
- the newframes because it thinks they are too old. This patch
- addresses this by changing the SSRC on a stream update unless
- constantssrc=true is set in sip.conf. Review:
- https://reviewboard.asterisk.org/r/374/ ........ ................
-
-2009-09-30 16:57 +0000 [r221202] Tilghman Lesher <tlesher@digium.com>
-
- * main/channel.c, /: Merged revisions 221201 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r221201 | tilghman | 2009-09-30 11:56:42 -0500 (Wed, 30 Sep 2009)
- | 14 lines Merged revisions 221200 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r221200 | tilghman | 2009-09-30 11:55:21 -0500 (Wed, 30 Sep 2009)
- | 7 lines Avoid a potential NULL dereference. (closes issue
- #15865) Reported by: kobaz Patches: 20090915__issue15865.diff.txt
- uploaded by tilghman (license 14) Tested by: kobaz ........
- ................
-
-2009-09-30 14:52 +0000 [r221087] Sean Bright <sean@malleable.com>
-
- * apps/app_voicemail.c, /: Merged revisions 221085 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r221085 | seanbright | 2009-09-30 10:47:58 -0400 (Wed, 30 Sep
- 2009) | 9 lines Clarify documentation for VoiceMailMain()'s a()
- option. We require box numbers, not names as the documentation
- implies. (issue #14740) Reported by: pj Patches:
- __20090729-app_voicemail-documentation.patch uploaded by lmadsen
- (license 10) Tested by: seanbright, lmadsen ........
-
-2009-09-30 04:34 +0000 [r220976-221045] Tilghman Lesher <tlesher@digium.com>
-
- * /, funcs/func_lock.c: Recorded merge of revisions 221044 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r221044 | tilghman | 2009-09-29 23:32:36 -0500 (Tue, 29
- Sep 2009) | 8 lines Allow locks to be inherited through a
- masquerade without causing starvation. (closes issue #14859)
- Reported by: atis Patches: 20090821__issue14859.diff.txt uploaded
- by tilghman (license 14) 20090925__issue14859__1.6.1.diff.txt
- uploaded by tilghman (license 14) Tested by: atis, tilghman
- ........
-
- * /, channels/chan_sip.c: Merged revisions 220906 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r220906 | tilghman | 2009-09-29 14:57:37 -0500 (Tue, 29 Sep 2009)
- | 16 lines Merged revisions 220873 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r220873 | tilghman | 2009-09-29 12:59:26 -0500 (Tue, 29 Sep 2009)
- | 9 lines Reduce CPU usage related to building a peer merely for
- devicestates. This fixes a 100% CPU problem in the SIP driver,
- found by profiling the driver while the problem was occurring.
- (closes issue #14309) Reported by: pkempgen Patches:
- 20090924__issue14309.diff.txt uploaded by tilghman (license 14)
- Tested by: pkempgen, vrban ........ ................
-
-2009-09-29 20:25 +0000 [r220940] Matthew Nicholson <mnicholson@digium.com>
-
- * apps/app_chanspy.c: Avoid a deadlock in chanspy, just in case the
- spyee is masqueraded and chanspy_ds_chan_fixup() is called with
- the channel locked. (closes issue #15965) Reported by: atis
- Patches: chanspy-deadlock-fix1.diff uploaded by mnicholson
- (license 96) Tested by: atis
-
-2009-09-29 17:04 +0000 [r220834] Jeff Peeler <jpeeler@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 220833 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r220833 | jpeeler | 2009-09-29 11:58:29 -0500 (Tue, 29 Sep 2009)
- | 12 lines Make deletion of temporary greetings work properly
- with IMAP_STORAGE When imapgreetings was set to yes, the message
- was being deleted but wasn't actually being expunged. When
- imapgreetings was set to no, the file based message was not being
- deleted at all. All good now! (closes issue #14949) Reported by:
- noahisaac Patches: vm_tempgreeting_removal.patch uploaded by
- noahisaac (license 748), modified by me ........
-
-2009-09-28 19:13 +0000 [r220723] Sean Bright <sean@malleable.com>
-
- * /, Makefile.rules: Merged revisions 220721 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r220721 | seanbright | 2009-09-28 15:11:20 -0400 (Mon, 28 Sep
- 2009) | 10 lines Merged revisions 220717 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r220717 | seanbright | 2009-09-28 15:09:25 -0400 (Mon, 28 Sep
- 2009) | 3 lines When selecting DONT_OPTIMIZE in menuselect,
- explicitly pass -O0 to the compiler so we override any default
- optimization levels for a particular install. ........
- ................
-
-2009-09-26 15:11 +0000 [r220587] Tilghman Lesher <tlesher@digium.com>
-
- * /, include/asterisk/aes.h: Merged revisions 220586 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r220586 | tilghman | 2009-09-26 10:10:28 -0500 (Sat, 26 Sep 2009)
- | 2 lines Allow AES to compile, when OpenSSL is not present.
- ........
-
-2009-09-24 20:42 +0000 [r220372] David Vossel <dvossel@digium.com>
-
- * main/tcptls.c, /: Merged revisions 220365 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r220365 |
- dvossel | 2009-09-24 15:37:20 -0500 (Thu, 24 Sep 2009) | 8 lines
- fixes tcptls_session memory leak caused by ref count error
- (closes issue #15939) Reported by: dvossel Review:
- https://reviewboard.asterisk.org/r/375/ ........
-
-2009-09-24 19:42 +0000 [r220290] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_playback.c, main/pbx.c, /, apps/app_disa.c: Merged
- revisions 220289 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r220289 | tilghman | 2009-09-24 14:41:02 -0500 (Thu, 24 Sep 2009)
- | 13 lines Merged revisions 220288 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r220288 | tilghman | 2009-09-24 14:39:41 -0500 (Thu, 24 Sep 2009)
- | 6 lines Implicitly sending a progress signal breaks some
- applications. Call Progress() in your dialplan if you explicitly
- want progress to be sent. (Reverts change 216430, closes issue
- #15957) Reported by: Pavel Troller on the Asterisk-Dev mailing
- list
- http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html
- ........ ................
-
-2009-09-24 18:22 +0000 [r220101-220219] Sean Bright <sean@malleable.com>
-
- * Makefile, /: Merged revisions 220217 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r220217 | seanbright | 2009-09-24 14:19:41 -0400 (Thu, 24 Sep
- 2009) | 9 lines Merged revisions 220213 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r220213 | seanbright | 2009-09-24 14:18:18 -0400 (Thu, 24 Sep
- 2009) | 1 line Resolve parallel build warnings. Reported by Klaus
- Darilion on the asterisk-dev mailing list. ........
- ................
-
- * Makefile, build_tools/mkpkgconfig, /: Merged revisions 220100 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r220100 | seanbright | 2009-09-24 10:44:08 -0400
- (Thu, 24 Sep 2009) | 9 lines Merged revisions 220099 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r220099 | seanbright | 2009-09-24 10:41:57 -0400 (Thu,
- 24 Sep 2009) | 2 lines Remove the remaining bashisms in the
- Makefile/mkpkgconfig ........ ................
-
-2009-09-24 08:37 +0000 [r220029] Michiel van Baak <michiel@vanbaak.info>
-
- * build_tools/mkpkgconfig, /: Merged revisions 220028 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r220028 | mvanbaak | 2009-09-24 10:36:18 +0200
- (Thu, 24 Sep 2009) | 14 lines Merged revisions 220027 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r220027 | mvanbaak | 2009-09-24 10:33:50 +0200 (Thu, 24 Sep 2009)
- | 7 lines mkpkgconfig does not need bash so make it use /bin/sh
- This fixes building on all systems that don't have bash at
- /bin/bash Reported by _ys on #asterisk-dev Tested by _ys on
- #asterisk-dev ........ ................
-
-2009-09-22 21:47 +0000 [r219819] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 219818 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r219818 | tilghman | 2009-09-22 16:43:22 -0500
- (Tue, 22 Sep 2009) | 17 lines Merged revisions 219816 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r219816 | tilghman | 2009-09-22 16:37:03 -0500 (Tue, 22 Sep 2009)
- | 10 lines When IMAP variables were changed during a reload,
- Voicemail did not use the new values. This change introduces a
- configuration version variable, which ensures that connections
- with the old values are not reused but are allowed to expire
- normally. (closes issue #15934) Reported by: viniciusfontes
- Patches: 20090922__issue15934.diff.txt uploaded by tilghman
- (license 14) Tested by: viniciusfontes ........ ................
-
-2009-09-21 17:03 +0000 [r219724] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 219721 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r219721 | dvossel | 2009-09-21 11:59:05 -0500
- (Mon, 21 Sep 2009) | 9 lines Merged revisions 219720 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r219720 | dvossel | 2009-09-21 11:55:53 -0500 (Mon, 21
- Sep 2009) | 3 lines Reverting merge 219520. This change was not
- necessary. ........ ................
-
-2009-09-20 18:20 +0000 [r219663] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/file.c: Merged revisions 219654 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r219654 | tilghman | 2009-09-20 12:55:49 -0500 (Sun, 20 Sep 2009)
- | 15 lines Merged revisions 219653 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r219653 | tilghman | 2009-09-20 12:52:05 -0500 (Sun, 20 Sep 2009)
- | 8 lines Really stop the stream, when ast_closestream() is
- called. (closes issue #15129) Reported by: bmh Patches:
- 20090918__issue15129.diff.txt uploaded by tilghman (license 14)
- Review: https://reviewboard.asterisk.org/r/372/ ........
- ................
-
-2009-09-19 03:06 +0000 [r219588] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 219587 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r219587 | russell | 2009-09-18 21:59:52 -0500
- (Fri, 18 Sep 2009) | 13 lines Merged revisions 219586 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r219586 | russell | 2009-09-18 21:51:13 -0500 (Fri, 18 Sep 2009)
- | 6 lines Make sure the iax_pvt exists before dereferencing it.
- This fixes the latest crash posted on issue 15609. (issue #15609)
- ........ ................
-
-2009-09-18 23:23 +0000 [r219454-219523] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 219520 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r219520 | dvossel | 2009-09-18 18:20:58 -0500
- (Fri, 18 Sep 2009) | 15 lines Merged revisions 219519 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r219519 | dvossel | 2009-09-18 18:19:50 -0500 (Fri, 18 Sep 2009)
- | 9 lines iax2 frame double free The iax frame's retrans sched id
- was written over right before iax2_frame_free was called. In
- iax2_frame_free that retrans id is used to delete the sched item.
- By writing over the retrans field before the sched item could be
- deleted, it was possible for a retransmit to occur on a freed
- frame. ........ ................
-
- * /, channels/chan_sip.c: Merged revisions 219451 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r219451 | dvossel | 2009-09-18 11:20:41 -0500 (Fri, 18 Sep 2009)
- | 20 lines Merged revisions 219450 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r219450 | dvossel | 2009-09-18 11:19:15 -0500 (Fri, 18 Sep 2009)
- | 14 lines via-header branches not updated correctly on INVITE
- INVITE requests must always contain a new unique branch id. When
- a new branch id is created for an INVITE, the dialog's
- invite_branch variable must be updated so CANCEL requests use the
- correct branch id. (closes issue #15262) Reported by: maniax
- Patches: asterisk-1.6.1.0-sip-branch.patch uploaded by tweety
- (license 608) invite_new_branch_trunk.diff uploaded by dvossel
- (license 671) Tested by: maniax, dvossel ........
- ................
-
-2009-09-18 13:57 +0000 [r219413] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 219412 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r219412 | tilghman | 2009-09-18 08:54:51 -0500 (Fri, 18 Sep 2009)
- | 6 lines Missing value setting line for maxsecs/maxmessage
- (closes issue #15696) Reported by: fhackenberger Patches:
- maxsecs.patch uploaded by fhackenberger (license 592) ........
-
-2009-09-17 22:36 +0000 [r219365] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 219324 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r219324 | mmichelson | 2009-09-17 17:22:01 -0500 (Thu, 17 Sep
- 2009) | 12 lines Merged revisions 219320 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r219320 | mmichelson | 2009-09-17 17:20:50 -0500 (Thu, 17 Sep
- 2009) | 6 lines Send a 100 Trying response when we detect a
- spiral. This was problematic during spiral tests at SIPit...
- along with some other things as well. ........ ................
-
-2009-09-17 22:01 +0000 [r219305] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 219304 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r219304 | dvossel | 2009-09-17 16:59:21 -0500 (Thu, 17 Sep 2009)
- | 27 lines Merged revisions 219303 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r219303 | dvossel | 2009-09-17 16:29:37 -0500 (Thu, 17 Sep 2009)
- | 21 lines INVITE w/Replaces deadlock fix This patch cleans up
- the locking logic in chan_sip.c's handle_invite_replaces()
- function as well as making use of ast_do_masquerade() rather than
- forcing the masquerade on an ast_read(). The code had several
- redundant unlocks that would result in 'freed more times than
- we've locked!' errors. I cleaned these up as well as moving all
- the unlock logic to the end of the function. This patch should
- also resolve the issue people were having with the replacecall
- channel never being unlocked with one legged calls. (closes issue
- #15151) Reported by: irroot Patches: invite_w_replaces_1.4.diff
- uploaded by dvossel (license 671) Tested by: irroot, dvossel
- Review: https://reviewboard.asterisk.org/r/371/ ........
- ................
-
-2009-09-17 19:58 +0000 [r219265] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 219264 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r219264 |
- file | 2009-09-17 14:57:39 -0500 (Thu, 17 Sep 2009) | 2 lines
- Ensure no spaces exist before "refresher=" when doing the
- comparison. ........
-
-2009-09-17 Leif Madsen <lmadsen@digium.com>
-
- * Released Asterisk 1.6.0.16-rc1
-
-2009-09-17 15:44 +0000 [r219198] Matthew Nicholson <mnicholson@digium.com>
-
- * main/channel.c, /, include/asterisk/cdr.h,
- include/asterisk/channel.h: Merged revisions 219139 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r219139 | mnicholson | 2009-09-17 10:18:01 -0500
- (Thu, 17 Sep 2009) | 17 lines Merged revisions 219136 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r219136 | mnicholson | 2009-09-17 09:58:39 -0500 (Thu, 17 Sep
- 2009) | 10 lines Prevent a potential race condition and crash
- when hanging up a channel by removing the channel from the
- channel list before begining channel tear down. This fix may
- potentially cause problems with CDR backends that access the
- channel a CDR is associated with via the channel list. This fix
- makes the channel unavabile at the time when the CDR backend is
- invoked. This has been documented in include/asterisk/cdr.h.
- (closes issue #15316) Reported by: vmarrone Tested by: mnicholson
- Review: https://reviewboard.asterisk.org/r/362/ ........
- ................
-
-2009-09-16 23:52 +0000 [r219064] Tilghman Lesher <tlesher@digium.com>
-
- * main/config.c, configs/extensions.conf.sample, /: Merged
- revisions 219061 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r219061 | tilghman | 2009-09-16 18:42:12 -0500 (Wed, 16 Sep 2009)
- | 15 lines Merged revisions 219023 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r219023 | tilghman | 2009-09-16 18:21:53 -0500 (Wed, 16 Sep 2009)
- | 8 lines Properly deal with quotes in the arguments of '#exec'
- includes. (closes issue #15583) Reported by: pkempgen Patches:
- 20090726__issue15583.diff.txt uploaded by tilghman (license 14)
- 20090726__issue15583-1.4-4.diff.txt uploaded by pkempgen (license
- 169) Tested by: pkempgen ........ ................
-
-2009-09-16 19:26 +0000 [r218935] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 218933 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r218933 |
- mmichelson | 2009-09-16 14:25:36 -0500 (Wed, 16 Sep 2009) | 12
- lines Reverse order of args to fread. This way, we don't always
- write a null byte into byte 1 of the buffer (closes issue #15905)
- Reported by: ebroad Patches: freadfix.patch uploaded by ebroad
- (license 878) Tested by: ebroad ........
-
-2009-09-16 18:44 +0000 [r218931] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 218918 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r218918 |
- file | 2009-09-16 13:31:47 -0500 (Wed, 16 Sep 2009) | 5 lines On
- TCP and TLS connections do not attempt to stop retransmission of
- the packet internally. This was preventing responses from being
- properly processed because the packet was not being found causing
- handle_response to return prematurely. ........
-
-2009-09-16 18:11 +0000 [r218869] David Brooks <dbrooks@digium.com>
-
- * main/pbx.c, /: Merged revisions 218868 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r218868 | dbrooks | 2009-09-16 13:06:42 -0500 (Wed, 16 Sep 2009)
- | 20 lines Merged revisions 218867 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r218867 | dbrooks | 2009-09-16 13:00:45 -0500 (Wed, 16 Sep 2009)
- | 13 lines Fixes CID pattern matching behavior to mirror that of
- extension pattern matching. Pattern matching for extensions uses
- a type of scoring system, giving values for specificity to each
- character in the pattern. Unfortunately, this is done character
- by character, in order. This does lead to some less specific
- patterns being first in line for matching, but it will usually
- get the job done. This patch merely brings CID matching to the
- same level as extension matching. This patch does not attempt to
- tackle the problem shared by extension matching. (closes issue
- #14708) Reported by: klaus3000 ........ ................
-
-2009-09-16 13:36 +0000 [r218800] Russell Bryant <russell@digium.com>
-
- * contrib/firmware/iax/iaxy.bin (removed), /, UPGRADE.txt: Merged
- revisions 218799 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r218799 | russell | 2009-09-16 08:34:41 -0500 (Wed, 16 Sep 2009)
- | 16 lines Merged revisions 218798 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r218798 | russell | 2009-09-16 08:33:43 -0500 (Wed, 16 Sep 2009)
- | 9 lines Remove the IAXy firmware from Asterisk. The firmware
- can now be found on downloads.digium.com, where the rest of our
- binary downloads live. This was the last part of our Asterisk
- tarballs that was considered non-free by Debian. :-) (closes
- issue #15838) Reported by: paravoid ........ ................
-
-2009-09-15 22:39 +0000 [r218732] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 218731 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r218731 | tilghman | 2009-09-15 17:33:10 -0500
- (Tue, 15 Sep 2009) | 13 lines Merged revisions 218730 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r218730 | tilghman | 2009-09-15 17:27:41 -0500 (Tue, 15 Sep 2009)
- | 6 lines If the user enters the same password as before, don't
- signal an error when the change does nothing. (closes issue
- #15492) Reported by: cbbs70a Patches:
- 20090713__issue15492.diff.txt uploaded by tilghman (license 14)
- ........ ................
-
-2009-09-15 19:31 +0000 [r218690] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 218687 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r218687 |
- dvossel | 2009-09-15 14:22:37 -0500 (Tue, 15 Sep 2009) | 2 lines
- upward bound checking for port string to int conversion ........
-
-2009-09-15 16:21 +0000 [r218601] Matthew Nicholson <mnicholson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 218586 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r218586 | mnicholson | 2009-09-15 11:15:02 -0500 (Tue, 15 Sep
- 2009) | 15 lines Merged revisions 218578 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r218578 | mnicholson | 2009-09-15 11:03:54 -0500 (Tue, 15 Sep
- 2009) | 8 lines Send request contact header field with response
- to registrer queries instead of the address of record. (closes
- issue #14438) Reported by: ravindrad Patches: regquerypatch
- uploaded by ravindrad (license 684) Tested by: ravindrad ........
- ................
-
-2009-09-15 16:05 +0000 [r218580] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_followme.c: Merged revisions 218579 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r218579 | tilghman | 2009-09-15 11:04:41 -0500 (Tue, 15 Sep 2009)
- | 16 lines Merged revisions 218577 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r218577 | tilghman | 2009-09-15 11:01:17 -0500 (Tue, 15 Sep 2009)
- | 9 lines Ensure FollowMe sets language in channels it creates.
- Also, not in the original bug report, but related fields are
- accountcode and musicclass, and the inheritance of datastores.
- (closes issue #15372) Reported by: Romik Patches:
- 20090828__issue15372.diff.txt uploaded by tilghman (license 14)
- Tested by: cervajs ........ ................
-
-2009-09-15 15:42 +0000 [r218505-218573] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 218566 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r218566 |
- mmichelson | 2009-09-15 10:40:14 -0500 (Tue, 15 Sep 2009) | 4
- lines Use a better method of ensuring null-termination of the
- buffer while reading the SDP when using TCP. ........
-
- * /, channels/chan_sip.c: Merged revisions 218499,218504 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r218499 | mmichelson | 2009-09-15 09:59:50 -0500 (Tue,
- 15 Sep 2009) | 3 lines Fix off-by-one error when reading SDP sent
- over TCP. ........ r218504 | mmichelson | 2009-09-15 10:05:53
- -0500 (Tue, 15 Sep 2009) | 3 lines Ensure that SDP read from TCP
- socket is null-terminated. ........
-
-2009-09-15 15:03 +0000 [r218501] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, sounds/Makefile: Merged revisions 218500 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r218500 | kpfleming | 2009-09-15 11:02:21 -0400 (Tue, 15 Sep
- 2009) | 9 lines Merged revisions 218497 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r218497 | kpfleming | 2009-09-15 10:55:58 -0400 (Tue, 15 Sep
- 2009) | 1 line Use proper hostname for downloading sound files.
- ........ ................
-
-2009-09-14 22:49 +0000 [r218431] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c: Merged revisions 218430 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r218430 | jpeeler | 2009-09-14 17:38:25 -0500 (Mon, 14 Sep 2009)
- | 18 lines Merged revisions 218401 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r218401 | jpeeler | 2009-09-14 16:47:11 -0500 (Mon, 14 Sep 2009)
- | 11 lines Fix handling of DAHDI_EVENT_REMOVED event to prevent
- crash in do_monitor. After talking to rmudgett about some of his
- recent iflist locking changes, it was determined that the only
- place that would destroy a channel without being explicitly to do
- so was in handle_init_event. The loop to walk the interface list
- has been modified to wait to destroy the channel until the
- dahdi_pvt of the channel to be destroyed is no longer needed.
- (closes issue #15378) Reported by: samy ........ ................
-
-2009-09-14 19:49 +0000 [r218362] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c, /, configs/voicemail.conf.sample,
- sounds/Makefile: Merged revisions 218361 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r218361 | tilghman | 2009-09-14 14:29:48 -0500 (Mon, 14 Sep 2009)
- | 11 lines Recorded merge of revisions 218331 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r218331 | tilghman | 2009-09-14 14:16:35 -0500 (Mon, 14 Sep 2009)
- | 4 lines Don't say "Please try again" if we don't give the user
- another chance to try again. (issue #15055, SWP-129) Reported by:
- jthurman ........ ................
-
-2009-09-14 15:22 +0000 [r218244] Matthew Nicholson <mnicholson@digium.com>
-
- * /, apps/app_directed_pickup.c: Merged revisions 218224 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r218224 | mnicholson | 2009-09-14 09:57:23 -0500
- (Mon, 14 Sep 2009) | 14 lines Merged revisions 218223 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r218223 | mnicholson | 2009-09-14 09:53:57 -0500 (Mon, 14 Sep
- 2009) | 8 lines Ensure we don't pickup ourselves when doing
- pickup by exten. (closes issue #15100) Reported by: lmsteffan
- Patches: (modified) pickup.patch uploaded by lmsteffan (license
- 779) ........ ................
-
-2009-09-13 19:10 +0000 [r218216] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
-
- * channels/chan_phone.c, /: gcc 4.4: Remove a nop memset size 0
- that annoys gcc This memset doesn't write beyond the end of the
- buffer. (tmpbuf has size of 4). Merged revisions 218184 via
- svnmerge from http://svn.digium.com/svn/asterisk/trunk
-
-2009-09-12 13:10 +0000 [r218108] Michiel van Baak <michiel@vanbaak.info>
-
- * main/rtp.c: Use the ip for the new 'rtp set debug ip <foo>'.
- Since 1.6.X still has the deprecated 'rtp debug ip <foo>' this
- patch is different from the fix that went into trunk (closes
- issue #15711) Reported by: davidw Patches:
- 2009082800-rtpdebug.diff.txt uploaded by mvanbaak (license 7)
- Tested by: davidw
-
-2009-09-11 05:58 +0000 [r217920-218051] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 217990 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r217990 | tilghman | 2009-09-10 18:54:51 -0500 (Thu, 10 Sep 2009)
- | 10 lines Merged revisions 217989 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r217989 | tilghman | 2009-09-10 18:52:22 -0500 (Thu, 10 Sep 2009)
- | 3 lines Don't ring another channel, if there's not enough time
- for a queue member to answer. (Fixes AST-228) ........
- ................
-
- * contrib/scripts/iax-friends.sql, /, channels/chan_sip.c,
- channels/chan_iax2.c: Merged revisions 217916 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r217916 |
- tilghman | 2009-09-10 18:12:16 -0500 (Thu, 10 Sep 2009) | 2 lines
- Make calltoken support work with realtime users and peers.
- ........
-
-2009-09-10 22:31 +0000 [r217858-217913] David Vossel <dvossel@digium.com>
-
- * channels/chan_sip.c: sip peer matching by address only with
- TCP/TLS This patch removes the contact header matching logic and
- adds logic to match all tcp/tls connections by ip only. Thanks to
- oej for finding the issue and suggesting solutions. Review:
- https://reviewboard.asterisk.org/r/355/
-
- * /, channels/chan_iax2.c: Merged revisions 217807 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r217807 | dvossel | 2009-09-10 16:07:47 -0500
- (Thu, 10 Sep 2009) | 28 lines Merged revisions 217806 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r217806 | dvossel | 2009-09-10 16:06:07 -0500 (Thu, 10 Sep 2009)
- | 22 lines IAX2 encryption regression The IAX2 Call Token
- security patch inadvertently broke the use of encryption due to
- the reorganization of code in the socket_process() function. When
- encryption is used, an incoming full frame must first be
- decrypted before the information elements can be parsed. The
- security release mistakenly moved IE parsing before decryption in
- order to process the new Call Token IE. To resolve this,
- decryption of full frames is once again done before looking into
- the frame. This involves searching for an existing callno,
- checking the pvt to see if encryption is turned on, and
- decrypting the packet before the internal fields of the full
- frame are accessed. (closes issue #15834) Reported by: karesmakro
- Patches: iax2_encryption_fix_1.4.diff uploaded by dvossel
- (license 671) Tested by: dvossel, karesmakro Review:
- https://reviewboard.asterisk.org/r/355/ ........ ................
-
-2009-09-10 19:53 +0000 [r217736] mnick <mnick@localhost>:
-
- * /, res/res_musiconhold.c: Merged revisions 217730 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r217730 | mnick | 2009-09-10 14:39:41 -0500 (Thu, 10 Sep 2009) |
- 17 lines Sets the correct musicclass after an announcement
- (closes issue #15279) Reported by: mbeckwell Patches: patch.txt
- uploaded by mnick (license ) Tested by: mnick (closes issue
- #15832) Reported by: mbeckwell Patches: patch.txt uploaded by
- mnick (license 874) Tested by: mnick ........
-
-2009-09-10 12:16 +0000 [r217596] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 217593 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r217593 |
- oej | 2009-09-10 14:06:55 +0200 (Tor, 10 Sep 2009) | 8 lines
- Include ActionID in all events that are responsed to AMI Action
- SIPShowRegistry (closes issue #15868) Reported by: nic_bellamy
- Patches: manager_SIPshowregistry_actionid.patch uploaded by nic
- bellamy (license 299) ........
-
-2009-09-09 20:15 +0000 [r217484] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
-
- * /, res/res_phoneprov.c: gcc 4.4 fix: union instead of cast gcc
- 4.4 has more strict rules for aliasing. It doesn't like a struct
- sockaddr_in pointer pointing to a struct sockaddr. So we make it
- a union. Merged revisions 217445 via svnmerge from
- http://svn.digium.com/svn/asterisk/trunk
-
-2009-09-09 11:33 +0000 [r217405] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 217368 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r217368 |
- oej | 2009-09-09 12:39:43 +0200 (Ons, 09 Sep 2009) | 2 lines Not
- having any TLS session to write to is a serious XMIT_ERROR.
- ........
-
-2009-09-08 21:45 +0000 [r217281] Kevin P. Fleming <kpfleming@digium.com>
-
- * configure: Commit regenerated configure script that I missed
- earlier.
-
-2009-09-08 20:31 +0000 [r217209] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 217199 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r217199 | tilghman | 2009-09-08 15:28:41 -0500 (Tue, 08 Sep 2009)
- | 14 lines Merged revisions 217156 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r217156 | tilghman | 2009-09-08 15:01:45 -0500 (Tue, 08 Sep 2009)
- | 7 lines When MOH is playing on the channel, announcements sent
- through the conference are not heard. (closes issue #14588)
- Reported by: voipas Patches: 20090716__issue14588__2.diff.txt
- uploaded by tilghman (license 14) Tested by: lmadsen, twisted,
- tilghman ........ ................
-
-2009-09-08 16:38 +0000 [r217075] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, include/asterisk/autoconfig.h.in, configure.ac: Merged
- revisions 217074 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r217074 |
- kpfleming | 2009-09-08 11:37:28 -0500 (Tue, 08 Sep 2009) | 9
- lines Ensure that the default autoconf CFLAGS are not used. A
- recent change to the configure script that allows the user to
- specify CFLAGS and/or LDFLAGS to the script had the unfortunate
- side effect of letting autoconf's default CFLAGS (-g -O2) feed in
- to the rest of the build system, thereby overriding the
- DONT_OPTIMIZE setting in menuselect. That problem is now
- corrected. ........
-
-2009-09-08 15:35 +0000 [r217034] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_limit.c: Merged revisions 217033 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r217033 |
- tilghman | 2009-09-08 10:30:18 -0500 (Tue, 08 Sep 2009) | 4 lines
- Remove what appears to be an unnecessary define. (closes issue
- #15851) Reported by: tzafrir ........
-
-2009-09-08 14:28 +0000 [r216996] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 216993 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r216993 |
- dvossel | 2009-09-08 09:26:30 -0500 (Tue, 08 Sep 2009) | 14 lines
- caller id number empty parse_uri was not being given the correct
- scheme's, as a result, uri parsing did not parse the username
- correctly. One of the side effects of this is an empty caller id.
- (closes issue #15839) Reported by: ebroad Patches:
- blank_cidv2.patch uploaded by ebroad (license 878)
- parse_uri_fix.diff uploaded by dvossel (license 671) Tested by:
- ebroad, dvossel ........
-
-2009-09-07 16:38 +0000 [r216645-216843] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 216842 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r216842 |
- oej | 2009-09-07 18:35:12 +0200 (MÃ¥n, 07 Sep 2009) | 2 lines
- Make sure we reset global_exclude_static at channel reload
- ........
-
- * /, channels/chan_sip.c: Merged revisions 216695 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r216695 |
- oej | 2009-09-07 15:06:19 +0200 (MÃ¥n, 07 Sep 2009) | 8 lines If
- there is no session timer in the INVITE, set it to default value
- (not unset minimum = -1) Patch by oej closes issue #15621
- Reported by: fnordian Tested by: atis ........
-
- * configs/sip.conf.sample: fix documentation so it agrees with code
-
- * channels/chan_sip.c, CHANGES: Add doc and turn off premature
- media filter by default
-
- * apps/app_playback.c, main/pbx.c, /, channels/chan_sip.c,
- apps/app_disa.c, configs/sip.conf.sample: Merged revisions 216438
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r216438 | oej | 2009-09-04 16:02:34 +0200 (Fre,
- 04 Sep 2009) | 35 lines Merged revisions 216430 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27
- lines Make apps send PROGRESS control frame for early media and
- fix too early media issue in SIP The issue at hand is that some
- legacy (dying) PBX systems send empty media frames on PRI links
- *before* any call progress. The SIP channel receives these frames
- and by default signals 183 Session progress and starts sending
- media. This will cause phones to play silence and ignore the
- later 180 ringing message. A bad user experience. The fix is
- twofold: - We discovered that asterisk apps that support early
- media ("noanswer") did not send any PROGRESS frame to indicate
- early media. Fixed. - We introduce a setting in chan_sip so that
- users can disable any relay of media frames before the outbound
- channel actually indicates any sort of call progress. In 1.4,
- 1.6.0 and 1.6.1, this will be disabled for backward
- compatibility. In later versions of Asterisk, this will be
- enabled. We don't assume that it will change your Asterisk phone
- experience - only for the better. We encourage third-party
- application developers to make sure that if they have
- applications that wants to send early media, add a PROGRESS
- control frame transmission to make sure that all channel drivers
- actually will start sending early media. This has not been the
- default in Asterisk previous to this patch, so if you got
- inspiration from our code, you need to update accordingly. Sorry
- for the trouble and thanks for your support. This code has been
- running for a few months in a large scale installation (over 250
- servers with PRI and/or BRI links to old PBX systems). That's no
- proof that this is an excellent patch, but, well, it's tested :-)
- ........ ................
-
-2009-09-04 19:32 +0000 [r216595] Sean Bright <sean@malleable.com>
-
- * apps/app_voicemail.c, /: Merged revisions 216593 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r216593 | seanbright | 2009-09-04 15:29:02 -0400 (Fri, 04 Sep
- 2009) | 1 line Use ast_free() instead of free(). ........
-
-2009-09-04 17:32 +0000 [r216548] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c, /, UPGRADE-1.6.txt: Merged revisions 216547 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r216547 | tilghman | 2009-09-04 12:31:44 -0500 (Fri, 04
- Sep 2009) | 3 lines Enable turning off the application delimiter
- warning with the 'dontwarn' option. Suggested on the -dev list,
- and implemented in an alternate way by me. ........
-
-2009-09-04 15:07 +0000 [r216507] Michiel van Baak <michiel@vanbaak.info>
-
- * /, main/utils.c: Merged revisions 216506 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r216506 | mvanbaak | 2009-09-04 17:05:05 +0200 (Fri, 04 Sep 2009)
- | 9 lines Merged revisions 216435 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r216435 | mvanbaak | 2009-09-04 15:56:10 +0200 (Fri, 04 Sep 2009)
- | 2 lines make asterisk compile under devmode with DEBUG_THREADS
- enabled on OpenBSD ........ ................
-
-2009-09-04 10:49 +0000 [r216265] Russell Bryant <russell@digium.com>
-
- * doc/IAX2-security.txt (added), /: Merged revisions 216264 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r216264 | russell | 2009-09-04 05:48:44 -0500
- (Fri, 04 Sep 2009) | 16 lines Merged revisions 216263 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r216263 | russell | 2009-09-04 05:48:00 -0500
- (Fri, 04 Sep 2009) | 9 lines Merged revisions 216262 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r216262 | russell | 2009-09-04 05:47:37 -0500 (Fri, 04
- Sep 2009) | 2 lines Add a plain text version of the IAX2 security
- document. ........ ................ ................
-
-2009-09-04 06:11 +0000 [r216223] Michiel van Baak <michiel@vanbaak.info>
-
- * main/astobj2.c, /: Merged revisions 216222 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r216222 |
- mvanbaak | 2009-09-04 08:08:33 +0200 (Fri, 04 Sep 2009) | 3 lines
- make sure 'start' is always initialized. Makes asterisk compile
- with --enable-dev-mode ........
-
-2009-09-03 19:42 +0000 [r216011-216097] Russell Bryant <russell@digium.com>
-
- * UPGRADE.txt: tweak
-
- * /, UPGRADE.txt: Merged revisions 216092 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r216092 | russell | 2009-09-03 14:38:35 -0500 (Thu, 03 Sep 2009)
- | 16 lines Merged revisions 216085 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r216085 | russell | 2009-09-03 14:36:46 -0500
- (Thu, 03 Sep 2009) | 9 lines Merged revisions 216080 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r216080 | russell | 2009-09-03 14:35:23 -0500 (Thu, 03
- Sep 2009) | 2 lines Add a note about IAX2 to UPGRADE.txt.
- ........ ................ ................
-
- * /, doc/IAX2-security.pdf (added): Merged revisions 216009 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r216009 | russell | 2009-09-03 13:45:54 -0500
- (Thu, 03 Sep 2009) | 16 lines Merged revisions 216008 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r216008 | russell | 2009-09-03 13:44:58 -0500
- (Thu, 03 Sep 2009) | 9 lines Merged revisions 216005 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.2
- ........ r216005 | russell | 2009-09-03 13:42:24 -0500 (Thu, 03
- Sep 2009) | 2 lines Add IAX2 security document related to
- AST-2009-006. ........ ................ ................
-
-2009-09-03 18:40 +0000 [r216003] David Vossel <dvossel@digium.com>
-
- * channels/iax2-parser.c, main/astobj2.c, configs/iax.conf.sample,
- include/asterisk/acl.h, channels/iax2-parser.h, /,
- include/asterisk/astobj2.h, channels/iax2.h, main/acl.c,
- channels/chan_iax2.c: Merged revisions 215955 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r215955 |
- dvossel | 2009-09-03 11:31:54 -0500 (Thu, 03 Sep 2009) | 6 lines
- Merge code associated with AST-2009-006 (closes issue #12912)
- Reported by: rathaus Tested by: tilghman, russell, dvossel,
- dbrooks ........
-
-2009-09-03 Leif Madsen <lmadsen@digium.com>
-
- * Asterisk 1.6.0.15 released
-
- * AST-2009-006
-
-2009-08-28 Leif Madsen <lmadsen@digium.com>
-
- * Asterisk 1.6.0.14 released
-
-2009-08-11 Tilghman Lesher <tlesher@digium.com>
-
- * Released 1.6.0.14-rc1
-
-2009-08-10 19:51 +0000 [r211551-211587] Tilghman Lesher <tlesher@digium.com>
-
- * doc/CODING-GUIDELINES, /: Merged revisions 211584 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r211584 | tilghman | 2009-08-10 14:49:41 -0500
- (Mon, 10 Aug 2009) | 9 lines Merged revisions 211583 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r211583 | tilghman | 2009-08-10 14:48:48 -0500 (Mon, 10
- Aug 2009) | 1 line Conversion specifiers, not format specifiers
- ........ ................
-
- * res/res_config_curl.c, apps/app_waitforring.c,
- channels/chan_misdn.c, funcs/func_channel.c, apps/app_macro.c,
- pbx/pbx_config.c, apps/app_chanspy.c, apps/app_mixmonitor.c,
- res/res_odbc.c, main/asterisk.c, apps/app_voicemail.c,
- doc/CODING-GUIDELINES, utils/muted.c, apps/app_meetme.c,
- main/utils.c, cdr/cdr_pgsql.c, res/res_musiconhold.c,
- apps/app_followme.c, channels/misdn_config.c, utils/frame.c,
- main/channel.c, main/cdr.c, res/ael/pval.c, funcs/func_enum.c,
- channels/chan_phone.c, apps/app_osplookup.c,
- apps/app_setcallerid.c, main/manager.c, funcs/func_odbc.c,
- apps/app_minivm.c, res/res_agi.c, res/res_config_ldap.c,
- apps/app_rpt.c, channels/chan_mgcp.c, apps/app_adsiprog.c,
- funcs/func_dialplan.c, main/dnsmgr.c, channels/chan_sip.c,
- res/res_limit.c, apps/app_waitforsilence.c, agi/eagi-test.c,
- apps/app_waituntil.c, main/acl.c, apps/app_queue.c,
- channels/chan_oss.c, agi/eagi-sphinx-test.c,
- channels/chan_usbradio.c, res/snmp/agent.c, pbx/pbx_dundi.c,
- apps/app_sms.c, utils/extconf.c, apps/app_verbose.c,
- apps/app_stack.c, apps/app_dahdibarge.c, funcs/func_rand.c,
- apps/app_readfile.c, main/frame.c, /, apps/app_record.c,
- funcs/func_strings.c, cdr/cdr_adaptive_odbc.c,
- apps/app_alarmreceiver.c, channels/chan_iax2.c,
- main/indications.c, main/config.c, main/cli.c,
- pbx/pbx_loopback.c, channels/chan_dahdi.c, pbx/pbx_spool.c,
- res/res_smdi.c, channels/chan_skinny.c, main/features.c,
- main/http.c, main/pbx.c, apps/app_privacy.c,
- codecs/codec_speex.c, funcs/func_math.c, channels/chan_agent.c,
- apps/app_morsecode.c, apps/app_disa.c, funcs/func_cut.c,
- channels/iax2-provision.c, pbx/dundi-parser.c,
- apps/app_talkdetect.c: AST-2009-005
-
-2009-08-10 14:10 +0000 [r211348] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 211347 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r211347 |
- file | 2009-08-10 11:07:44 -0300 (Mon, 10 Aug 2009) | 5 lines Fix
- retrieval of the port used for the video stream when adding SDP
- to a SIP message. (closes issue #15121) Reported by: jsmith
- ........
-
-2009-08-09 15:43 +0000 [r211233-211276] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/astfd.c: Merged revisions 211275 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r211275 | tilghman | 2009-08-09 10:42:02 -0500 (Sun, 09 Aug 2009)
- | 9 lines Merged revisions 211274 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r211274 | tilghman | 2009-08-09 10:41:01 -0500 (Sun, 09 Aug 2009)
- | 2 lines Small oops. Clear the flags which have been checked.
- ........ ................
-
- * apps/app_stack.c, /: Merged revisions 211232 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r211232 |
- tilghman | 2009-08-09 02:11:22 -0500 (Sun, 09 Aug 2009) | 4 lines
- Check for NULL frame, before dereferencing pointer. (closes issue
- #15617) Reported by: rain ........
-
-2009-08-07 20:14 +0000 [r211114] Russell Bryant <russell@digium.com>
-
- * apps/app_chanspy.c, /: Merged revisions 211113 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r211113 | russell | 2009-08-07 15:12:21 -0500 (Fri, 07 Aug 2009)
- | 11 lines Recorded merge of revisions 211112 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r211112 | russell | 2009-08-07 15:11:31 -0500 (Fri, 07 Aug 2009)
- | 4 lines Resolve a deadlock involving app_chanspy and
- masquerades. (ABE-1936) ........ ................
-
-2009-08-07 18:18 +0000 [r211044] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 211040 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r211040 | tilghman | 2009-08-07 13:17:41 -0500 (Fri, 07 Aug 2009)
- | 21 lines Merged revisions 211038 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r211038 | tilghman | 2009-08-07 13:16:28 -0500 (Fri, 07 Aug 2009)
- | 14 lines QUEUE_MEMBER_LIST _really_ wants the interface name,
- not the membername. This is a partial revert of revision 82590,
- which was an attempted cleanup, but in reality, it broke
- QUEUE_MEMBER_LIST, which has always been intended as a method by
- which component interfaces could be queried from the queue.
- Membername isn't useful here, because that field cannot be used
- to obtain further information about the member. See the
- documentation on QUEUE_MEMBER_LIST, RemoveQueueMember,
- QUEUE_MEMBER_PENALTY, and the various AMI commands which take a
- member argument for further justification. (closes issue #15664)
- Reported by: rain Patches: app_queue-queue_member_list.diff
- uploaded by rain (license 327) ........ ................
-
-2009-08-07 13:08 +0000 [r210993] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/udptl.c, /: Merged revisions 210992 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r210992 |
- kpfleming | 2009-08-07 08:08:00 -0500 (Fri, 07 Aug 2009) | 13
- lines Workaround broken T.38 endpoints that offer tiny
- MaxDatagram sizes. Some T.38 endpoints treat T38FaxMaxDatagram as
- the maximum IFP size that should be sent to them, rather than the
- maximum packet payload size. If such an endpoint also requests
- UDPRedundancy as the error correction mode, we'll end up
- calculating a tiny maximum IFP size, so small as to be unusable.
- This patch sets a lower bound on what we'll consider the remote's
- maximum IFP size to be, assuming that endpoints that do this
- really can accept larger packets than they've offered to accept.
- (closes issue #15649) Reported by: dazza76 ........
-
-2009-08-06 21:46 +0000 [r210909-210915] Tilghman Lesher <tlesher@digium.com>
-
- * main/channel.c, /: Merged revisions 210914 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r210914 | tilghman | 2009-08-06 16:46:01 -0500 (Thu, 06 Aug 2009)
- | 14 lines Merged revisions 210913 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r210913 | tilghman | 2009-08-06 16:45:01 -0500 (Thu, 06 Aug 2009)
- | 7 lines Because channel information can be accessed outside of
- the channel thread, we must lock the channel prior to modifying
- it. (closes issue #15397) Reported by: caspy Patches:
- 20090714__issue15397.diff.txt uploaded by tilghman (license 14)
- Tested by: caspy ........ ................
-
- * apps/app_stack.c, include/asterisk/app.h, /, main/app.c: Merged
- revisions 210908 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r210908 |
- tilghman | 2009-08-06 16:29:26 -0500 (Thu, 06 Aug 2009) | 9 lines
- Allow Gosub to recognize quote delimiters without consuming them.
- (closes issue #15557) Reported by: rain Patches:
- 20090723__issue15557.diff.txt uploaded by tilghman (license 14)
- Tested by: rain Review: https://reviewboard.asterisk.org/r/316/
- ........
-
-2009-08-06 17:47 +0000 [r210818] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 210817 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r210817 |
- file | 2009-08-06 14:47:04 -0300 (Thu, 06 Aug 2009) | 11 lines
- Accept additional T.38 reinvites after an initial one has been
- handled. Discussion of this subject has yielded that it is not
- actually acceptable to change T.38 parameters after the initial
- reinvite but declining is harsh and can cause the fax to fail
- when it may be possible to allow it to continue. This patch
- changes things so that additional T.38 reinvites are accepted but
- parameter changes ignored. This gives the fax a fighting chance.
- (closes issue #15610) Reported by: huangtx2009 ........
-
-2009-08-05 20:07 +0000 [r210647] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 210640 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r210640 | rmudgett | 2009-08-05 14:40:03 -0500
- (Wed, 05 Aug 2009) | 21 lines Merged revisions 210575 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r210575 | rmudgett | 2009-08-05 14:18:56 -0500 (Wed, 05 Aug 2009)
- | 14 lines Dialplan starts execution before the channel setup is
- complete. * Issue 15655: For the case where dialing is complete
- for an incoming call, dahdi_new() was asked to start the PBX and
- then the code set more channel variables. If the dialplan hungup
- before these channel variables got set, asterisk would likely
- crash. * Fixed potential for overlap incoming call to erroneously
- set channel variables as global dialplan variables if the
- ast_channel structure failed to get allocated. * Added missing
- set of CALLINGSUBADDR in the dialing is complete case. (closes
- issue #15655) Reported by: alecdavis ........ ................
-
-2009-08-05 18:57 +0000 [r210568] Leif Madsen <lmadsen@digium.com>
-
- * doc/tex/imapstorage.tex, /: Merged revisions 210564 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r210564 | lmadsen | 2009-08-05 13:49:58 -0500
- (Wed, 05 Aug 2009) | 19 lines Merged revisions 210563 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r210563 | lmadsen | 2009-08-05 13:46:21 -0500 (Wed, 05 Aug 2009)
- | 11 lines Update imapstorage.txt documentation. Updated the
- imapstorage.txt documentation to reflect that issues with
- c-client versions older than 2007 seem to cause crashing issues
- that are not seen with more recent versions. Documentation has
- been updated to reflect this. (closes issue #14496) Reported by:
- vbcrlfuser Patches: __20090727-imap-documentation-patch.txt
- uploaded by lmadsen (license 10) Tested by: lmadsen, mmichelson,
- dbrooks ........ ................
-
-2009-08-04 14:54 +0000 [r210239] Kevin P. Fleming <kpfleming@digium.com>
-
- * Makefile, /: Merged revisions 210238 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r210238 | kpfleming | 2009-08-04 09:53:00 -0500 (Tue, 04 Aug
- 2009) | 16 lines Merged revisions 210237 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r210237 | kpfleming | 2009-08-04 09:51:39 -0500 (Tue, 04 Aug
- 2009) | 10 lines Eliminate spurious compiler warnings from system
- headers on *BSD platforms. Ensure that system headers located in
- /usr/local/include are actually treated as system headers by the
- compiler, and not as local headers which are subject to warnings
- from the -Wundef compiler option and others. (closes issue
- #15606) Reported by: mvanbaak ........ ................
-
-2009-08-01 11:31 +0000 [r209840-209896] Russell Bryant <russell@digium.com>
-
- * main/db1-ast/mpool/mpool.c, /: Merged revisions 209887 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r209887 | russell | 2009-08-01 06:29:25 -0500
- (Sat, 01 Aug 2009) | 12 lines Merged revisions 209879 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r209879 | russell | 2009-08-01 06:27:25 -0500 (Sat, 01 Aug 2009)
- | 5 lines Resolve a valgrind warning about a read from
- uninitialized memory. (issue #15396) Reported by: aragon ........
- ................
-
- * apps/app_milliwatt.c, /: Merged revisions 209839 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r209839 | russell | 2009-08-01 06:02:07 -0500
- (Sat, 01 Aug 2009) | 20 lines Merged revisions 209838 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r209838 | russell | 2009-08-01 05:59:05 -0500 (Sat, 01 Aug 2009)
- | 13 lines Modify how Playtones() is used in Milliwatt() to
- resolve gain issue. When Milliwatt() was changed internally to
- use Playtones() so that the proper tone was used, it introduced a
- drop in gain in the output signal. So, use the playtones API
- directly and specify a volume argument such that the output
- matches the gain of the original Milliwatt() code. (closes issue
- #15386) Reported by: rue_mohr Patches: issue_15386.rev2.diff
- uploaded by russell (license 2) Tested by: rue_mohr ........
- ................
-
-2009-08-01 01:13 +0000 [r209762] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/misdn/isdn_lib.c, utils/frame.c, /, main/Makefile,
- channels/misdn/ie.c, main/event.c: Merged revisions 209760-209761
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r209760 | kpfleming | 2009-07-31 20:03:07 -0500
- (Fri, 31 Jul 2009) | 13 lines Merged revisions 209759 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r209759 | kpfleming | 2009-07-31 19:52:00 -0500 (Fri, 31 Jul
- 2009) | 7 lines Minor changes inspired by testing with latest
- GCC. The latest GCC (what will become 4.5.x) has a few new
- warnings, that in these cases found some either downright buggy
- code, or at least seriously poorly designed code that could be
- improved. ........ ................ r209761 | kpfleming |
- 2009-07-31 20:04:06 -0500 (Fri, 31 Jul 2009) | 1 line Revert
- accidental Makefile change. ................
-
-2009-07-31 21:56 +0000 [r209712] Russell Bryant <russell@digium.com>
-
- * /, main/event.c: Merged revisions 209711 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r209711 |
- russell | 2009-07-31 16:53:31 -0500 (Fri, 31 Jul 2009) | 2 lines
- Fix some places where ast_event_type was used instead of
- ast_event_ie_type. ........
-
-2009-07-30 16:37 +0000 [r209555-209587] David Brooks <dbrooks@digium.com>
-
- * channels/chan_dahdi.c: Merged revisions 209554 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r209554 |
- dbrooks | 2009-07-30 11:07:05 -0500 (Thu, 30 Jul 2009) | 6 lines
- Fixes numerous spelling errors. Patch submitted by alecdavis.
- (closes issue #15595) Reported by: alecdavis ........
-
- * include/asterisk/abstract_jb.h,
- contrib/init.d/rc.debian.asterisk, /, apps/app_sms.c,
- codecs/lpc10/pitsyn.c, channels/chan_console.c: Merged revisions
- 209554 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r209554 |
- dbrooks | 2009-07-30 11:07:05 -0500 (Thu, 30 Jul 2009) | 6 lines
- Fixes numerous spelling errors. Patch submitted by alecdavis.
- (closes issue #15595) Reported by: alecdavis ........
-
-2009-07-28 12:01 +0000 [r209394] Kevin P. Fleming <kpfleming@digium.com>
-
- * apps/app_fax.c: Correct error in backport of latest app_fax
- fixes.
-
-2009-07-28 00:19 +0000 [r209325] Tilghman Lesher <tlesher@digium.com>
-
- * /, sounds/sounds.xml: Merged revisions 209317 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r209317 | tilghman | 2009-07-27 19:14:12 -0500 (Mon, 27 Jul 2009)
- | 9 lines Merged revisions 209315 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r209315 | tilghman | 2009-07-27 19:12:03 -0500 (Mon, 27 Jul 2009)
- | 2 lines Publish French extra sounds ........ ................
-
-2009-07-27 21:44 +0000 [r209259-209280] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, apps/app_fax.c: Merged revisions 209279 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r209279 |
- kpfleming | 2009-07-27 16:43:36 -0500 (Mon, 27 Jul 2009) | 7
- lines Cleanup T.38 negotiation changes. Convert LOG_NOTICE
- messages about T.38 negotiation in debug level 1 messages, clean
- up some looping logic, and correct an improper use of ast_free()
- for freeing an ast_frame. ........
-
- * /, apps/app_fax.c: Merged revisions 209256 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r209256 |
- kpfleming | 2009-07-27 16:21:43 -0500 (Mon, 27 Jul 2009) | 10
- lines Make T.38 switchover in ReceiveFAX synchronous. In receive
- mode, if the channel that ReceiveFAX is running on supports T.38,
- we should *always* attempt to switch T.38, rather than listening
- for an incoming CNG tone and only triggering on that. The channel
- may be using a low-bitrate codec that distorts the CNG tone, the
- sending FAX endpoint may not send CNG at all, or there could be a
- variety of other reasons that we don't detect it, but in all
- those cases if T.38 is available we certainly want to use it.
- ........
-
-2009-07-27 20:23 +0000 [r209221] David Brooks <dbrooks@digium.com>
-
- * channels/chan_dahdi.c, channels/chan_vpb.cc, res/res_smdi.c, /,
- include/asterisk/module.h, main/features.c, res/res_agi.c,
- res/res_jabber.c: Merged revisions 209098 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r209098 |
- dbrooks | 2009-07-27 11:33:50 -0500 (Mon, 27 Jul 2009) | 6 lines
- Fixing typos. Replaces "recieved" with "received" and "initilize"
- with "initialize" (closes issue #15571) Reported by: alecdavis
- ........
-
-2009-07-27 20:16 +0000 [r209133-209198] Mark Michelson <mmichelson@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 209197 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r209197 | mmichelson | 2009-07-27 15:11:42 -0500 (Mon, 27 Jul
- 2009) | 9 lines Honor channel's music class when using realtime
- music on hold. (closes issue #15051) Reported by: alexh Patches:
- 15051.patch uploaded by mmichelson (license 60) Tested by: alexh
- ........
-
- * main/udptl.c, /, configs/udptl.conf.sample: Merged revisions
- 209132 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r209132 | mmichelson | 2009-07-27 12:50:04 -0500 (Mon, 27 Jul
- 2009) | 24 lines Merged revisions 209131 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r209131 | mmichelson | 2009-07-27 12:44:06 -0500 (Mon, 27 Jul
- 2009) | 18 lines Allow for UDPTL to use only even-numbered ports
- if desired. There are some VoIP providers out there that will not
- accept SDP offers with odd numbered UDPTL ports. While it is my
- personal opinion that these VoIP providers are misinterpreting
- RFC 2327, it really is not a big deal to play along with their
- silly little games. Of course, since restricting UDPTL ports to
- only even numbers reduces the range of available ports by half,
- so the option to use only even port numbers is off by default. A
- user can enable the behavior by setting use_even_ports=yes in
- udptl.conf. (closes issue #15182) Reported by: CGMChris Patches:
- 15182.patch uploaded by mmichelson (license 60) Tested by:
- CGMChris ........ ................
-
-2009-07-27 16:06 +0000 [r209061] David Brooks <dbrooks@digium.com>
-
- * res/res_smdi.c, pbx/pbx_dundi.c: Just replacing typos "recieved"
- with "received". From issue #15360, forgot to apply to trunk and
- other branches.
-
-2009-07-27 15:39 +0000 [r209057] Kevin P. Fleming <kpfleming@digium.com>
-
- * Makefile, /: Merged revisions 209056 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r209056 |
- kpfleming | 2009-07-27 10:38:59 -0500 (Mon, 27 Jul 2009) | 10
- lines Restore explicit export of ASTCFLAGS/ASTLDFLAGS and
- underscore-variants to sub-makes. During the recent Makefile
- improvements I made, it seemed the 'make' was automatically
- carrying down the ASTCFLAGS/ASTLDFLAGS settings to sub-makes, so
- I removed the explict export of them. However, there are some
- circumstances where make does this, and some where it does not,
- so I've brought them back to ensure they are always exported. I
- also removed an extraneous double setting of _ASTLDFLAGS on *BSD
- platforms. ........
-
-2009-07-27 01:21 +0000 [r208925] Jeff Peeler <jpeeler@digium.com>
-
- * /, main/translate.c, channels/chan_iax2.c: Merged revisions
- 208924 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r208924 | jpeeler | 2009-07-26 20:20:37 -0500 (Sun, 26 Jul 2009)
- | 9 lines Merged revisions 208923 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r208923 | jpeeler | 2009-07-26 20:18:31 -0500 (Sun, 26 Jul 2009)
- | 2 lines Fix logic errors from 208746 ........ ................
-
-2009-07-25 06:24 +0000 [r208752] Jeff Peeler <jpeeler@digium.com>
-
- * /, channels/chan_skinny.c, main/translate.c,
- channels/chan_iax2.c: Merged revisions 208749 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r208749 | jpeeler | 2009-07-25 01:23:18 -0500 (Sat, 25 Jul 2009)
- | 13 lines Merged revisions 208746 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r208746 | jpeeler | 2009-07-25 01:19:50 -0500 (Sat, 25 Jul 2009)
- | 7 lines Fix compiling under dev-mode with gcc 4.4.0. Mostly
- trivial changes, but I did not know of any other way to fix the
- "dereferencing type-punned pointer will break strict-aliasing
- rules" error without creating a tmp variable in chan_skinny.
- ........ ................
-
-2009-07-24 18:49 +0000 [r208594] Russell Bryant <russell@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 208593 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r208593 | russell | 2009-07-24 13:42:32 -0500 (Fri, 24 Jul 2009)
- | 14 lines Merged revisions 208592 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r208592 | russell | 2009-07-24 13:38:24 -0500 (Fri, 24 Jul 2009)
- | 7 lines Do not log an ERROR if autoservice_stop() returns -1.
- This does not indicate an error. A return of -1 just means that
- the channel has been hung up. (reported in #asterisk-dev)
- ........ ................
-
-2009-07-24 18:31 +0000 [r208589] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 208588 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r208588 | mmichelson | 2009-07-24 13:31:04 -0500 (Fri, 24 Jul
- 2009) | 16 lines Merged revisions 208587 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r208587 | mmichelson | 2009-07-24 13:26:50 -0500 (Fri, 24 Jul
- 2009) | 10 lines Only send a BYE when hanging up a channel that
- is up. For cases where Asterisk sends an INVITE and receives a
- non 2XX final response, Asterisk would follow the INVITE
- transaction by immediately sending a BYE, which was unnecessary.
- (closes issue #14575) Reported by: chris-mac ........
- ................
-
-2009-07-24 15:04 +0000 [r208468-208549] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/udptl.c, /, channels/chan_sip.c, include/asterisk/udptl.h:
- Merged revisions 208548 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r208548 |
- kpfleming | 2009-07-24 10:02:53 -0500 (Fri, 24 Jul 2009) | 8
- lines Resolve a T.38 negotiation issue left over from the
- udptl-updates merge. The udptl-updates branch that was merged
- yesterday failed to properly send back T.38 SDP responses with
- the correct error correction mode, if the incoming SDP from the
- other end caused us to change error correction modes. This patch
- corrects that situation. ........
-
- * UPGRADE.txt: Use correct formatting for T.38 change note in
- UPGRADE.txt
-
- * main/rtp.c, main/channel.c, main/udptl.c, main/frame.c, /,
- channels/chan_sip.c, apps/app_fax.c, UPGRADE.txt,
- include/asterisk/udptl.h, include/asterisk/frame.h: Merged
- revisions 208464 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r208464 |
- kpfleming | 2009-07-23 16:57:24 -0500 (Thu, 23 Jul 2009) | 46
- lines Rework of T.38 negotiation and UDPTL API to address
- interoperability problems Over the past couple of months, a
- number of issues with Asterisk negotiating (and successfully
- completing) T.38 sessions with various endpoints have been found.
- This patch attempts to address many of them, primarily focused
- around ensuring that the endpoints' MaxDatagram size is honored,
- and in addition by ensuring that T.38 session parameter
- negotiation is performed correctly according to the ITU T.38
- Recommendation. The major changes here are: 1) T.38 applications
- in Asterisk (app_fax) only generate/receive IFP packets, they do
- not ever work with UDPTL packets. As a result of this, they
- cannot be allowed to generate packets that would overflow the
- other endpoints' MaxDatagram size after the UDPTL stack adds any
- error correction information. With this patch, the application is
- told the maximum *IFP* size it can generate, based on a
- calculation using the far end MaxDatagram size and the active
- error correction mode on the T.38 session. The same is true for
- sending *our* MaxDatagram size to the remote endpoint; it is
- computed from the value that the application says it can accept
- (for a single IFP packet) combined with the active error
- correction mode. 2) All treatment of T.38 session parameters as
- 'capabilities' in chan_sip has been removed; these parameters are
- not at all like audio/video stream capabilities. There are strict
- rules to follow for computing an answer to a T.38 offer, and
- chan_sip now follows those rules, using the desired parameters
- from the application (or channel) that wants to accept the T.38
- negotiation. 3) chan_sip now stores and forwards
- ast_control_t38_parameters structures for tracking 'our' and
- 'their' T.38 session parameters; this greatly simplifies
- negotiation, especially for pass-through calls. 4) Since T.38
- negotiation without specifying parameters or receiving the final
- negotiated parameters is not very worthwhile, the AST_CONTROL_T38
- control frame has been removed. A note has been added to
- UPGRADE.txt about this removal, since any out-of-tree
- applications that use it will no longer function properly until
- they are upgraded to use AST_CONTROL_T38_PARAMETERS. Review:
- https://reviewboard.asterisk.org/r/310/ ........
-
-2009-07-23 19:35 +0000 [r208389] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 208388 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r208388 | mmichelson | 2009-07-23 14:34:49 -0500 (Thu, 23 Jul
- 2009) | 24 lines Merged revisions 208386 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r208386 | mmichelson | 2009-07-23 14:24:21 -0500 (Thu, 23 Jul
- 2009) | 17 lines Fix a problem where a 491 response could be sent
- out of dialog. This generalizes the fix for issue 13849. The
- initial fix corrected the problem that Asterisk would reply with
- a 491 if a reinvite were received from an endpoint and we had not
- yet received an ACK from that endpoint for the initial INVITE it
- had sent us. This expansion also allows Asterisk to appropriately
- handle an INVITE with authorization credentials if Asterisk had
- not received an ACK from the previous transaction in which
- Asterisk had responded to an unauthorized INVITE with a 407.
- (closes issue #14239) Reported by: klaus3000 Patches: 14239.patch
- uploaded by mmichelson (license 60) Tested by: klaus3000 ........
- ................
-
-2009-07-23 19:23 +0000 [r208384] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 208383 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r208383 | jpeeler | 2009-07-23 14:21:50 -0500
- (Thu, 23 Jul 2009) | 12 lines Merged revisions 208380 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r208380 | jpeeler | 2009-07-23 14:19:53 -0500 (Thu, 23 Jul 2009)
- | 6 lines Only set the priindication setting when not performing
- a reload (closes issue #14696) Reported by: fdecher ........
- ................
-
-2009-07-23 16:30 +0000 [r208264-208316] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 208314 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r208314 | mmichelson | 2009-07-23 11:29:37 -0500 (Thu, 23 Jul
- 2009) | 9 lines Merged revisions 208312 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r208312 | mmichelson | 2009-07-23 11:29:18 -0500 (Thu, 23 Jul
- 2009) | 3 lines Remove inaccurate XXX comment. ........
- ................
-
- * /, channels/chan_sip.c: Merged revisions 208263 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r208263 | mmichelson | 2009-07-23 10:46:34 -0500 (Thu, 23 Jul
- 2009) | 15 lines Merged revisions 208262 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r208262 | mmichelson | 2009-07-23 10:43:07 -0500 (Thu, 23 Jul
- 2009) | 8 lines Properly handle 183 responses which do not
- contain an SDP. (closes issue #15442) Reported by: ffloimair
- Patches: 15442.patch uploaded by mmichelson (license 60) Tested
- by: tkarl, ffloimair ........ ................
-
-2009-07-21 22:47 +0000 [r207947] Tilghman Lesher <tlesher@digium.com>
-
- * /, funcs/func_strings.c: Merged revisions 207946 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r207946 | tilghman | 2009-07-21 17:45:32 -0500
- (Tue, 21 Jul 2009) | 15 lines Merged revisions 207945 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r207945 | tilghman | 2009-07-21 17:38:54 -0500 (Tue, 21 Jul 2009)
- | 8 lines Force an error if a blank is passed to QUOTE (because
- the documentation states the argument is not optional). This
- change makes URIENCODE and QUOTE behave similarly, since the
- documentation states that the argument is not optional, for both.
- (closes issue #15439) Reported by: pkempgen Patches:
- 20090706__issue15439.diff.txt uploaded by tilghman (license 14)
- ........ ................
-
-2009-07-21 20:27 +0000 [r207783-207860] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 207854 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r207854 | jpeeler | 2009-07-21 15:26:02 -0500
- (Tue, 21 Jul 2009) | 16 lines Merged revisions 207827 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r207827 | jpeeler | 2009-07-21 15:16:55 -0500 (Tue, 21 Jul 2009)
- | 9 lines Wait for wink before dialing when using E&M wink
- signaling There was already code for other signaling types in
- dahdi_handle_event to handle dialing if a dial operation dial
- string was present. Simply add SIG_EMWINK to the list. (closes
- issue #14434) Reported by: araasch ........ ................
-
- * channels/chan_dahdi.c: Revert r207636, this approach could
- potentially block for an unacceptable amount of time.
-
-2009-07-21 14:30 +0000 [r207725] Mark Michelson <mmichelson@digium.com>
-
- * main/manager.c, /: Merged revisions 207723 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r207723 | mmichelson | 2009-07-21 09:29:40 -0500 (Tue, 21 Jul
- 2009) | 11 lines Merged revisions 207714 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r207714 | mmichelson | 2009-07-21 09:26:00 -0500 (Tue, 21 Jul
- 2009) | 5 lines Document default timeout for AMI originations.
- AST-224 ........ ................
-
-2009-07-21 13:39 +0000 [r207683] Kevin P. Fleming <kpfleming@digium.com>
-
- * funcs/Makefile, codecs/lpc10/Makefile, main/db1-ast/Makefile,
- Makefile, agi/Makefile, codecs/Makefile, utils/Makefile, /,
- main/Makefile, codecs/gsm/Makefile, Makefile.moddir_rules,
- Makefile.rules, pbx/Makefile, res/Makefile, channels/Makefile:
- Merged revisions 207680 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r207680 | kpfleming | 2009-07-21 08:28:04 -0500 (Tue, 21 Jul
- 2009) | 18 lines Merged revisions 207647 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r207647 | kpfleming | 2009-07-21 08:04:44 -0500 (Tue, 21 Jul
- 2009) | 12 lines Ensure that user-provided CFLAGS and LDFLAGS are
- honored. This commit changes the build system so that
- user-provided flags (in ASTCFLAGS and ASTLDFLAGS) are supplied to
- the compiler/linker *after* all flags provided by the build
- system itself, so that the user can effectively override the
- build system's flags if desired. In addition, ASTCFLAGS and
- ASTLDFLAGS can now be provided *either* in the environment before
- running 'make', or as variable assignments on the 'make' command
- line. As a result, the use of COPTS and LDOPTS is no longer
- necessary, so they are no longer documented, but are still
- supported so as not to break existing build systems that supply
- them when building Asterisk. ........ ................
-
-2009-07-21 04:38 +0000 [r207636] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c: Wait for wink before dialing when using
- E&M wink signaling This patch adds a new dahdi_wait function to
- specifically wait for the wink event. If the wink is not
- eventually received the channel is hung up. (closes issue #14434)
- Reported by: araasch Patches: emwinkmod uploaded by araasch
- (license 693)
-
-2009-07-20 19:55 +0000 [r207425] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 207424 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r207424 | mmichelson | 2009-07-20 14:48:12 -0500 (Mon, 20 Jul
- 2009) | 39 lines Merged revisions 207423 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r207423 | mmichelson | 2009-07-20 14:39:59 -0500 (Mon, 20 Jul
- 2009) | 33 lines Answer video SDP offers properly when
- videosupport is not enabled. Copied from Review board: In issue
- 12434, the reporter describes a situation in which audio and
- video is offered on the call, but because videosupport is
- disabled in sip.conf, Asterisk gives no response at all to the
- video offer. According to RFC 3264, all media offers should have
- a corresponding answer. For offers we do not intend to actually
- reply to with meaningful values, we should still reply with the
- port for the media stream set to 0. In this patch, we take note
- of what types of media have been offered and save the information
- on the sip_pvt. The SDP in the response will take into account
- whether media was offered. If we are not otherwise going to
- answer a media offer, we will insert an appropriate m= line with
- the port set to 0. It is important to note that this patch is
- pretty much a bandage being applied to a broken bone. The patch
- *only* helps for situations where video is offered but
- videosupport is disabled and when udptl_pt is disabled but T.38
- is offered. Asterisk is not guaranteed to respond to every media
- offer. Notable cases are when multiple streams of the same type
- are offered. The 2 media stream limit is still present with this
- patch, too. In trunk and the 1.6.X branches, things will be a bit
- different since Asterisk also supports text in SDPs as well.
- (closes issue #12434) Reported by: mnnojd Review:
- https://reviewboard.asterisk.org/r/311 Review:
- https://reviewboard.asterisk.org/r/313 ........ ................
-
-2009-07-20 16:37 +0000 [r207362] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /: Merged revisions 207361 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r207361 | russell | 2009-07-20 11:36:15 -0500 (Mon, 20 Jul 2009)
- | 16 lines Merged revisions 207360 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r207360 | russell | 2009-07-20 11:26:24 -0500 (Mon, 20 Jul 2009)
- | 9 lines Only do the chan->fdno check in ast_read() in a
- developer build. I changed this check to only happen in a
- dev-mode build. I also added a comment explaining what is going
- on. I also made it so that detection of this situation does not
- affect ast_read() operation. (closes issue #14723) Reported by:
- seadweller ........ ................
-
-2009-07-18 01:35 +0000 [r207286] Richard Mudgett <rmudgett@digium.com>
-
- * channels/misdn/isdn_lib.c, channels/misdn_config.c,
- channels/misdn/isdn_lib_intern.h, channels/misdn/isdn_lib.h,
- doc/tex/misdn.tex, channels/chan_misdn.c, main/callerid.c,
- configs/misdn.conf.sample: Merged revisions 145293,158010 from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 to make
- merging easier. These changes are already on trunk.
- ................ r145293 | rmudgett | 2008-09-30 18:55:24 -0500
- (Tue, 30 Sep 2008) | 54 lines channels/chan_misdn.c
- channels/misdn/isdn_lib.c * Miscellaneous other fixes from trunk
- to make merging easier later. ........ r145200 | rmudgett |
- 2008-09-30 16:00:54 -0500 (Tue, 30 Sep 2008) | 7 lines *
- Miscellaneous formatting changes to make v1.4 and trunk more
- merge compatible in the mISDN area. channels/chan_misdn.c *
- Eliminated redundant code in cb_events() EVENT_SETUP ........
- r144257 | crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008)
- | 9 lines improved helptext of misdn_set_opt. ........ r142181 |
- rmudgett | 2008-09-09 12:30:52 -0500 (Tue, 09 Sep 2008) | 1 line
- Cleaned up comment ........ r138738 | rmudgett | 2008-08-18
- 16:07:28 -0500 (Mon, 18 Aug 2008) | 30 lines
- channels/chan_misdn.c * Made bearer2str() use
- allowed_bearers_array[] * Made use the causes.h defines instead
- of hardcoded numbers. * Made use Asterisk presentation indicator
- values if either of the mISDN presentation or screen options are
- negative. * Updated the misdn_set_opt application option
- descriptions. * Renamed the awkward Caller ID presentation
- misdn_set_opt application option value not_screened to
- restricted. Deprecated the not_screened option value.
- channels/misdn/isdn_lib.c * Made use the causes.h defines instead
- of hardcoded numbers. * Fixed some spelling errors and typos. *
- Added all defined facility code strings to fac2str().
- channels/misdn/isdn_lib.h * Added doxygen comments to struct
- misdn_bchannel. channels/misdn/isdn_lib_intern.h * Added doxygen
- comments to struct misdn_stack. channels/misdn_config.c
- configs/misdn.conf.sample * Updated the mISDN presentation and
- screen parameter descriptions. doc/misdn.txt (doc/tex/misdn.tex)
- * Updated the misdn_set_opt application option descriptions. *
- Fixed some spelling errors and typos. ................ r158010 |
- rmudgett | 2008-11-19 19:46:09 -0600 (Wed, 19 Nov 2008) | 9 lines
- 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.
- ................
-
-2009-07-17 19:38 +0000 [r207097-207157] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 207156 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r207156 | jpeeler | 2009-07-17 14:37:38 -0500
- (Fri, 17 Jul 2009) | 14 lines Merged revisions 207155 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r207155 | jpeeler | 2009-07-17 14:36:19 -0500 (Fri, 17 Jul 2009)
- | 7 lines Fix format specifier to print out an unsigned long
- long. Yep, it's even ifdefed out code. But it made it to the RR
- list... (closes issue #14726) Reported by: lmadsen ........
- ................
-
- * configs/chan_dahdi.conf.sample, /: Merged revisions 207095 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r207095 | jpeeler | 2009-07-17 14:16:35 -0500 (Fri, 17
- Jul 2009) | 2 lines Update some missing allowed options for
- overlapdial ........
-
-2009-07-17 17:53 +0000 [r206871-207032] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 207029 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r207029 |
- dvossel | 2009-07-17 12:51:44 -0500 (Fri, 17 Jul 2009) | 6 lines
- sip option flags handled incorrectly (closes issue #15376)
- Reported by: Takehiko Ooshima Tested by: dvossel,
- Takehiko_Ooshima ........
-
- * /, channels/chan_sip.c: Merged revisions 206939 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r206939 | dvossel | 2009-07-17 11:13:22 -0500 (Fri, 17 Jul 2009)
- | 20 lines Merged revisions 206938 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r206938 | dvossel | 2009-07-17 11:05:06 -0500 (Fri, 17 Jul 2009)
- | 14 lines SIP incorrect From: header information when callpres
- is prohib Some ITSP make use of the "Anonymous" display name to
- detect a requirement to withhold caller id across the PSTN. This
- does not work if the display name is "Unknown". (closes issue
- #14465) Reported by: Nick_Lewis Patches:
- chan_sip.c-callerpres.patch uploaded by Nick (license 657)
- chan_sip.c-callerpres_trunk.patch uploaded by dvossel (license
- 671) Tested by: Nick_Lewis, dvossel ........ ................
-
- * configs/iax.conf.sample, /: Merged revisions 206873 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r206873 | dvossel | 2009-07-16 16:33:51 -0500
- (Thu, 16 Jul 2009) | 12 lines Merged revisions 206872 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r206872 | dvossel | 2009-07-16 16:33:19 -0500 (Thu, 16 Jul 2009)
- | 6 lines error in iax.conf related IP-based access control
- (closes issue #15518) Reported by: pkempgen ........
- ................
-
- * /, main/callerid.c: Merged revisions 206868 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r206868 | dvossel | 2009-07-16 16:25:22 -0500 (Thu, 16 Jul 2009)
- | 14 lines Merged revisions 206867 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r206867 | dvossel | 2009-07-16 16:24:16 -0500 (Thu, 16 Jul 2009)
- | 8 lines avoid segfault caused by user error If the CALLERPRES()
- dialplan function is set to nothing, a segfault occurs. This is
- user error to begin with, but I'd rather see a cli warning
- message than have Asterisk crash on me. ........ ................
-
-2009-07-16 16:52 +0000 [r206809] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_realtime.c, /: Merged revisions 206808 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r206808 | tilghman | 2009-07-16 11:51:05 -0500
- (Thu, 16 Jul 2009) | 13 lines Merged revisions 206807 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r206807 | tilghman | 2009-07-16 11:27:35 -0500 (Thu, 16 Jul 2009)
- | 6 lines Fix a memory leak. (closes issue #15517) Reported by:
- adomjan Patches: func_realtime.c-ast_variable_destroy.diff
- uploaded by adomjan (license 487) ........ ................
-
-2009-07-15 22:06 +0000 [r206775] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 206768 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r206768 |
- dvossel | 2009-07-15 17:04:13 -0500 (Wed, 15 Jul 2009) | 8 lines
- Session timer were not activated if Supported header field in
- INVITE had both "timer" and other options. (closes issue #15403)
- Reported by: makoto Patches: sip-session-timer.patch uploaded by
- makoto (license ........
-
-2009-07-15 21:34 +0000 [r206762] Richard Mudgett <rmudgett@digium.com>
-
- * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h, /:
- Merged revisions 206707 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r206707 | rmudgett | 2009-07-15 16:14:41 -0500 (Wed, 15 Jul 2009)
- | 33 lines Merged revisions 206706 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r206706 | rmudgett | 2009-07-15 15:44:55 -0500
- (Wed, 15 Jul 2009) | 26 lines Merged revision 206700 from
- https://origsvn.digium.com/svn/asterisk/be/branches/C.2-...
- .......... Fixed chan_misdn crash because mISDNuser library is
- not thread safe. With Asterisk the mISDNuser library is driven by
- two threads concurrently: 1.
- channels/misdn/isdn_lib.c::manager_event_handler() 2.
- channels/misdn/isdn_lib.c::misdn_lib_isdn_event_catcher() Calls
- into the library are done concurrently and recursively from
- isdn_lib.c. Both threads can fiddle with the master/child
- layer3_proc_t lists. One thread may traverse the list when the
- other interrupts it and then removes the list element which the
- first thread was currently handling. This is exactly what caused
- the crash. About 60 calls were needed to a Gigaset CX475 before
- it occurred once. This patch adds locking when calling into the
- mISDNuser library. This also fixes some cb_log calls with wrong
- port parameter. JIRA ABE-1913 Patches: misdn-locking.patch
- (Modified with mostly cosmetic changes) ..........
- ................ ................
-
-2009-07-15 20:21 +0000 [r206705] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 206702 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r206702 |
- dvossel | 2009-07-15 15:20:01 -0500 (Wed, 15 Jul 2009) | 10 lines
- callerid(num) is wrong when username is missing A domain only sip
- uri <sip:123.123.123.123> would return 123.123.123.123 as callid
- num. Now, if the username is missing from a uri, the callerid num
- field is left empty. (closes issue #15476) Reported by: viraptor
- ........
-
-2009-07-15 16:02 +0000 [r206637] Sean Bright <sean@malleable.com>
-
- * /, codecs/codec_dahdi.c: Merged revisions 206636 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r206636 | seanbright | 2009-07-15 12:00:24 -0400
- (Wed, 15 Jul 2009) | 9 lines Merged revisions 206635 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r206635 | seanbright | 2009-07-15 11:57:51 -0400 (Wed,
- 15 Jul 2009) | 1 line Only print debug info in codec_dahdi if we
- are asking for it. ........ ................
-
-2009-07-14 20:22 +0000 [r206585] Tilghman Lesher <tlesher@digium.com>
-
- * /, contrib/scripts/meetme.sql: Recorded merge of revisions 206567
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r206567 | tilghman | 2009-07-14 15:14:45 -0500 (Tue, 14
- Jul 2009) | 6 lines Document all meetme realtime fields, and in
- the process, make some field lengths more consistent. (closes
- issue #15493) Reported by: lasko Patches: meetme.diff uploaded by
- lasko (license 833) ........
-
-2009-07-14 18:17 +0000 [r206555] Richard Mudgett <rmudgett@digium.com>
-
- * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
- channels/chan_misdn.c, /: Merged revisions 206489 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r206489 | rmudgett | 2009-07-14 12:01:48 -0500
- (Tue, 14 Jul 2009) | 35 lines Merged revisions 206487 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r206487 | rmudgett | 2009-07-14 11:44:47 -0500 (Tue, 14 Jul 2009)
- | 28 lines Fixes several call transfer issues with chan_misdn. *
- issue #14355 - Crash if attempt to transfer a call to an
- application. Masquerade the other pair of the four asterisk
- channels involved in the two calls. The held call already must be
- a bridged call (not an applicaton) or it would have been
- rejected. * issue #14692 - Held calls are not automatically
- cleared after transfer. Allow the core to initate disconnect of
- held calls to the ISDN port. This also fixes a similar case where
- the party on hold hangs up before being transferred or taken off
- hold. * JIRA ABE-1903 - Orphaned held calls left in
- music-on-hold. Do not simply block passing the hangup event on
- held calls to asterisk core. * Fixed to allow held calls to be
- transferred to ringing calls. Previously, held calls could only
- be transferred to connected calls. * Eliminated unused call
- states to simplify hangup code. * Eliminated most uses of
- "holded" because it is not a word. (closes issue #14355) (closes
- issue #14692) Reported by: sodom Patches:
- misdn_xfer_v14_r205839.patch uploaded by rmudgett (license 664)
- Tested by: rmudgett ........ ................
-
-2009-07-14 14:54 +0000 [r206387] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 206386 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r206386 | russell | 2009-07-14 09:51:44 -0500
- (Tue, 14 Jul 2009) | 20 lines Merged revisions 206385 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r206385 | russell | 2009-07-14 09:48:00 -0500
- (Tue, 14 Jul 2009) | 13 lines Merged revisions 206384 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
- r206384 | russell | 2009-07-14 09:45:47 -0500 (Tue, 14 Jul 2009)
- | 6 lines Ensure apathetic replies are sent out on the proper
- socket. chan_iax2 supports multiple address bindings. The
- send_apathetic_reply() function did not attempt to send its
- response on the same socket that the incoming message came in on.
- ........ ................ ................
-
-2009-07-14 01:25 +0000 [r206369] Richard Mudgett <rmudgett@digium.com>
-
- * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
- revisions 206341 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r206341 | rmudgett | 2009-07-13 19:48:59 -0500 (Mon, 13 Jul 2009)
- | 11 lines Merged revisions 206284 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r206284 | rmudgett | 2009-07-13 19:17:28 -0500 (Mon, 13 Jul 2009)
- | 4 lines Fix some memory leaks in chan_misdn. JIRA ABE-1911
- ........ ................
-
-2009-07-10 22:50 +0000 [r206017] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 205985 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r205985 |
- dvossel | 2009-07-10 16:42:10 -0500 (Fri, 10 Jul 2009) | 16 lines
- SIP register not using peer's outbound proxy If callbackextension
- is defined for a peer it successfully causes a registration to
- occur, but the registration ignores the outboundproxy settings
- for the peer. This patch allows the peer to be passed to
- obproxy_get() in transmit_register(). (closes issue #14344)
- Reported by: Nick_Lewis Patches:
- callbackextension_peer_trunk.diff uploaded by dvossel (license
- 671) Tested by: dvossel Review:
- https://reviewboard.asterisk.org/r/294/ ........
-
-2009-07-10 18:44 +0000 [r205940] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/udptl.c, /: Merged revisions 205939 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r205939 |
- kpfleming | 2009-07-10 13:44:09 -0500 (Fri, 10 Jul 2009) | 1 line
- Update comments about the level of T.38 support in Asterisk.
- ........
-
-2009-07-10 17:44 +0000 [r205879-205880] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: Fix build.
-
- * /, channels/chan_sip.c: Merged revisions 205878 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r205878 | mmichelson | 2009-07-10 12:39:57 -0500 (Fri, 10 Jul
- 2009) | 30 lines Merged revisions 205877 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r205877 | mmichelson | 2009-07-10 12:39:13 -0500
- (Fri, 10 Jul 2009) | 23 lines Merged revisions 205776 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r205776 | mmichelson | 2009-07-10 10:56:45 -0500
- (Fri, 10 Jul 2009) | 16 lines Merged revisions 205775 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul
- 2009) | 10 lines Ensure that outbound NOTIFY requests are
- properly routed through stateful proxies. With this change, we
- make note of Record-Route headers present in any SUBSCRIBE
- request that we receive so that our outbound NOTIFY requests will
- have the proper Route headers in them. (closes issue #14725)
- Reported by: ibc ........ ................ ................
- ................
-
-2009-07-10 16:48 +0000 [r205843] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 205840 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r205840 | dvossel | 2009-07-10 11:42:04 -0500 (Fri, 10 Jul 2009)
- | 37 lines Merged revisions 205804 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r205804 | dvossel | 2009-07-10 11:23:59 -0500 (Fri, 10 Jul 2009)
- | 31 lines SIP registration auth loop caused by stale nonce If an
- endpoint sends two registration requests in a very short period
- of time with the same nonce, both receive 401 responses from
- Asterisk, each with a different nonce (the second 401 containing
- the current nonce and the first one being stale). If the endpoint
- responds to the first 401, it does not match the current nonce so
- Asterisk sends a third 401 with a newly generated nonce (which
- updates the current nonce)... Now if the endpoint responds to the
- second 401, it does not match the current nonce either and
- Asterisk sends a fourth 401 with a newly generated nonce... This
- loop goes on and on. There appears to be a simple fix for this.
- If the nonce from the request does not match our nonce, but is a
- good response to a previous nonce, instead of sending a 401 with
- a newly generated nonce, use the current one instead. This breaks
- the loop as the nonce is not updated until a response is
- received. Additional logic has been added to make sure no nonce
- can be responded to twice though. (closes issue #15102) Reported
- by: Jamuel Patches: patch-bug_0015102 uploaded by Jamuel (license
- 809) nonce_sip.diff uploaded by dvossel (license 671) Tested by:
- Jamuel Review: https://reviewboard.asterisk.org/r/289/ ........
- ................
-
-2009-07-10 15:57 +0000 [r205777] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 205776 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r205776 | mmichelson | 2009-07-10 10:56:45 -0500 (Fri, 10 Jul
- 2009) | 16 lines Merged revisions 205775 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul
- 2009) | 10 lines Ensure that outbound NOTIFY requests are
- properly routed through stateful proxies. With this change, we
- make note of Record-Route headers present in any SUBSCRIBE
- request that we receive so that our outbound NOTIFY requests will
- have the proper Route headers in them. (closes issue #14725)
- Reported by: ibc ........ ................
-
-2009-07-10 15:35 +0000 [r205771] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, apps/app_fax.c: Merged revisions 205770 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r205770 |
- kpfleming | 2009-07-10 10:28:11 -0500 (Fri, 10 Jul 2009) | 12
- lines Fix some remaining T.38 negotiation problems in app_fax.
- Revision 205696 did not quite fix all the issues with the T.38
- negotiation changes and app_fax; this patch corrects them, along
- with a couple of other minor issues. (closes issue #15480)
- Reported by: dimas Patches: test2-15480.patch uploaded by dimas
- (license 88) ........
-
-2009-07-09 23:46 +0000 [r205729] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_dahdi.c: Merged revisions 205728 via svn merge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r205728 | rmudgett | 2009-07-09 18:37:53 -0500 (Thu, 09 Jul 2009)
- | 21 lines No audio on calls from Asterisk to various ISDN
- devices until DTMF sent by caller. Add missing clearing of the
- dialing flag when the ISDN call is CONNECTED. (i.e. When libpri
- generates the event PRI_EVENT_ANSWER.) (closes issue #15420)
- Reported by: scottbmilne Patches: bug15420-1.4.25.1-diff2.txt
- uploaded by alecdavis (license 585) Tested by: scottbmilne,
- alecdavis (closes issue #15416) Reported by: avinoash (closes
- issue #15389) Reported by: alecdavis This patch should also fix
- the following issue: (issue #15205) Reported by: vinsik ........
-
-2009-07-09 21:26 +0000 [r205697] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, channels/chan_sip.c, apps/app_fax.c, include/asterisk/frame.h:
- Merged revisions 205696 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r205696 |
- kpfleming | 2009-07-09 16:20:23 -0500 (Thu, 09 Jul 2009) | 16
- lines Repair ability of SendFAX/ReceiveFAX to respond to T.38
- switchover. Recent changes in T.38 negotiation in Asterisk caused
- these applications to not respond when the other endpoint
- initiated a switchover to T.38; this resulted in the T.38
- switchover failing, and the FAX attempt to be made using an audio
- connection, instead of T.38 (which would usually cause the FAX to
- fail completely). This patch corrects this problem, and the
- applications will now correctly respond to the T.38 switchover
- request. In addition, the response will include the appopriate
- T.38 session parameters based on what the other end offered and
- what our end is capable of. (closes issue #14849) Reported by:
- afosorio ........
-
-2009-07-09 16:21 +0000 [r205597-205608] David Vossel <dvossel@digium.com>
-
- * include/asterisk/time.h, /: Merged revisions 205600 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r205600 | dvossel | 2009-07-09 11:19:09 -0500
- (Thu, 09 Jul 2009) | 9 lines Merged revisions 205599 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r205599 | dvossel | 2009-07-09 11:18:09 -0500 (Thu, 09
- Jul 2009) | 2 lines Changing ast_samp2tv to not use floating
- point. ........ ................
-
- * main/rtp.c, /, channels/chan_iax2.c, include/asterisk/frame.h:
- Merged revisions 205479 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r205479 | dvossel | 2009-07-08 18:19:09 -0500 (Wed, 08 Jul 2009)
- | 16 lines Merged revisions 205471 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r205471 | dvossel | 2009-07-08 18:15:54 -0500 (Wed, 08 Jul 2009)
- | 10 lines Fixes 8khz assumptions Many calculations assume 8khz
- is the codec rate. This is not always the case. This patch only
- addresses chan_iax.c and res_rtp_asterisk.c, but I am sure there
- are other areas that make this assumption as well. Review:
- https://reviewboard.asterisk.org/r/306/ ........ ................
-
-2009-07-09 08:32 +0000 [r205533] Michiel van Baak <michiel@vanbaak.info>
-
- * /, main/ssl.c: Merged revisions 205532 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r205532 |
- mvanbaak | 2009-07-09 10:31:24 +0200 (Thu, 09 Jul 2009) | 5 lines
- pthread_self returns a pthread_t which is not an unsigned int on
- all pthread implementations. Casting it to an unsigned int fixes
- compiler warnings. Tested on OpenBSD and Linux both 32 and 64 bit
- ........
-
-2009-07-08 22:17 +0000 [r205415] David Vossel <dvossel@digium.com>
-
- * include/asterisk/devicestate.h, main/pbx.c, /,
- main/devicestate.c, include/asterisk/pbx.h: Merged revisions
- 205412 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r205412 | dvossel | 2009-07-08 17:15:06 -0500 (Wed, 08 Jul 2009)
- | 12 lines Merged revisions 205409 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r205409 | dvossel | 2009-07-08 16:35:12 -0500 (Wed, 08 Jul 2009)
- | 6 lines moving ast_devstate_to_extenstate to pbx.c from
- devicestate.c ast_devstate_to_extenstate belongs in pbx.c. This
- change fixes a compile time error with chan_vpb as well. ........
- ................
-
-2009-07-08 19:27 +0000 [r205351] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 205350 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r205350 | mmichelson | 2009-07-08 14:26:55 -0500 (Wed, 08 Jul
- 2009) | 20 lines Merged revisions 205349 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r205349 | mmichelson | 2009-07-08 14:26:13 -0500 (Wed, 08 Jul
- 2009) | 14 lines Prevent phantom calls to queue members. If a
- caller were to hang up while a periodic announcement or position
- were being said, the return value for those functions would
- incorrectly indicate that the caller was still in the queue. With
- these changes, the problem does not occur. (closes issue #14631)
- Reported by: latinsud Patches: queue_announce_ghost_call2.diff
- uploaded by latinsud (license 745) (with small modification from
- me) ........ ................
-
-2009-07-08 18:20 +0000 [r205296] Jason Parker <jparker@digium.com>
-
- * config.guess, config.sub, /: Merged revisions 205291 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r205291 | qwell | 2009-07-08 13:19:46 -0500
- (Wed, 08 Jul 2009) | 9 lines Merged revisions 205288 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r205288 | qwell | 2009-07-08 13:19:03 -0500 (Wed, 08 Jul
- 2009) | 1 line Update config.guess and config.sub from the
- savannah.gnu.org git repo. ........ ................
-
-2009-07-08 17:01 +0000 [r205224] Tilghman Lesher <tlesher@digium.com>
-
- * main/say.c: oops, fixing build
-
-2009-07-08 16:56 +0000 [r205220] David Vossel <dvossel@digium.com>
-
- * include/asterisk/time.h, /: Merged revisions 205216 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r205216 | dvossel | 2009-07-08 11:54:24 -0500
- (Wed, 08 Jul 2009) | 17 lines Merged revisions 205215 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r205215 | dvossel | 2009-07-08 11:53:40 -0500 (Wed, 08 Jul 2009)
- | 10 lines ast_samp2tv needs floating point for 16khz audio In
- ast_samp2tv(), (1000000 / _rate) = 62.5 when _rate is 16000. The
- .5 is currently stripped off because we don't calculate using
- floating points. This causes madness with 16khz audio. (issue
- ABE-1899) Review: https://reviewboard.asterisk.org/r/305/
- ........ ................
-
-2009-07-08 16:28 +0000 [r205200] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/say.c: Merged revisions 205196 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r205196 | tilghman | 2009-07-08 11:27:50 -0500 (Wed, 08 Jul 2009)
- | 9 lines Merged revisions 205188 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r205188 | tilghman | 2009-07-08 11:26:15 -0500 (Wed, 08 Jul 2009)
- | 2 lines Add redirection warnings for the invalid language codes
- previously removed. ........ ................
-
-2009-07-08 15:56 +0000 [r205139-205152] Russell Bryant <russell@digium.com>
-
- * /, main/ssl.c: Merged revisions 205151 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r205151 |
- russell | 2009-07-08 10:56:28 -0500 (Wed, 08 Jul 2009) | 2 lines
- Use tabs instead of spaces for indentation. ........
-
- * main/asterisk.c, /, main/Makefile, res/res_crypto.c, main/ssl.c
- (added), include/asterisk/_private.h, res/res_jabber.c: Merged
- revisions 205120 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r205120 |
- russell | 2009-07-08 10:17:19 -0500 (Wed, 08 Jul 2009) | 16 lines
- Move OpenSSL initialization to a single place, make library usage
- thread-safe. While doing some reading about OpenSSL, I noticed a
- couple of things that needed to be improved with our usage of
- OpenSSL. 1) We had initialization of the library done in multiple
- modules. This has now been moved to a core function that gets
- executed during Asterisk startup. We already link OpenSSL into
- the core for TCP/TLS functionality, so this was the most logical
- place to do it. 2) OpenSSL is not thread-safe by default.
- However, making it thread safe is very easy. We just have to
- provide a couple of callbacks. One callback returns a thread ID.
- The other handles locking. For more information, start with the
- "Is OpenSSL thread-safe?" question on the FAQ page of
- openssl.org. ........
-
-2009-07-08 14:35 +0000 [r205117] David Vossel <dvossel@digium.com>
-
- * channels/chan_sip.c, include/asterisk/sched.h: SIP Dialog ref
- counting This patch adds reference counting for sip dialogs into
- 1.6.0. When proc_session_timer() is called from the scheduler
- thread it has no guarantee the session timer's dialog won't be
- freed from underneath it. Now the session timer holds a reference
- to the dialog, preventing it from being destroyed during the
- middle of proc_session_timer(). (closes issue #13623) Reported
- by: Nik Soggia Review: https://reviewboard.asterisk.org/r/302/
-
-2009-07-06 15:17 +0000 [r204980] Tilghman Lesher <tlesher@digium.com>
-
- * main/say.c: Restore Hungarian (mistakenly removed during merge)
-
-2009-07-06 13:39 +0000 [r204949] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/channel.c, /: Merged revisions 204948 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r204948 |
- kpfleming | 2009-07-06 08:38:29 -0500 (Mon, 06 Jul 2009) | 7
- lines Improve handling of AST_CONTROL_T38 and
- AST_CONTROL_T38_PARAMETERS for non-T.38-capable channels. This
- change allows applications that request T.38 negotiation on a
- channel that does not support it to get the proper indication
- that it is not supported, rather than thinking that negotiation
- was started when it was not. ........
-
-2009-07-02 22:03 +0000 [r204836] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_misdn.c, /: Merged revisions 204835 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r204835 | rmudgett | 2009-07-02 17:01:28 -0500
- (Thu, 02 Jul 2009) | 17 lines Merged revisions 204834 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r204834 | rmudgett | 2009-07-02 16:59:43 -0500 (Thu, 02 Jul 2009)
- | 10 lines Removed confusing warning message "Got Busy in
- Connected State" If an incoming mISDN call is answered with the
- Answer application and a subsequent Dial gets a busy endpoint
- then it is valid for that already connected channel to get the
- busy indication. Asterisk will play the busy tones until the
- dialplan plays something else or hangs up the call. (closes issue
- #11974) Reported by: fvdb ........ ................
-
-2009-07-02 18:07 +0000 [r204652-204754] David Vossel <dvossel@digium.com>
-
- * include/asterisk/devicestate.h, main/pbx.c, /,
- main/devicestate.c: Merged revisions 204710 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r204710 | dvossel | 2009-07-02 11:03:44 -0500 (Thu, 02 Jul 2009)
- | 21 lines Merged revisions 204681 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r204681 | dvossel | 2009-07-02 10:05:57 -0500 (Thu, 02 Jul 2009)
- | 14 lines Improved mapping of extension states from combined
- device states. This fixes a few issues with incorrect extension
- states and adds a cli command, core show device2extenstate, to
- display all possible state mappings. (closes issue #15413)
- Reported by: legart Patches: exten_helper.diff uploaded by
- dvossel (license 671) Tested by: dvossel, legart, amilcar Review:
- https://reviewboard.asterisk.org/r/301/ ........ ................
-
- * channels/chan_sip.c: removes fake dialog_unref and dialog_ref
- function calls. dialog_unref() and dialog_ref() in 1.6.0 where
- only place holders for reference counting once it was
- implemented. The functions did nothing but return the pointer on
- ref and NULL on unref. These calls have been removed to make way
- for a patch that actually does dialog ref counting.
-
-2009-06-30 21:21 +0000 [r204581] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/say.c, UPGRADE.txt: Merged revisions 204563 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r204563 | tilghman | 2009-06-30 15:41:04 -0500
- (Tue, 30 Jun 2009) | 13 lines Merged revisions 204556 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r204556 | tilghman | 2009-06-30 15:23:51 -0500 (Tue, 30 Jun 2009)
- | 6 lines More incorrect language codes, plus ensuring that
- regionalizations use the specified language, and not English for
- grammar. (closes issue #15022) Reported by: greenfieldtech
- Patches: 20090519__issue15022.diff.txt uploaded by tilghman
- (license 14) ........ ................
-
-2009-06-30 18:50 +0000 [r204476] Jason Parker <jparker@digium.com>
-
- * /, main/say.c: Merged revisions 204475 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r204475 | qwell | 2009-06-30 13:48:35 -0500 (Tue, 30 Jun 2009) |
- 9 lines Merged revisions 204474 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r204474 | qwell | 2009-06-30 13:47:06 -0500 (Tue, 30 Jun 2009) |
- 1 line Fix ast_say_counted_noun to correctly handle Polish. Fix a
- comment typo in passing. ........ ................
-
-2009-06-30 18:44 +0000 [r204471] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c, /, main/say.c, UPGRADE.txt: Recorded merge
- of revisions 204470 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r204470 | tilghman | 2009-06-30 13:36:24 -0500 (Tue, 30 Jun 2009)
- | 18 lines Recorded merge of revisions 204469 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r204469 | tilghman | 2009-06-30 13:23:35 -0500 (Tue, 30 Jun 2009)
- | 11 lines "tw" is the language specification for Twi (from
- Ghana) not Taiwanese. (closes issue #15346) Reported by: volivier
- Patches: 20090617__issue15346__1.4.diff.txt uploaded by tilghman
- (license 14) 20090617__issue15346__trunk.diff.txt uploaded by
- tilghman (license 14) 20090617__issue15346__1.6.0.diff.txt
- uploaded by tilghman (license 14)
- 20090617__issue15346__1.6.1.diff.txt uploaded by tilghman
- (license 14) 20090617__issue15346__1.6.2.diff.txt uploaded by
- tilghman (license 14) Tested by: volivier ........
- ................
-
-2009-06-29 22:52 +0000 [r204248-204302] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 204301 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r204301 | mmichelson | 2009-06-29 17:50:35 -0500 (Mon, 29 Jun
- 2009) | 15 lines Merged revisions 204300 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r204300 | mmichelson | 2009-06-29 17:45:34 -0500 (Mon, 29 Jun
- 2009) | 9 lines Add error message so that it is clear why a SIP
- peer was not processed when a DNS lookup fails on a host or
- outboundproxy. (closes issue #13432) Reported by: p_lindheimer
- Patches: outboundproxy.patch uploaded by p (license 558) ........
- ................
-
- * /, channels/chan_sip.c: Merged revisions 204247 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r204247 | mmichelson | 2009-06-29 16:48:54 -0500 (Mon, 29 Jun
- 2009) | 32 lines Merged revisions 204243,204246 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r204243 | mmichelson | 2009-06-29 16:23:43 -0500 (Mon, 29 Jun
- 2009) | 22 lines Fix a problem where chan_sip would ignore "old"
- but valid responses. chan_sip has had a problem for quite a long
- time that would manifest when Asterisk would send multiple SIP
- responses on the same dialog before receiving a response. The
- problem occurred because chan_sip only kept track of the highest
- outgoing sequence number used on the dialog. If Asterisk sent two
- requests out, and a response arrived for the first request sent,
- then Asterisk would ignore the response. The result was that
- Asterisk would continue retransmitting the requests and ignoring
- the responses until the maximum number of retransmissions had
- been reached. The fix here is to rearrange the code a bit so that
- instead of simply comparing the sequence number of the response
- to our latest outgoing sequence number, we walk our list of
- outstanding packets and determine if there is a match. If there
- is, we continue. If not, then we ignore the response. In doing
- this, I found a few completely useless variables that I have now
- removed. (closes issue #11231) Reported by: flefoll Review:
- https://reviewboard.asterisk.org/r/298 ........ r204246 |
- mmichelson | 2009-06-29 16:37:05 -0500 (Mon, 29 Jun 2009) | 3
- lines Fix build oops. ........ ................
-
-2009-06-27 01:14 +0000 [r203910] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 203909 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r203909 | rmudgett | 2009-06-26 20:07:52 -0500
- (Fri, 26 Jun 2009) | 23 lines Merged revisions 203908 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r203908 | rmudgett | 2009-06-26 19:55:12 -0500 (Fri, 26 Jun 2009)
- | 16 lines The ISDN CPE side should not exclusively pick B
- channels normally. Before this patch, Asterisk unconditionally
- picked B channels exclusively on the CPE side and normally
- allowed alternative B channels on the network side. Now Asterisk
- does the opposite. Reasons for the CPE side to normally not pick
- B channels exclusively: * For CPE point-to-multipoint mode (i.e.
- phone side), the CPE side does not have enough information to
- exclusively pick B channels. (There may be other devices on the
- line.) * Q.931 gives preference to the network side picking B
- channels. * Some telcos require the CPE side to not pick B
- channels exclusively. (closes issue #14383) Reported by:
- mbrancaleoni ........ ................
-
-2009-06-26 22:12 +0000 [r203855] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 203853 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r203853 | jpeeler | 2009-06-26 17:11:31 -0500
- (Fri, 26 Jun 2009) | 12 lines Merged revisions 203848 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r203848 | jpeeler | 2009-06-26 17:09:19 -0500 (Fri, 26 Jun 2009)
- | 5 lines Make sure to recreate the dahdi pseudo channel after
- dahdi restart (closes issue #14477) Reported by: timking ........
- ................
-
-2009-06-26 21:25 +0000 [r203780-203818] Russell Bryant <russell@digium.com>
-
- * /, main/file.c: Merged revisions 203802 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r203802 | russell | 2009-06-26 16:21:48 -0500 (Fri, 26 Jun 2009)
- | 22 lines Merged revisions 203785 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r203785 | russell | 2009-06-26 16:16:39 -0500 (Fri, 26 Jun 2009)
- | 15 lines Don't fast forward past the end of a message. This is
- nice change for users of the voicemail application. If someone
- gets a little carried away with fast forwarding through a
- message, they can easily get to the end and accidentally exit the
- voicemail application by hitting the fast forward key during the
- following prompt. This adds some safety by not allowing a fast
- forward past the end of a message. (closes issue #14554) Reported
- by: lacoursj Patches: 21761.patch uploaded by lacoursj (license
- 707) Tested by: lacoursj ........ ................
-
- * /, channels/chan_sip.c: Merged revisions 203779 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r203779 |
- russell | 2009-06-26 15:45:00 -0500 (Fri, 26 Jun 2009) | 5 lines
- Ensure the TCP read buffer is fully initialized before handling
- each packet. (closes issue #14452) Reported by: umberto71
- ........
-
-2009-06-26 20:16 +0000 [r203722] David Brooks <dbrooks@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 203721 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r203721 | dbrooks | 2009-06-26 15:13:51 -0500 (Fri, 26 Jun 2009)
- | 16 lines Fixing voicemail's error in checking max silence vs
- min message length Max silence was represented in milliseconds,
- yet vmminsecs (minmessage) was represented as seconds. Also, the
- inequality was reversed. The warning, if triggered, was "Max
- silence should be less than minmessage or you may get empty
- messages", which should have been logged if max silence was
- greater than minmessage, but the check was for less than. Also,
- conforming if statement to coding guidelines. closes issue
- #15331) Reported by: markd Review:
- https://reviewboard.asterisk.org/r/293/ ........
-
-2009-06-26 19:54 +0000 [r203717] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c: reverse whitespace change 203711 that was
- based on looking at sig_analog (which has about a 1000 line
- indentation change that is not worth doing here)
-
-2009-06-26 19:49 +0000 [r203716] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 203710 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r203710 | dvossel | 2009-06-26 14:47:11 -0500 (Fri, 26 Jun 2009)
- | 7 lines moving debug message from level 0 to 1. (closes issue
- #15404) Reported by: leobrown Patches: iax_codec_debug.patch
- uploaded by leobrown (license 541) ........
-
-2009-06-26 19:47 +0000 [r203711] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c: whitespace fix
-
-2009-06-26 19:29 +0000 [r203701] Joshua Colp <jcolp@digium.com>
-
- * main/rtp.c, main/channel.c, main/frame.c, /, channels/chan_sip.c,
- apps/app_fax.c, configs/sip.conf.sample,
- include/asterisk/frame.h: Merged revisions 203699 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r203699 | file | 2009-06-26 16:27:24 -0300 (Fri, 26 Jun 2009) | 2
- lines Improve T.38 negotiation by exchanging session parameters
- between application and channel. ........
-
-2009-06-26 19:25 +0000 [r203698] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 203672 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r203672 | jpeeler | 2009-06-26 14:03:25 -0500 (Fri, 26 Jun 2009)
- | 16 lines Check if polarityonanswerdelay has elapsed before
- setting a channel as answered after a polarity reversal.
- Previously on a polarity switch event chan_dahdi would set the
- channel immediately as answered. This would cause problems if a
- polarity reversal occurred when the line was picked up as the
- dial would not have yet occurred. Now if the polarity reversal
- occurs before delay has elapsed after coming off hook or an
- answer, it is ignored. Also, some refactoring was done in
- _handle_event. (closes issue #13917) Reported by: alecdavis
- Patches: chan_dahdi.bug13917.feb09.diff2.txt uploaded by
- alecdavis (license 585) Tested by: alecdavis ........
-
-2009-06-25 21:47 +0000 [r203447] David Vossel <dvossel@digium.com>
-
- * main/ast_expr2.fl, main/ast_expr2.c, /: Merged revisions 203444
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r203444 | dvossel | 2009-06-25 16:45:32 -0500 (Thu, 25
- Jun 2009) | 4 lines fixes a few redundant conditions (issue
- #15269) ........
-
-2009-06-25 21:18 +0000 [r203387] Terry Wilson <twilson@digium.com>
-
- * main/cli.c, /: Merged revisions 203381 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r203381 | twilson | 2009-06-25 16:15:11 -0500 (Thu, 25 Jun 2009)
- | 11 lines Merged revisions 203380 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r203380 | twilson | 2009-06-25 16:13:10 -0500 (Thu, 25 Jun 2009)
- | 4 lines I didn't see that Mark already fixed the underlying
- issue! Yay for removing useless code. ........ ................
-
-2009-06-25 21:06 +0000 [r203117-203377] Russell Bryant <russell@digium.com>
-
- * /, main/features.c: Merged revisions 203376 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r203376 | russell | 2009-06-25 16:04:55 -0500 (Thu, 25 Jun 2009)
- | 16 lines Merged revisions 203375 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r203375 | russell | 2009-06-25 16:02:18 -0500 (Thu, 25 Jun 2009)
- | 9 lines Fix a case where CDR answer time could be before the
- start time involving parking. (closes issue #13794) Reported by:
- davidw Patches: 13794.patch uploaded by murf (license 17)
- 13794.patch.160 uploaded by murf (license 17) Tested by: murf,
- dbrooks ........ ................
-
- * /, channels/chan_sip.c: Merged revisions 203116 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r203116 | russell | 2009-06-25 11:04:10 -0500 (Thu, 25 Jun 2009)
- | 18 lines Merged revisions 203115 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r203115 | russell | 2009-06-25 11:02:16 -0500 (Thu, 25 Jun 2009)
- | 11 lines Resolve a crash related to a T.38 reinvite race
- condition. This change resolves a crash observed locally during
- some T.38 testing. A call was set up using a call file, and when
- the T.38 reinvite came in, the channel state was still
- AST_STATE_DOWN. The reason is explained by a comment in the code
- that previously lived in the handling of AST_STATE_RINGING. This
- change modifies the logic to handle the same race condition for
- any channel state that is not UP. (closes ABE-1895) ........
- ................
-
-2009-06-24 21:18 +0000 [r203044] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 203037 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r203037 | rmudgett | 2009-06-24 16:08:55 -0500
- (Wed, 24 Jun 2009) | 15 lines Merged revisions 203036 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r203036 | rmudgett | 2009-06-24 16:01:43 -0500 (Wed, 24 Jun 2009)
- | 8 lines Improved chan_dahdi.conf pritimer error checking. Valid
- format is: pritimer=timer_name,timer_value * Fixed segfault if
- the ',' is missing. * Completely check the range returned by
- pri_timer2idx() to prevent possible access outside array bounds.
- ........ ................
-
-2009-06-24 18:29 +0000 [r202968] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 202967 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r202967 | mmichelson | 2009-06-24 13:29:10 -0500 (Wed, 24 Jun
- 2009) | 9 lines Merged revisions 202966 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r202966 | mmichelson | 2009-06-24 13:28:47 -0500 (Wed, 24 Jun
- 2009) | 3 lines Use the handy UNLINK macro instead of hand-coding
- the same thing in-line. ........ ................
-
-2009-06-24 18:09 +0000 [r202926] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 202925 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r202925 |
- file | 2009-06-24 15:08:17 -0300 (Wed, 24 Jun 2009) | 2 lines
- Ensure the default settings are applied for T.38 when we set it
- up for a peer. ........
-
-2009-06-23 22:09 +0000 [r202763] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 202761 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r202761 | mattf | 2009-06-23 17:08:43 -0500 (Tue, 23 Jun 2009) |
- 1 line I could have sworn I committed this patch ages ago, but...
- bug fix with setting NAI properly on linksets in certain
- situations. ........
-
-2009-06-23 21:26 +0000 [r202754] Ryan Brindley <rbrindley@digium.com>
-
- * main/config.c, /: Merged revisions 202753 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r202753 |
- rbrindley | 2009-06-23 16:25:17 -0500 (Tue, 23 Jun 2009) | 9
- lines If we delete the info, lets also delete the lines (closes
- issue #14509) Reported by: timeshell Patches:
- 20090504__bug14509.diff.txt uploaded by tilghman (license 14)
- Tested by: awk, timeshell ........
-
-2009-06-23 16:40 +0000 [r202675] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 202672 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r202672 | dvossel | 2009-06-23 11:31:30 -0500 (Tue, 23 Jun 2009)
- | 18 lines Merged revisions 202671 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r202671 | dvossel | 2009-06-23 11:28:46 -0500 (Tue, 23 Jun 2009)
- | 12 lines MWI NOTIFY contains a wrong URI if Asterisk listens to
- non-standard port and transport (closes issue #14659) Reported
- by: klaus3000 Patches: patch_chan_sip_fixMWIuri_1.4.txt uploaded
- by klaus3000 (license 65) mwi_port-transport_trunk.diff uploaded
- by dvossel (license 671) Tested by: dvossel, klaus3000 Review:
- https://reviewboard.asterisk.org/r/288/ ........ ................
-
-2009-06-22 20:12 +0000 [r202498] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /: Merged revisions 202497 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r202497 | russell | 2009-06-22 15:11:04 -0500 (Mon, 22 Jun 2009)
- | 11 lines Merged revisions 202496 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r202496 | russell | 2009-06-22 15:08:53 -0500 (Mon, 22 Jun 2009)
- | 4 lines Report CallerID change during a masquerade. Reported
- by: markster ........ ................
-
-2009-06-22 16:30 +0000 [r202471] Sean Bright <sean@malleable.com>
-
- * cdr/cdr_sqlite3_custom.c, /: Merged revisions 202417 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r202417 | seanbright | 2009-06-22 12:09:50 -0400 (Mon, 22 Jun
- 2009) | 4 lines Fix lock usage in cdr_sqlite3_custom to avoid
- potential crashes during reload. Pointed out by Russell while
- working on the CEL branch. ........
-
-2009-06-22 16:06 +0000 [r202416] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 202415 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r202415 | russell | 2009-06-22 11:05:08 -0500 (Mon, 22 Jun 2009)
- | 9 lines Merged revisions 202414 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r202414 | russell | 2009-06-22 11:00:00 -0500 (Mon, 22 Jun 2009)
- | 2 lines Make Polycom subscription type override check more
- explicit. ........ ................
-
-2009-06-22 15:05 +0000 [r202338-202344] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 202343 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r202343 | mmichelson | 2009-06-22 09:58:24 -0500 (Mon, 22 Jun
- 2009) | 36 lines Merged revisions 202341-202342 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r202341 | mmichelson | 2009-06-22 09:42:55 -0500 (Mon, 22 Jun
- 2009) | 26 lines Fix a situation in which Asterisk would not stop
- retransmitting 487s. If a CANCEL were received by Asterisk, we
- would send a 487 in response to the original INVITE and a 200 OK
- for the CANCEL. If there were a network hiccup which caused the
- 200 OK and the 487 to be lost, then the UA communicating with
- Asterisk may try to retransmit its CANCEL. Asterisk's response to
- this used to be to try sending another 487 to the canceled INVITE
- and another 200 OK to the CANCEL. The problem here is that the
- originally-sent 487 was sent "reliably" meaning that it will be
- retransmitted until it is received properly. So when we receive
- the second CANCEL it is likely that the first batch of 487s we
- sent is still going strong and reaches the UA. The result was
- that the second set of 487s would be retransmitted constantly
- until the maximum number of retries had been reached. The fix for
- this is that if we receive a second CANCEL for an INVITE, then we
- cancel the retransmission of the first set of 487s and start a
- second set. This causes the dialog to be terminated reasonably.
- (closes issue #14584) Reported by: klaus3000 Patches:
- 14584_v2.patch uploaded by mmichelson (license 60) Tested by:
- klaus3000 ........ r202342 | mmichelson | 2009-06-22 09:44:58
- -0500 (Mon, 22 Jun 2009) | 3 lines Remove an extra debug line
- left from previous commit. ........ ................
-
- * /, channels/chan_sip.c: Merged revisions 202337 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r202337 | mmichelson | 2009-06-22 09:35:09 -0500 (Mon, 22 Jun
- 2009) | 31 lines Merged revisions 202336 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r202336 | mmichelson | 2009-06-22 09:34:05 -0500 (Mon, 22 Jun
- 2009) | 25 lines Fix a possible infinite loop in SDP parsing
- during glare situation. There was a while loop in
- get_ip_and_port_from_sdp which was controlled by a call to
- get_sdp_iterate. The loop would exit either if what we were
- searching for was found or if the return was NULL. The problem is
- that get_sdp_iterate never returns NULL. This means that if what
- we were searching for was not present, the loop would run
- infinitely. This modification of the loop fixes the problem.
- (closes issue #15213) Reported by: schmidts (closes issue #15349)
- Reported by: samy (closes issue #14464) Reported by: pj (closes
- issue #15345) Reported by: aragon Patches: sip_inf_loop.patch
- uploaded by mmichelson (license 60) Tested by: aragon ........
- ................
-
-2009-06-21 16:14 +0000 [r202259-202263] Russell Bryant <russell@digium.com>
-
- * cdr/cdr_manager.c, /: Merged revisions 202262 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r202262 |
- russell | 2009-06-21 11:11:48 -0500 (Sun, 21 Jun 2009) | 2 lines
- Fix possibility of crashiness during reload in custom fields
- handling. ........
-
- * cdr/cdr_manager.c, /: Merged revisions 202258 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r202258 |
- russell | 2009-06-21 11:00:23 -0500 (Sun, 21 Jun 2009) | 2 lines
- Standardize return values of load_config() so reload() doesn't
- report an error on success. ........
-
-2009-06-20 19:14 +0000 [r202184] Sean Bright <sean@malleable.com>
-
- * /, apps/app_fax.c: Merged revisions 202183 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r202183 |
- seanbright | 2009-06-20 15:09:47 -0400 (Sat, 20 Jun 2009) | 5
- lines Fix version detection for API changes in spandsp. (closes
- issue #15355) Reported by: deuffy ........
-
-2009-06-19 21:07 +0000 [r202006] Matthew Nicholson <mnicholson@digium.com>
-
- * channels/chan_sip.c: Added deadlock protection to
- try_suggested_sip_codec in chan_sip.c. Review:
- https://reviewboard.asterisk.org/r/287/
-
-2009-06-19 20:27 +0000 [r201997] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 201994 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r201994 | dvossel | 2009-06-19 15:24:37 -0500
- (Fri, 19 Jun 2009) | 14 lines Merged revisions 201993 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r201993 | dvossel | 2009-06-19 15:22:02 -0500 (Fri, 19 Jun 2009)
- | 8 lines timestamp was being converted to host order as a short
- rather than a long (closes issue #15361) Reported by: ffloimair
- Patches: ts_issue.diff uploaded by dvossel (license 671) ........
- ................
-
-2009-06-19 00:44 +0000 [r201786-201830] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/features.c: Merged revisions 201829 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r201829 | tilghman | 2009-06-18 19:43:41 -0500 (Thu, 18 Jun 2009)
- | 13 lines Merged revisions 201828 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r201828 | tilghman | 2009-06-18 19:40:41 -0500 (Thu, 18 Jun 2009)
- | 6 lines If the "h" extension fails, give it another chance in
- main/pbx.c. If the "h" extension fails, give it another chance in
- main/pbx.c, when it returns from the bridge code. Fixes an issue
- where the "h" extension may occasionally not fire, when a Dial is
- executed from a Macro. Debugged in #asterisk with user tompaw.
- ........ ................
-
- * /, apps/Makefile: Merged revisions 201783 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r201783 |
- tilghman | 2009-06-18 15:52:36 -0500 (Thu, 18 Jun 2009) | 6 lines
- One of the changes in 1.6.1 was to allow app_directory to use
- functionality within app_voicemail for directory functions. It is
- therefore no longer necessary for app_directory to be linked
- against the ODBC libraries (and it never was necessary for
- app_directory to be linked against IMAP, though it was). ........
-
-2009-06-18 16:58 +0000 [r201682] David Vossel <dvossel@digium.com>
-
- * channels/misdn/isdn_lib.c, main/asterisk.c, utils/conf2ael.c,
- main/ast_expr2.c, utils/stereorize.c,
- codecs/gsm/src/gsm_destroy.c, /, channels/h323/ast_h323.cxx,
- main/ast_expr2f.c, res/ael/ael_lex.c, utils/ael_main.c,
- utils/extconf.c, pbx/pbx_config.c, res/res_config_ldap.c: Merged
- revisions 201678 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r201678 |
- dvossel | 2009-06-18 11:37:42 -0500 (Thu, 18 Jun 2009) | 11 lines
- fixes some memory leaks and redundant conditions (closes issue
- #15269) Reported by: contactmayankjain Patches: patch.txt
- uploaded by contactmayankjain (license 740)
- memory_leak_stuff.trunk.diff uploaded by dvossel (license 671)
- Tested by: contactmayankjain, dvossel ........
-
-2009-06-18 15:32 +0000 [r201612] Russell Bryant <russell@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 201610 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r201610 | russell | 2009-06-18 10:27:10 -0500
- (Thu, 18 Jun 2009) | 36 lines Merged revisions 201600 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r201600 | russell | 2009-06-18 10:24:31 -0500 (Thu, 18 Jun 2009)
- | 29 lines Fix memory corruption and leakage related reloads of
- non files mode MoH classes. For Music on Hold classes that are
- not files mode, meaning that we are executing an application that
- will feed us audio data, we use a thread to monitor the external
- application and read audio from it. This thread also makes use of
- the MoH class object. In the MoH class destructor, we used
- pthread_cancel() to ask the thread to exit. Unfortunately, the
- code did not wait to ensure that the thread actually went away.
- What needed to be done is a pthread_join() to ensure that the
- thread fully cleans up before we proceed. By adding this one
- line, we resolve two significant problems: 1) Since the thread
- was never joined, it never fully goes away. So, on every reload
- of non-files mode MoH, an unused thread was sticking around. 2)
- There was a race condition here where the application monitoring
- thread could still try to access the MoH class, even though the
- thread executing the MoH reload has already destroyed it. (issue
- #15109) Reported by: jvandal (issue #15123) Reported by:
- axisinternet (issue #15195) Reported by: amorsen (issue AST-208)
- ........ ................
-
-2009-06-17 20:10 +0000 [r201459-201463] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 201462 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r201462 |
- mmichelson | 2009-06-17 15:10:01 -0500 (Wed, 17 Jun 2009) | 12
- lines Fix problem with no audio due to ignoring the SDP. A recent
- change to our SDP version comparison made audio not function on
- some calls. This was because of a test wherein we were trying to
- see if an unsigned value was less than 0. This is a dumb
- comparison and arguably the compiler should have warned about it.
- Alas, though, it slipped past. Now it's fixed by changing the
- variable to be a signed type. Found by several developers. Tested
- by mnicholson and dbrooks. ........
-
- * main/channel.c, /: Merged revisions 201458 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r201458 | mmichelson | 2009-06-17 15:04:12 -0500 (Wed, 17 Jun
- 2009) | 15 lines Merged revisions 201450 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r201450 | mmichelson | 2009-06-17 14:59:31 -0500 (Wed, 17 Jun
- 2009) | 9 lines Change the datastore traversal in
- ast_do_masquerade to use a safe list traversal. It is possible
- for datastore fixup functions to remove the datastore from the
- list and free it. In particular, the queue_transfer_fixup in
- app_queue does this. While I don't yet know of this causing any
- crashes, it certainly could. Found while discussing a separate
- issue with Brian Degenhardt. ........ ................
-
-2009-06-17 19:55 +0000 [r201449] David Vossel <dvossel@digium.com>
-
- * apps/app_mixmonitor.c, /: Merged revisions 201445 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r201445 | dvossel | 2009-06-17 14:45:35 -0500
- (Wed, 17 Jun 2009) | 25 lines Merged revisions 201423 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r201423 | dvossel | 2009-06-17 14:28:12 -0500 (Wed, 17 Jun 2009)
- | 19 lines StopMixMonitor race condition (not giving up file
- immediately) StopMixMonitor only indicates to the MixMonitor
- thread to stop writing to the file. It does not guarantee that
- the recording's file handle is available to the dialplan
- immediately after execution. This results in a race condition. To
- resolve this, the filestream pointer is placed in a datastore on
- the channel. When StopMixMonitor is called, the datastore is
- retrieved from the channel and the filestream is closed
- immediately before returning to the dialplan. Documentation
- indicating the use of StopMixMonitor to free files has been
- updated as well. (closes issue #15259) Reported by: travisghansen
- Tested by: dvossel Review:
- https://reviewboard.asterisk.org/r/283/ ........ ................
-
-2009-06-17 19:35 +0000 [r201443] David Brooks <dbrooks@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 201381 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r201381 | dbrooks | 2009-06-17 14:15:07 -0500 (Wed, 17 Jun 2009)
- | 16 lines Merged revisions 201380 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r201380 | dbrooks | 2009-06-17 13:45:50 -0500 (Wed, 17 Jun 2009)
- | 9 lines Checks for NULL sip_pvt pointer in
- chan_sip.c->acf_channel_read() Zombie channels could be passed,
- and chan_sip.c wasn't checking for it. Could crash Asterisk. Now
- checking for NULL pointer. (closes issue #15330) Reported by:
- okrief Tested by: dbrooks ........ ................
-
-2009-06-17 12:05 +0000 [r201263] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, include/asterisk/linkedlists.h: Merged revisions 201262 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r201262 | kpfleming | 2009-06-17 07:04:17 -0500
- (Wed, 17 Jun 2009) | 15 lines Merged revisions 201261 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r201261 | kpfleming | 2009-06-17 07:03:25 -0500 (Wed, 17 Jun
- 2009) | 9 lines Correct AST_LIST_APPEND_LIST behavior when list
- to be appended is empty. When the list to be appended is empty,
- and the list to be appended to is *not*, AST_LIST_APPEND_LIST
- would actually cause the target list to become broken, and no
- longer have a pointer to its last entry. This patch fixes the
- problem. (reported by Stanislaw Pitucha on the asterisk-dev
- mailing list) ........ ................
-
-2009-06-16 22:31 +0000 [r201226] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 201223 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r201223 |
- dvossel | 2009-06-16 17:29:30 -0500 (Tue, 16 Jun 2009) | 2 lines
- fix issue with build_contact introduced by the "SIP trasnport
- type issues" commit ........
-
-2009-06-16 19:34 +0000 [r201093] Kevin P. Fleming <kpfleming@digium.com>
-
- * apps/app_chanspy.c, apps/app_mixmonitor.c, main/channel.c,
- main/autoservice.c, main/frame.c, /, apps/app_meetme.c,
- configure, main/slinfactory.c, autoconf/ast_gcc_attribute.m4,
- configure.ac, include/asterisk/linkedlists.h, main/file.c,
- include/asterisk/channel.h, include/asterisk/frame.h: Merged
- revisions 201056,201090 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r201056 | kpfleming | 2009-06-16 13:54:30 -0500 (Tue, 16 Jun
- 2009) | 18 lines Merged revisions 200991 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun
- 2009) | 11 lines Improve support for media paths that can
- generate multiple frames at once. There are various media paths
- in Asterisk (codec translators and UDPTL, primarily) that can
- generate more than one frame to be generated when the application
- calling them expects only a single frame. This patch addresses a
- number of those cases, at least the primary ones to solve the
- known problems. In addition it removes the broken TRACE_FRAMES
- support, fixes a number of bugs in various frame-related API
- functions, and cleans up various code paths affected by these
- changes. https://reviewboard.asterisk.org/r/175/ ........
- ................ r201090 | kpfleming | 2009-06-16 14:27:12 -0500
- (Tue, 16 Jun 2009) | 5 lines Another minor fix to compiler
- attribute checking. Defaulting to 'static' for the function scope
- was bad... so remove it. ................
-
-2009-06-16 17:11 +0000 [r200992] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 200946 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r200946 |
- dvossel | 2009-06-16 11:03:30 -0500 (Tue, 16 Jun 2009) | 32 lines
- SIP transport type issues What this patch addresses: 1.
- ast_sip_ouraddrfor() by default binds to the UDP address/port
- reguardless if the sip->pvt is of type UDP or not. Now when no
- remapping is required, ast_sip_ouraddrfor() checks the sip_pvt's
- transport type, attempting to set the address and port to the
- correct TCP/TLS bindings if necessary. 2. It is not necessary to
- send the port number in the Contact header unless the port is
- non-standard for the transport type. This patch fixes this and
- removes the todo note. 3. In sip_alloc(), the default dialog
- built always uses transport type UDP. Now sip_alloc() looks at
- the sip_request (if present) and determines what transport type
- to use by default. 4. When changing the transport type of a
- sip_socket, the file descriptor must be set to -1 and in some
- cases the tcptls_session's ref count must be decremented and set
- to NULL. I've encountered several issues associated with this
- process and have created a function, set_socket_transport(), to
- handle the setting of the socket type. (closes issue #13865)
- Reported by: st Patches: dont_add_port_if_tls.patch uploaded by
- Kristijan (license 753) 13865.patch uploaded by mmichelson
- (license 60) tls_port_v5.patch uploaded by vrban (license 756)
- transport_issues.diff uploaded by dvossel (license 671) Tested
- by: mmichelson, Kristijan, vrban, jmacz, dvossel Review:
- https://reviewboard.asterisk.org/r/278/ ........
-
-2009-06-16 16:34 +0000 [r200986] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, configure, autoconf/ast_gcc_attribute.m4, configure.ac: Merged
- revisions 200985 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r200985 |
- kpfleming | 2009-06-16 11:32:36 -0500 (Tue, 16 Jun 2009) | 7
- lines Fix problems with new compiler attribute checking in
- configure script. The last changes to ast_gcc_attribute.m4 caused
- some problems checking for various attributes, because the scope
- of the symbol the attribute is applied to can be important; this
- patch allows the scope to be specified for the check. ........
-
-2009-06-16 16:02 +0000 [r200945] Michiel van Baak <michiel@vanbaak.info>
-
- * apps/app_voicemail.c, /: Merged revisions 200943 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r200943 | mvanbaak | 2009-06-16 17:51:36 +0200 (Tue, 16 Jun 2009)
- | 9 lines add FILE_STORAGE to Voicemail Build Options Voicemail
- can only use one storage module at the moment. Because it's
- unclear that selecting one of the storage modules in menuselect
- will disable filesystem storage we now have a FILE_STORAGE option
- that conflicts with the other modules. (closes issue #15333)
- ........
-
-2009-06-16 01:33 +0000 [r200724-200767] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, configure, include/asterisk/autoconfig.h.in,
- autoconf/ast_gcc_attribute.m4: Merged revisions 200764 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r200764 | kpfleming | 2009-06-15 20:28:08 -0500 (Mon, 15
- Jun 2009) | 11 lines Ensure that configure-script testing for
- compiler attributes actually works. The configure script tests
- for compiler attributes didn't actually enable enough warnings or
- provide a proper test harness to determine whether the compiler
- supports the attribute in question or not; this caused gcc 4.1 to
- report that it supports 'weakref', but it doesn't actually
- support it in the way that is needed for our optional API
- mechanism. The new configure script test will properly
- distinguish between full support and partial support for this
- attribute, among others. ........
-
- * /, CHANGES: Merged revisions 200726 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r200726 |
- kpfleming | 2009-06-15 20:03:22 -0500 (Mon, 15 Jun 2009) | 6
- lines Document the new automatic 'ignoresdpversion' behavior.
- Asterisk will now automatically ignore incorrect incoming SDP
- version numbers when necessary to complete a T.38 re-INVITE
- operation. ........
-
- * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
- 165180,200689 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r165180 |
- mnicholson | 2008-12-17 12:49:12 -0600 (Wed, 17 Dec 2008) | 14
- lines This patch adds a new 'ignoresdpversion' option to
- sip.conf. When this is enabled (either globally or for a specific
- peer), chan_sip will treat any SDP data it receives as new data
- and update the media stream accordingly. By default, Asterisk
- will only modify the media stream if the SDP session version
- received is different from the current SDP session version. This
- option is required to interoperate with devices that have
- non-standard SDP session version implementations (observed by toc
- on the bug tracker with Microsoft OCS which always uses 0 as the
- session version). http://reviewboard.digium.com/r/94/ (closes
- issue #13958) Reported by: toc Tested by: toc ........ r200689 |
- kpfleming | 2009-06-15 15:42:38 -0500 (Mon, 15 Jun 2009) | 12
- lines Accept T.38 re-INVITE responses with invalid SDP versions.
- This commit changes the 'incoming SDP version' check logic a bit
- more; when 'ignoresdpversion' is *not* set for a peer, if we
- initiate a re-INVITE to switch to T.38, we'll always accept the
- peer's SDP response, even if they don't properly increment the
- SDP version number as they should. If this situation occurs, a
- warning message will be generated suggesting that the peer's
- configuration be changed to include the 'ignoresdpversion'
- configuration option (although ideally they'd fix their SIP
- implementation to be RFC compliant). AST-221 ........
-
-2009-06-15 15:22 +0000 [r200515] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 200514 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r200514 | mmichelson | 2009-06-15 10:22:11 -0500 (Mon, 15 Jun
- 2009) | 11 lines Merged revisions 200513 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r200513 | mmichelson | 2009-06-15 10:21:46 -0500 (Mon, 15 Jun
- 2009) | 5 lines Add INFO to our allowed methods so that endpoints
- know they may send it to us. AST-223 ........ ................
-
-2009-06-12 19:08 +0000 [r200362] Mark Michelson <mmichelson@digium.com>
-
- * main/channel.c, /: Merged revisions 200361 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r200361 | mmichelson | 2009-06-12 14:07:51 -0500 (Fri, 12 Jun
- 2009) | 16 lines Merged revisions 200360 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r200360 | mmichelson | 2009-06-12 14:06:41 -0500 (Fri, 12 Jun
- 2009) | 10 lines Suppress a warning message and give a better
- return code when generating inband ringing after a call is
- answered. (closes issue #15158) Reported by: madkins Patches:
- 15158.patch uploaded by mmichelson (license 60) Tested by:
- madkins ........ ................
-
-2009-06-11 22:42 +0000 [r200228] Sean Bright <sean@malleable.com>
-
- * Makefile, /: Merged revisions 199781 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r199781 |
- seanbright | 2009-06-09 14:08:53 -0400 (Tue, 09 Jun 2009) | 2
- lines Fix all of the parallel build warnings issued when running
- make -j#. ........
-
-2009-06-11 21:18 +0000 [r200149] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 200146 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r200146 |
- mmichelson | 2009-06-11 16:17:14 -0500 (Thu, 11 Jun 2009) | 5
- lines Fix a crash due to a potentially NULL p->options. Thanks to
- mnicholson for pointing it out. ........
-
-2009-06-11 12:16 +0000 [r200040] Leif Madsen <lmadsen@digium.com>
-
- * /, build_tools/make_version_c, build_tools/make_version_h: Merged
- revisions 200039 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r200039 |
- lmadsen | 2009-06-11 08:15:09 -0400 (Thu, 11 Jun 2009) | 8 lines
- Fix path for .flavor and .version (issue #14737) Reported by:
- davidw Patches: flavor.patch uploaded by davidw (license 780)
- Tested by: davidw ........
-
-2009-06-10 20:29 +0000 [r199994] David Brooks <dbrooks@digium.com>
-
- * main/pbx.c, /: Fixes the argument order in definition of
- new_find_extension(). In the definition of new_find_extension(),
- the arguments 'callerid' and 'label' were swapped. The prototype
- declaration and all calls to the function are ordered 'callerid'
- then 'label', but the function itself was ordered 'label' then
- 'callerid'. (closes issue #15303) Reported by: JimDickenson
-
-2009-06-10 20:20 +0000 [r199975] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: The 1.6.0 branch was missing all
- invite_branch logic. It has now been added.
-
-2009-06-10 16:13 +0000 [r199858] Sean Bright <sean@malleable.com>
-
- * include/asterisk/utils.h, /: Merged revisions 199857 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r199857 | seanbright | 2009-06-10 12:10:23 -0400
- (Wed, 10 Jun 2009) | 9 lines Merged revisions 199856 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r199856 | seanbright | 2009-06-10 12:08:35 -0400 (Wed,
- 10 Jun 2009) | 2 lines __WORDSIZE is not available on all
- platforms, so use sizeof(void *) instead. ........
- ................
-
-2009-06-09 20:54 +0000 [r199821] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 199818 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r199818 |
- dvossel | 2009-06-09 15:47:57 -0500 (Tue, 09 Jun 2009) | 11 lines
- CLI NOTIFY sending wrong transport type. SIP's cli NOTIFY command
- only used UDP rather than copying the transport type from the
- peer. (closes issue #15283) Reported by: jthurman Patches:
- sip-notify-tcp-svn199728.patch uploaded by jthurman (license 614)
- Tested by: jthurman, dvossel ........
-
-2009-06-08 19:39 +0000 [r199632] Sean Bright <sean@malleable.com>
-
- * include/asterisk/utils.h, /: Merged revisions 199630 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r199630 | seanbright | 2009-06-08 15:33:09 -0400
- (Mon, 08 Jun 2009) | 32 lines Merged revisions 199626,199628 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r199626 | seanbright | 2009-06-08 15:24:32 -0400 (Mon, 08 Jun
- 2009) | 21 lines Increase the size of our thread stack on 64 bit
- processors. We were setting the stack size for each thread to
- 240KB regardless of architecture, which meant that in some
- scenarios we actually had less available stack space on 64 bit
- processors (pointers use 8 bytes instead of 4). So now we
- calculate the stack size we reserve based on the platform's
- __WORDSIZE, which gives us: 32 bit -> 240KB 64 bit -> 496KB 128
- bit -> 1008KB (that's right, we're ready for 128 bit processors)
- Patch typed by me but written by several members of
- #asterisk-dev, including Kevin, Tilghman, and Qwell. (closes
- issue #14932) Reported by: jpiszcz Patches:
- 06052009_issue14932.patch uploaded by seanbright (license 71)
- Tested by: seanbright ........ r199628 | seanbright | 2009-06-08
- 15:28:33 -0400 (Mon, 08 Jun 2009) | 2 lines Fix a typo in the
- stack size calculation just introduced. ........ ................
-
-2009-06-05 21:37 +0000 [r199301] David Vossel <dvossel@digium.com>
-
- * main/pbx.c, /: Merged revisions 199298 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r199298 | dvossel | 2009-06-05 16:21:22 -0500 (Fri, 05 Jun 2009)
- | 21 lines Merged revisions 199297 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r199297 | dvossel | 2009-06-05 16:19:56 -0500 (Fri, 05 Jun 2009)
- | 14 lines Fixes issue with hints giving unexpected results.
- Hints with two or more devices that include ONHOLD gave
- unexpected results. (closes issue #15057) Reported by:
- p_lindheimer Patches: onhold_trunk.diff uploaded by dvossel
- (license 671) pbx.c.1.4.patch uploaded by p (license 558)
- devicestate.c.trunk.patch uploaded by p (license 671) Tested by:
- p_lindheimer, dvossel Review:
- https://reviewboard.asterisk.org/r/254/ ........ ................
-
-2009-06-05 13:51 +0000 [r199228] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 199227 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r199227 | mmichelson | 2009-06-05 08:51:08 -0500 (Fri, 05 Jun
- 2009) | 14 lines Correct "dahdi show channels" output when
- specifying a group. Since a DAHDI channel may belong to multiple
- groups, we need to use a bitwise and instead of equivalence to
- determine whether to display the channel information. (closes
- issue #15248) Reported by: gentian Patches: 15248.patch uploaded
- by mmichelson (license 60) Tested by: gentian ........
-
-2009-06-04 19:16 +0000 [r199142] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 199139 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r199139 | dvossel | 2009-06-04 14:10:16 -0500
- (Thu, 04 Jun 2009) | 9 lines Merged revisions 199138 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r199138 | dvossel | 2009-06-04 14:00:15 -0500 (Thu, 04
- Jun 2009) | 3 lines Additional updates to AST-2009-001 ........
- ................
-
-2009-08-11 Tilghman Lesher <tlesher@digium.com>
-
- * Asterisk 1.6.0.13 released
-
- * channels/chan_sip.c: Bad merge from 1.6.0 branch
-
-2009-08-10 Tilghman Lesher <tlesher@digium.com>
-
- * Asterisk 1.6.0.12 released
-
- * AST-2009-005
-
-2009-06-05 Leif Madsen <lmadsen@digium.com>
-
- * Asterisk 1.6.0.10 released
-
-2009-06-04 David Vossel <dvossel@digium.com>
-
- * channels/chan_iax2.c: Additional updates for AST-2009-001
-
-2009-06-04 David Vossel <dvossel@digium.com>
-
- * channels/chan_iax2.c: REGAUTH loop fix related to AST-2009-001
-
-2009-04-06 Leif Madsen <lmadsen@digium.com>
-
- * Release Asterisk 1.6.0.9
-
-2009-04-03 16:27 -0500 [r186517] Mark Michelson <mmichelson@digium.com>
-
- * Remove an invalid call to free memory.
-
- A bad merge from trunk to 1.6.0 meant freeing memory that
- should not be freed. In trunk, pkt->data is an ast_str, but
- in 1.6.0, it is allocated in the same chunk of memory as the
- sip_pkt. This only affects 1.6.0.
-
- (closes issue #14819)
- Reported by: cwolff09
-
-2009-04-02 Leif Madsen <lmadsen@digium.com>
-
- * Release Asterisk 1.6.0.8
-
-2009-04-02 Tilghman Lesher <tlesher@digium.com>
-
- * Fix for security issue AST-2009-003
-
-2009-03-30 Leif Madsen <lmadsen@digium.com>
-
- * Release Asterisk 1.6.0.7
-
-2009-03-19 Leif Madsen <lmadsen@digium.com>
-
- * Release Asterisk 1.6.0.7-rc2
-
-2009-03-19 15:40 +0000 [r183066-183109] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 183108 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r183108 |
- file | 2009-03-19 12:37:23 -0300 (Thu, 19 Mar 2009) | 11 lines
- Improve our triggering of a T38 switchover internally when
- triggered by a received reinvite. Previously we reached across
- the channel bridge to get the other party's SIP dialog structure
- in order to trigger an outgoing reinvite. This is extremely
- dangerous to do and only works if bridged to another SIP channel.
- This patch changes this to use the T38 control frame method of
- requesting a switchover. This change also causes the SIP channel
- driver to propogate back whether the switchover worked or not
- instead of blindly accepting the incoming T38 reinvite. Review:
- http://reviewboard.digium.com/r/200/ ........
-
- * main/channel.c, /: Merged revisions 183057 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r183057 |
- file | 2009-03-18 19:22:56 -0300 (Wed, 18 Mar 2009) | 6 lines Fix
- an issue where a T38 control frame would get dropped. If two
- channels were bridged together using a generic bridge the T38
- control frame would get passed up instead of being indicated on
- the other channel. ........
-
-2009-03-18 21:19 +0000 [r183029] Jeff Peeler <jpeeler@digium.com>
-
- * /, channels/h323/ast_h323.cxx: Merged revisions 183028 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r183028 | jpeeler | 2009-03-18 16:18:27 -0500 (Wed, 18
- Mar 2009) | 4 lines Add some code removed by mistake from commit
- 182722 that works around a file descriptor leak in versions of
- PWLib prior to 1.12.0. ........
-
-2009-03-18 14:24 +0000 [r182945] Russell Bryant <russell@digium.com>
-
- * main/poll.c, main/io.c, main/channel.c, channels/chan_skinny.c,
- configure, apps/app_mp3.c, res/res_agi.c,
- include/asterisk/poll-compat.h, channels/chan_alsa.c,
- main/asterisk.c, apps/app_nbscat.c, /, main/Makefile,
- include/asterisk/autoconfig.h.in, configure.ac,
- include/asterisk/io.h, main/utils.c, include/asterisk/channel.h:
- Merged revisions 182847 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r182847 | russell | 2009-03-17 21:28:55 -0500 (Tue, 17 Mar 2009)
- | 52 lines Merged revisions 182810 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009)
- | 44 lines Fix cases where the internal poll() was not being used
- when it needed to be. We have seen a number of problems caused by
- poll() not working properly on Mac OSX. If you search around,
- you'll find a number of references to using select() instead of
- poll() to work around these issues. In Asterisk, we've had poll.c
- which implements poll() using select() internally. However, we
- were still getting reports of problems. vadim investigated a bit
- and realized that at least on his system, even though we were
- compiling in poll.o, the system poll() was still being used. So,
- the primary purpose of this patch is to ensure that we're using
- the internal poll() when we want it to be used. The changes are:
- 1) Remove logic for when internal poll should be used from the
- Makefile. Instead, put it in the configure script. The logic in
- the configure script is the same as it was in the Makefile.
- Ideally, we would have a functionality test for the problem, but
- that's not actually possible, since we would have to be able to
- run an application on the _target_ system to test poll()
- behavior. 2) Always include poll.o in the build, but it will be
- empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll()
- throughout the source tree to ast_poll(). I feel that it is good
- practice to give the API call a new name when we are changing its
- behavior and not using the system version directly in all cases.
- So, normally, ast_poll() is just redefined to poll(). On systems
- where AST_POLL_COMPAT is defined, ast_poll() is redefined to
- ast_internal_poll(). 4) Change poll() in main/poll.c to be
- ast_internal_poll(). It's worth noting that any code that still
- uses poll() directly will work fine (if they worked fine before).
- So, for example, out of tree modules that are using poll() will
- not stop working or anything. However, for modules to work
- properly on Mac OSX, ast_poll() needs to be used. (closes issue
- #13404) Reported by: agalbraith Tested by: russell, vadim
- http://reviewboard.digium.com/r/198/ ........ ................
-
-2009-03-17 20:51 +0000 [r182723] Jeff Peeler <jpeeler@digium.com>
-
- * channels/h323/compat_h323.cxx, /, channels/h323/ast_h323.cxx,
- configure, autoconf/ast_check_openh323.m4,
- channels/h323/compat_h323.h, channels/chan_h323.c,
- channels/h323/ast_h323.h, channels/h323/chan_h323.h: Merged
- revisions 182722 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r182722 |
- jpeeler | 2009-03-17 15:47:31 -0500 (Tue, 17 Mar 2009) | 15 lines
- Allow H.323 Plus library to be used in addition to the OpenH323
- library Chan_h323 can now be compiled against both the previously
- supported versions of OpenH323 as well as the current H.323 Plus
- (version 1.20.2). The configure script has been modified to look
- in the default install location of h323 to hopefully help avoid
- using the environment variables OPENH323DIR and PWLIBDIR. Also,
- the CLI command "h323 show version" has been added which
- indicates which version of h323 is in use. (closes issue #11261)
- Reported by: vhatz Patches: asterisk-1.6.0.6-h323plus.patch
- uploaded by jthurman (license 614) ........
-
-2009-03-17 15:27 +0000 [r182569] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /: Merged revisions 182553 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r182553 |
- russell | 2009-03-17 10:22:12 -0500 (Tue, 17 Mar 2009) | 5 lines
- Tweak the handling of the frame list inside of ast_answer(). This
- does not change any behavior, but moves the frames from the local
- frame list back to the channel read queue using an O(n) algorithm
- instead of O(n^2). ........
-
-2009-03-17 15:00 +0000 [r182526-182532] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/channel.c, /: Merged revisions 182530 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r182530 |
- kpfleming | 2009-03-17 09:59:33 -0500 (Tue, 17 Mar 2009) | 2
- lines correct logic flaw in ast_answer() changes in r182525
- ........
-
- * main/channel.c, /, main/features.c, include/asterisk/channel.h:
- Merged revisions 182525 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r182525 |
- kpfleming | 2009-03-17 09:38:11 -0500 (Tue, 17 Mar 2009) | 11
- lines Improve behavior of ast_answer() to not lose incoming
- frames ast_answer(), when supplied a delay before returning to
- the caller, use ast_safe_sleep() to implement the delay.
- Unfortunately during this time any incoming frames are discarded,
- which is problematic for T.38 re-INVITES and other sorts of
- channel operations. When a delay is not passed to ast_answer(),
- it still delays for up to 500 milliseconds, waiting for media to
- arrive. Again, though, it discards any control frames, or
- non-voice media frames. This patch rectifies this situation, by
- storing all incoming frames during the delay period on a list,
- and then requeuing them onto the channel before returning to the
- caller. http://reviewboard.digium.com/r/196/ ........
-
-2009-03-17 05:53 +0000 [r182451] Tilghman Lesher <tlesher@digium.com>
-
- * main/db.c, /: Merged revisions 182450 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r182450 | tilghman | 2009-03-17 00:51:54 -0500 (Tue, 17 Mar 2009)
- | 14 lines Merged revisions 182449 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r182449 | tilghman | 2009-03-17 00:50:52 -0500 (Tue, 17 Mar 2009)
- | 7 lines Fix race in astdb The underlying db1 implementation
- does not fully isolate the pages retrieved from astdb, so the
- lock protecting accesses needs to be extended until the copy from
- the shared memory structure is done. (closes issue #14682)
- Reported by: makoto ........ ................
-
-2009-03-16 17:52 +0000 [r182283] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 182282 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r182282 | dvossel | 2009-03-16 12:49:58 -0500
- (Mon, 16 Mar 2009) | 13 lines Merged revisions 182281 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r182281 | dvossel | 2009-03-16 12:47:42 -0500 (Mon, 16 Mar 2009)
- | 7 lines Randomize IAX2 encryption padding The 16-32 byte random
- padding at the beginning of an encrypted IAX2 frame turns out to
- not be all that random at all. This patch calls ast_random to
- fill the padding buffer with random data. The padding is
- randomized at the beginning of every encrypted call and for every
- encrypted retransmit frame. Review:
- http://reviewboard.digium.com/r/193/ ........ ................
-
-2009-03-16 17:36 +0000 [r182212-182279] Tilghman Lesher <tlesher@digium.com>
-
- * /, funcs/func_env.c: Merged revisions 182278 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r182278 |
- tilghman | 2009-03-16 12:33:38 -0500 (Mon, 16 Mar 2009) | 7 lines
- Fix an off-by-one error in the FILE() function, and extend
- FILE()'s length parameter to work like variable substitution.
- Previously, FILE() returned one less character than specified,
- due to the terminating NULL. Both the offset and length
- parameters now behave identically to the way variable
- substitution offsets and lengths also work. (closes issue #14670)
- Reported by: BMC ........
-
- * channels/chan_local.c, /: Merged revisions 182211 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r182211 | tilghman | 2009-03-16 10:50:55 -0500
- (Mon, 16 Mar 2009) | 14 lines Merged revisions 182208 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r182208 | tilghman | 2009-03-16 10:39:15 -0500 (Mon, 16 Mar 2009)
- | 7 lines Fixup glare detection, to fix a memory leak of a local
- pvt structure. (closes issue #14656) Reported by: caspy Patches:
- 20090313__bug14656__2.diff.txt uploaded by tilghman (license 14)
- Tested by: caspy ........ ................
-
-2009-03-16 13:59 +0000 [r182172] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /: Merged revisions 182171 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r182171 |
- file | 2009-03-16 10:58:24 -0300 (Mon, 16 Mar 2009) | 7 lines Fix
- a memory leak in the ast_answer / __ast_answer API call. For a
- channel that is not yet answered this API call will wait until a
- voice frame is received on the channel before returning. It does
- this by waiting for frames on the channel and reading them in.
- The frames read in were not freed when they should have been.
- ........
-
-2009-03-13 21:26 +0000 [r182064-182122] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 182121 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r182121 |
- mmichelson | 2009-03-13 16:26:20 -0500 (Fri, 13 Mar 2009) | 6
- lines Change faulty comparison used when announcing average hold
- minutes and seconds (closes issue #14227) Reported by: caspy
- ........
-
- * /, main/features.c: Merged revisions 182029 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r182029 | mmichelson | 2009-03-13 12:26:43 -0500 (Fri, 13 Mar
- 2009) | 41 lines Merged revisions 181990 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r181990 | mmichelson | 2009-03-13 12:12:32 -0500 (Fri, 13 Mar
- 2009) | 35 lines Check the DYNAMIC_FEATURES of both the chan and
- peer when interpreting DTMF. Dynamic features defined in the
- applicationmap section of features.conf allow one to specify
- whether the caller, callee, or both have the ability to use the
- feature. The documentation in the features.conf.sample file could
- be interpreted to mean that one only needs to set the
- DYNAMIC_FEATURES channel variable on the calling channel in order
- to allow for the callee to be able to use the features which he
- should have permission to use. However, the DYNAMIC_FEATURES
- variable would only be read from the channel of the participant
- that pressed the DTMF sequence to activate the feature. The
- result of this was that the callee was unable to use dynamic
- features unless the dialplan writer had taken measures to be sure
- that the DYNAMIC_FEATURES variable was set on the callee's
- channel. This commit changes the behavior of
- ast_feature_interpret to concatenate the values of
- DYNAMIC_FEATURES from both parties involved in the bridge. The
- features themselves determine who has permission to use them, so
- there is no reason to believe that one side of the bridge could
- gain the ability to perform an action that they should not have
- the ability to perform. Kevin Fleming pointed out on the
- asterisk-users list that the typical way that this was worked
- around in the past was by setting _DYNAMIC_FEATURES on the
- calling channel so that the value would be inherited by the
- called channel. While this works, the documentation alone is not
- enough to figure out why this is necessary for the callee to be
- able to use dynamic features. In this particular case, changing
- the code to match the documentation is safe, easy, and will
- generally make things easier for people for future installations.
- This bug was originally reported on the asterisk-users list by
- David Ruggles. (closes issue #14657) Reported by: mmichelson
- Patches: 14657.patch uploaded by mmichelson (license 60) ........
- ................
-
-2009-03-13 17:28 +0000 [r182036] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 182022 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r182022 |
- file | 2009-03-13 14:25:09 -0300 (Fri, 13 Mar 2009) | 7 lines Fix
- an issue with requesting a T38 reinvite before the call is
- answered. The code responsible for sending the T38 reinvite did
- not check if an INVITE was already being handled. This caused
- things to get confused and the call to fail. The code now defers
- sending the T38 reinvite until the current INVITE is done being
- handled. (issue AST-191) ........
-
-2009-03-12 21:44 +0000 [r181770-181848] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 181846 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r181846 |
- mmichelson | 2009-03-12 16:43:51 -0500 (Thu, 12 Mar 2009) | 3
- lines Run the macro on the queue member's channel when he
- answers, not the caller's channel. ........
-
- * /, channels/chan_sip.c: Merged revisions 181769 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r181769 | mmichelson | 2009-03-12 13:30:58 -0500 (Thu, 12 Mar
- 2009) | 28 lines Merged revisions 181768 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r181768 | mmichelson | 2009-03-12 13:29:48 -0500 (Thu, 12 Mar
- 2009) | 22 lines Properly send a 487 on an INVITE we have not
- responded to if we receive a BYE. If we receive an INVITE from an
- endpoint and then later receive a BYE from that same endpoint
- before we have sent a final response for the INVITE, then we need
- to respond to the INVITE with a 487. There was logic in the code
- prior to this commit which seemed to exist solely to handle this
- situation, but there was one condition in an if statement which
- was incorrect. The only way we would send a 487 was if the
- sip_pvt had no owner channel. This made no sense since we created
- the owner channel when we received the INVITE, meaning that the
- majority of the time we would never send the 487. The 487 being
- sent should not rely on whether we have created a channel. Its
- delivery should be dependent on the current state of the initial
- INVITE transaction. With this commit, that logic is now correctly
- in place. (closes issue #14149) Reported by: legranjl Patches:
- 14149.patch uploaded by mmichelson (license 60) Tested by:
- legranjl ........ ................
-
-2009-03-12 17:58 +0000 [r181732] Tilghman Lesher <tlesher@digium.com>
-
- * /, configure, main/translate.c: Merged revisions 181731 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r181731 | tilghman | 2009-03-12 12:32:13 -0500 (Thu, 12
- Mar 2009) | 9 lines Adjust translation table column widths based
- upon the translation times. Previously, only 5 columns were
- displayed, and if a translation time exceeded 99,999 useconds, it
- would be displayed as 0, instead of its actual time. (closes
- issue #14532) Reported by: pj Patches:
- 20090311__bug14532.diff.txt uploaded by tilghman (license 14)
- Tested by: pj ........
-
-2009-03-12 16:57 +0000 [r181613-181666] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 181665 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r181665 | file | 2009-03-12 13:56:58 -0300 (Thu,
- 12 Mar 2009) | 9 lines Merged revisions 181664 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r181664 | file | 2009-03-12 13:56:20 -0300 (Thu, 12 Mar 2009) | 2
- lines Fix incorrect usage of strncasecmp... I really meant to use
- strcasecmp. ........ ................
-
- * /, res/res_musiconhold.c: Merged revisions 181661 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r181661 | file | 2009-03-12 13:53:52 -0300 (Thu,
- 12 Mar 2009) | 19 lines Merged revisions 181659-181660 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r181659 | file | 2009-03-12 13:50:37 -0300 (Thu, 12 Mar 2009) | 8
- lines Fix another scenario where depending on configuration the
- stream would not get read. For custom commands we don't know
- whether the audio is coming from a stream or not so we are going
- to have to read the data despite no channels. (closes issue
- #14416) Reported by: caspy ........ r181660 | file | 2009-03-12
- 13:52:45 -0300 (Thu, 12 Mar 2009) | 2 lines Fix logic flaw in
- previous commit. ........ ................
-
- * /, res/res_musiconhold.c: Merged revisions 181656 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r181656 | file | 2009-03-12 13:32:20 -0300 (Thu,
- 12 Mar 2009) | 17 lines Merged revisions 181655 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r181655 | file | 2009-03-12 13:29:19 -0300 (Thu, 12 Mar 2009) |
- 10 lines Fix issue with streaming MOH failing if nobody is
- listening. When a music class is setup to actually provide music
- on hold from a stream we need to constantly read audio from it
- since it will constantly be providing audio. This is now done
- despite there being no channels listening to it. (closes issue
- #14416) Reported by: caspy ........ ................
-
- * apps/app_dial.c, /: Merged revisions 181612 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r181612 |
- file | 2009-03-12 10:24:12 -0300 (Thu, 12 Mar 2009) | 5 lines Fix
- crash when sleep and retries argument was not given to RetryDial
- application. (closes issue #14647) Reported by: sherpya ........
-
-2009-03-12 01:04 +0000 [r181543] Richard Mudgett <rmudgett@digium.com>
-
- * /, build_tools/make_version: Merged revisions 181542 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r181542 | rmudgett | 2009-03-11 20:00:29 -0500 (Wed, 11 Mar 2009)
- | 1 line Use the correct branch integrated property when
- generating the version string ........
-
-2009-03-11 23:19 +0000 [r181509] Michiel van Baak <michiel@vanbaak.info>
-
- * /, configs/sip.conf.sample: Merged revisions 181499 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk Provide
- correct hint to debug SIP trouble in the default config (closes
- issue #14646) Reported by: strk
-
-2009-03-11 22:22 +0000 [r181450] Jason Parker <jparker@digium.com>
-
- * /, configure, configure.ac: Merged revisions 181444 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r181444 | qwell | 2009-03-11 17:20:13 -0500
- (Wed, 11 Mar 2009) | 11 lines Merged revisions 181436 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r181436 | qwell | 2009-03-11 17:18:42 -0500 (Wed, 11 Mar 2009) |
- 4 lines Allow prefix to set localstatedir (when used and
- different from the default). This is similar to the /etc change
- that was made for the non-FreeBSD case. ........ ................
-
-2009-03-11 22:15 +0000 [r181425-181429] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /: Merged revisions 181428 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r181428 |
- russell | 2009-03-11 17:14:55 -0500 (Wed, 11 Mar 2009) | 2 lines
- Make handling of the BRIDGEPVTCALLID variable thread-safe.
- ........
-
- * main/channel.c, /: Merged revisions 181424 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r181424 | russell | 2009-03-11 16:49:29 -0500 (Wed, 11 Mar 2009)
- | 17 lines Merged revisions 181423 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r181423 | russell | 2009-03-11 16:42:58 -0500 (Wed, 11 Mar 2009)
- | 9 lines Make code that updates BRIDGEPEER variable thread-safe.
- It is not safe to read the name field of an ast_channel without
- the channel locked. This patch fixes some places in channel.c
- where this was being done, and lead to crashes related to
- masquerades. (closes issue #14623) Reported by: guillecabeza
- ........ ................
-
-2009-03-11 17:37 +0000 [r181372] David Vossel <dvossel@digium.com>
-
- * channels/iax2-parser.h, /, channels/chan_iax2.c: Merged revisions
- 181371 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r181371 | dvossel | 2009-03-11 12:34:57 -0500 (Wed, 11 Mar 2009)
- | 17 lines Merged revisions 181340 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r181340 | dvossel | 2009-03-11 12:25:31 -0500 (Wed, 11 Mar 2009)
- | 11 lines encrypted IAX2 during packet loss causes decryption to
- fail on retransmitted frames If an iax channel is encrypted, and
- a retransmit frame is sent, that packet's iseqno is updated while
- it is encrypted. This causes the entire frame to be corrupted.
- When the corrupted frame is sent, the other side decrypts it and
- sends a VNAK back because the decrypted frame doesn't make any
- sense. When we get the VNAK, we look through the sent queue and
- send the same corrupted frame causing a loop. To fix this,
- encrypted frames requiring retransmission are decrypted, updated,
- then re-encrypted. Since key-rotation may change the key held by
- the pvt struct, the keys used for encryption/decryption are held
- within the iax_frame to guarantee they remain correct. (closes
- issue #14607) Reported by: stevenla Tested by: dvossel Review:
- http://reviewboard.digium.com/r/192/ ........ ................
-
-2009-03-11 17:28 +0000 [r181297-181352] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 181345 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r181345 | file | 2009-03-11 14:26:40 -0300 (Wed, 11 Mar 2009) |
- 21 lines Merged revisions 181328 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r181328 | file | 2009-03-11 14:22:52 -0300 (Wed, 11 Mar 2009) |
- 14 lines Fix issue where an attended transfer could not be
- completed under a rare scenario. When completing an attended
- transfer chan_sip does a check to make sure the extension in the
- URI portion of the Refer-To header is a local valid extension. We
- don't actually need to check this since we know for sure the
- other channel is already up and talking to the extension. Some
- devices do not put the extension in the Refer-To header either,
- which can cause the extension check to fail. We now no longer do
- this check if it is an attended transfer. (closes issue #14628)
- Reported by: sverre Patches: 14628.diff uploaded by file (license
- 11) ........ ................
-
- * /, channels/chan_sip.c: Merged revisions 181296 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r181296 | file | 2009-03-11 13:40:48 -0300 (Wed, 11 Mar 2009) |
- 16 lines Merged revisions 181295 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r181295 | file | 2009-03-11 13:36:50 -0300 (Wed, 11 Mar 2009) | 9
- lines Fix a problem with inband DTMF detection on outgoing SIP
- calls when dtmfmode=auto. When dtmfmode was set to auto the
- inband DTMF detector was not setup on outgoing SIP calls. This
- caused inband DTMF detection to fail. The inband DTMF detector is
- now setup for both dtmfmode inband and auto. (closes issue
- #13713) Reported by: makoto ........ ................
-
-2009-03-11 15:54 +0000 [r181137-181284] Jeff Peeler <jpeeler@digium.com>
-
- * channels/h323/ast_h323.cxx: add missing header file
-
- * utils/extconf.c: Fix merge oops from 181137
-
- * utils/Makefile, include/asterisk/utils.h,
- include/asterisk/astmm.h, /, channels/chan_sip.c,
- channels/h323/ast_h323.cxx, utils/extconf.c: Merged revisions
- 181135 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r181135 |
- jpeeler | 2009-03-10 23:06:44 -0500 (Tue, 10 Mar 2009) | 20 lines
- Fix malloc debug macros to work properly with h323. The main
- problem here was that cstdlib was undefining free thereby causing
- the proper debug macros to not be used. ast_h323.cxx has been
- changed to call ast_free instead to avoid the issue. A few other
- issues were addressed: - There were a few instances of functions
- improperly passing ast_free instead of ast_free_ptr. - Some clean
- up was done to avoid the debug macros intentionally being
- redefined. (copied below from Kevin's commit, appreciate the
- help) - disable astmm.h from doing anything when STANDALONE is
- defined, which is used by the tools in the utils/ directory that
- use parts of Asterisk header files in hackish ways; also ensure
- that utils/extconf.c and utils/conf2ael.c are compiled with
- STANDALONE defined. (closes issue #13593) Reported by: pj
- ........
-
-2009-03-11 00:52 +0000 [r181034] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 181032-181033 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r181032 | mmichelson | 2009-03-10 19:46:47 -0500
- (Tue, 10 Mar 2009) | 19 lines Merged revisions 181029,181031 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r181029 | mmichelson | 2009-03-10 19:30:26 -0500 (Tue, 10 Mar
- 2009) | 9 lines Fix incorrect tag checking on transfers when
- pedantic=yes is enabled. (closes issue #14611) Reported by:
- klaus3000 Patches: patch_chan_sip_attended_transfer_1.4.23.txt
- uploaded by klaus3000 (license 65) Tested by: klaus3000 ........
- r181031 | mmichelson | 2009-03-10 19:32:40 -0500 (Tue, 10 Mar
- 2009) | 3 lines Remove unused variables. ........
- ................ r181033 | mmichelson | 2009-03-10 19:49:00 -0500
- (Tue, 10 Mar 2009) | 3 lines Add missing comment that quotes RFC
- 3891 ................
-
-2009-03-10 22:05 +0000 [r180946] Jason Parker <jparker@digium.com>
-
- * /, configure, configure.ac, autoconf/ast_prog_sed.m4,
- autoconf/ast_check_gnu_make.m4: Merged revisions 180944 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r180944 | qwell | 2009-03-10 17:03:41 -0500
- (Tue, 10 Mar 2009) | 9 lines Merged revisions 180941 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r180941 | qwell | 2009-03-10 17:02:18 -0500 (Tue, 10 Mar
- 2009) | 1 line Make things happier when using autoconf 2.62+
- ........ ................
-
-2009-03-10 14:41 +0000 [r180718-180801] Joshua Colp <jcolp@digium.com>
-
- * main/manager.c, /: Merged revisions 180800 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r180800 |
- file | 2009-03-10 11:40:38 -0300 (Tue, 10 Mar 2009) | 5 lines
- Reset the thread local string buffer when handling the UserEvent
- action. (closes issue #14593) Reported by: JimDickenson ........
-
- * channels/chan_sip.c: If a port is specified when dialing a peer
- then use it. (closes issue #14626) Reported by: acunningham
-
- * channels/chan_sip.c: Ensure that the new outgoing dialog to a
- peer is able to set the socket details, even if the default is
- present. (closes issue #14480) Reported by: jon
-
-2009-03-06 18:26 +0000 [r180582] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 180579 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r180579 | mmichelson | 2009-03-06 12:25:44 -0600
- (Fri, 06 Mar 2009) | 9 lines Merged revisions 180567 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r180567 | mmichelson | 2009-03-06 12:23:09 -0600 (Fri,
- 06 Mar 2009) | 2 lines Make compilation succeed in dev-mode when
- IMAP storage is enabled. ........ ................
-
-2009-03-06 Leif Madsen <lmadsen@digium.com>
-
- * Release 1.6.0.7-rc1
-
-2009-03-06 17:28 +0000 [r180535] David Vossel <dvossel@digium.com>
-
- * main/enum.c, /: Merged revisions 180534 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r180534 | dvossel | 2009-03-06 11:26:38 -0600 (Fri, 06 Mar 2009)
- | 15 lines Merged revisions 180532 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r180532 | dvossel | 2009-03-06 11:19:55 -0600 (Fri, 06 Mar 2009)
- | 9 lines Fix handling of backreferences for ENUM lookups enum.c
- did not handle regex backtraces correctly. The '\1' in the regex
- is a backreference that requires a pattern match to be inserted.
- The way the code used to work is that it would find the
- backreference and insert the entire input string minus the '+'.
- This is incorrect. The regexec() function takes in a variable
- called pmatch which is an array of structs containing the start
- and end indexes for each backreference substring. The original
- code actually passed the pmatch array pointer into regexec but
- never did anything with it. Now when a backtrace is found, the
- backtrace number is looked up in the pmatch array and the correct
- substring is inserted. (closes issue #14576) Reported by:
- chris-mac Review: http://reviewboard.digium.com/r/187/ ........
- ................
-
-2009-03-05 23:28 +0000 [r180404-180466] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 180465 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r180465 | mmichelson | 2009-03-05 17:26:58 -0600
- (Thu, 05 Mar 2009) | 22 lines Merged revisions 180464 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r180464 | mmichelson | 2009-03-05 17:26:11 -0600 (Thu, 05 Mar
- 2009) | 16 lines [IMAP] Fix message retrieval issues when
- identical mailbox names were defined in separate contexts. There
- was a fix put in a while back so that an X-Asterisk-VM-Context
- message header was added to stored IMAP voicemails. This would
- allow for us to differentiate if the same mailbox name was used
- in multiple contexts. The problem still left was that not all
- places where messages were retrieved actually attempted to use
- this header for information when retrieving messages. This commit
- fixes that so that MWI and message retrieval from VoiceMailMain
- work as expected. (closes issue #13853) Reported by: vicks1
- Patches: 13853_v2.patch uploaded by mmichelson (license 60)
- Tested by: lmadsen ........ ................
-
- * apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged
- revisions 180383 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r180383 | mmichelson | 2009-03-05 13:14:14 -0600 (Thu, 05 Mar
- 2009) | 31 lines Merged revisions 180380 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r180380 | mmichelson | 2009-03-05 12:58:48 -0600 (Thu, 05 Mar
- 2009) | 25 lines Fix broken mailbox parsing when searchcontexts
- option is enabled. When using the searchcontexts option in
- voicemail.conf, the code made the assumption that all mailbox
- names defined were unique across all contexts. However, the code
- did nothing to actually enforce this assumption, nor did it do
- anything to alert a user that he may have created an ambiguity in
- his voicemail.conf file by defining the same mailbox name in
- multiple contexts. With this change, we now will issue a nice
- long warning if searchcontexts is on and we encounter the same
- mailbox name in multiple contexts and ignore any duplicates after
- the first box. Whether searchcontexts is enabled or not, if we
- come across a duplicate mailbox in the same context, then we will
- issue a warning and ignore the duplicated mailbox. I have also
- added a small note to voicemail.conf.sample in the explanation
- for searchcontexts explaining that you cannot define the same
- mailbox in multiple contexts if you have enabled the option.
- (closes issue #14599) Reported by: lmadsen Patches: 14599.patch
- uploaded by mmichelson (license 60) (with slight modification)
- Tested by: lmadsen ........ ................
-
-2009-03-05 18:36 +0000 [r180377] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/rtp.c, main/frame.c, /, include/asterisk/frame.h: Merged
- revisions 180373 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r180373 | kpfleming | 2009-03-05 12:29:38 -0600 (Thu, 05 Mar
- 2009) | 15 lines Merged revisions 180372 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r180372 | kpfleming | 2009-03-05 12:22:16 -0600 (Thu, 05 Mar
- 2009) | 9 lines Fix problems when RTP packet frame size is
- changed During some code analysis, I found that calling
- ast_rtp_codec_setpref() on an ast_rtp session does not work as
- expected; it does not adjust the smoother that may on the RTP
- session, in fact it summarily drops it, even if it has data in
- it, even if the current format's framing size has not changed.
- This is not good. This patch changes this behavior, so that if
- the packetization size for the current format changes, any
- existing smoother is safely updated to use the new size, and if
- no smoother was present, one is created. A new API call for
- smoothers, ast_smoother_reconfigure(), was required to implement
- these changes. Review: http://reviewboard.digium.com/r/184/
- ........ ................
-
-2009-03-04 19:25 +0000 [r180121-180196] Joshua Colp <jcolp@digium.com>
-
- * /, main/callerid.c: Merged revisions 180195 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r180195 | file | 2009-03-04 15:24:59 -0400 (Wed, 04 Mar 2009) |
- 11 lines Merged revisions 180194 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r180194 | file | 2009-03-04 15:22:50 -0400 (Wed, 04 Mar 2009) | 4
- lines Look for the number in a callerid string starting from the
- end. This way a value using <> can exist in the name portion.
- (issue #AST-194) ........ ................
-
- * apps/app_dial.c, /: Merged revisions 180120 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r180120 |
- file | 2009-03-04 10:39:28 -0400 (Wed, 04 Mar 2009) | 7 lines
- Remove duplicate 'k' and 'K' Dial options. (closes issue #14601)
- Reported by: alecdavis Patches: app_dial.optionk.diff.txt
- uploaded by alecdavis (license 585) ........
-
-2009-03-03 23:35 +0000 [r180078] David Vossel <dvossel@digium.com>
-
- * main/channel.c, include/asterisk/app.h, apps/app_read.c, /,
- main/app.c: Merged revisions 180032 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r180032 |
- dvossel | 2009-03-03 17:21:18 -0600 (Tue, 03 Mar 2009) | 14 lines
- app_read does not break from prompt loop with user terminated
- empty string In app.c, ast_app_getdata is called to stream the
- prompts and receive DTMF input. If ast_app_getdata() receives an
- empty string caused by the user inputing the end of string
- character, in this case '#', it should break from the prompt loop
- and return to app_read, but instead it cycles through all the
- prompts. I've added a return value for this special case in
- ast_readstring() which uses an enum I've delcared in apps.h. This
- enum is now used as a return value for ast_app_getdata(). (closes
- issue #14279) Reported by: Marquis Patches: fix_app_read.patch
- uploaded by Marquis (license 32) read-ampersanmd.patch2 uploaded
- by dvossel (license 671) Tested by: Marquis, dvossel Review:
- http://reviewboard.digium.com/r/177/ ........
-
-2009-03-03 23:26 +0000 [r180058] Steve Murphy <murf@digium.com>
-
- * main/ast_expr2.fl, main/ast_expr2.c, utils/Makefile,
- utils/expr2.testinput, /, main/ast_expr2.h, main/ast_expr2.y,
- main/ast_expr2f.c: Merged revisions 179973 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r179973 | murf | 2009-03-03 15:12:02 -0700 (Tue, 03 Mar 2009) |
- 33 lines Merged revisions 179807 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 I had some
- work to do to port these changes to trunk; the check_expr stuff
- hasn't been updated here for quite some time, it appears. I added
- some more tests to the check_expr2 suite. I had to play around
- with the makefile a bit, etc. I added STANDALONE2 #ifdefs to
- ast_expr2.y so as not to conflict structure with aelparse.
- ........ r179807 | murf | 2009-03-03 11:11:34 -0700 (Tue, 03 Mar
- 2009) | 19 lines These changes allow AEL to better check ${}
- constructs within $[...], that are concatenated with text. I
- modified and added rules in ast_expr2.fl to better handle the
- concatenations. I added some default routines to ast_expr2.y so
- the standalone would compile. It also looks like I haven't run
- this thru bison since 2.1, so it's good to get this updated. The
- Makefile has comments added now for check_expr2 and check_expr to
- explain what they are for, and how to run them. The testexpr2s
- stuff has been removed, in favor of check_expr2. expr2.testinput
- has been updated to include the two expressions that inspired
- these changes (from mcnobody on #asterisk this morning) The
- regression has been run and all looks well. ........
- ................
-
-2009-03-03 22:49 +0000 [r179971-180008] Mark Michelson <mmichelson@digium.com>
-
- * /, configs/queues.conf.sample, apps/app_queue.c: Merged revisions
- 180007 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r180007 | mmichelson | 2009-03-03 16:49:07 -0600 (Tue, 03 Mar
- 2009) | 22 lines Merged revisions 180006 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r180006 | mmichelson | 2009-03-03 16:48:18 -0600 (Tue, 03 Mar
- 2009) | 17 lines Clarify some documentation of queues.conf.sample
- It had always been possible to explicitly specify a "blank" value
- for a sound file in queues.conf and have no sound played back.
- The problem with this is that it would result in some ugly CLI
- warnings from file.c. This commit introduces a check when playing
- a file in app_queue to see if the name of the file is zero-length
- and return early if that is the case. Also, the ability to
- specify the blank sound files in queues.conf is now mentioned
- more clearly in queues.conf.sample (closes issue #14227) Reported
- by: caspy ........ ................
-
- * apps/app_queue.c: Fix a memory leak when updating a realtime
- member field. This was discovered while looking at issue #14353
-
-2009-03-03 18:29 +0000 [r179842] Joshua Colp <jcolp@digium.com>
-
- * /, main/features.c: Merged revisions 179841 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r179841 | file | 2009-03-03 14:28:46 -0400 (Tue, 03 Mar 2009) |
- 16 lines Merged revisions 179840 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r179840 | file | 2009-03-03 14:27:09 -0400 (Tue, 03 Mar 2009) | 9
- lines Do not assume that the bridge_cdr is still attached to the
- channel when the 'h' exten is finished executing. It is possible
- for a masquerade operation to occur when the 'h' exten is
- operating. This operation moves the CDR records around causing
- the bridge_cdr to no longer exist on the channel where it is
- expected to. We can not safely modify it afterwards because of
- this, so don't even try. (closes issue #14564) Reported by: meric
- ........ ................
-
-2009-03-03 16:48 +0000 [r179743] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /: Merged revisions 179742 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r179742 | russell | 2009-03-03 10:47:28 -0600 (Tue, 03 Mar 2009)
- | 14 lines Merged revisions 179741 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r179741 | russell | 2009-03-03 10:45:46 -0600 (Tue, 03 Mar 2009)
- | 6 lines Ensure chan->fdno always gets reset to -1 after
- handling a channel fd event. Since setting fdno to -1 had to be
- moved, a couple of other code paths that do process an fd event
- return early and do not pass through the code path where it was
- moved to. So, set it to -1 in a few other places, too. ........
- ................
-
-2009-03-03 14:40 +0000 [r179673] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /: Merged revisions 179672 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r179672 | file | 2009-03-03 10:40:04 -0400 (Tue, 03 Mar 2009) |
- 10 lines Merged revisions 179671 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r179671 | file | 2009-03-03 10:38:09 -0400 (Tue, 03 Mar 2009) | 3
- lines Move where fdno is set to the default value to *after* the
- read callback of the channel driver is called. We have to do this
- as the underlying channel driver may need the fdno value to
- determine what to read. ........ ................
-
-2009-03-03 13:55 +0000 [r179610] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /: Merged revisions 179609 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r179609 | russell | 2009-03-03 07:54:41 -0600 (Tue, 03 Mar 2009)
- | 17 lines Merged revisions 179608 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r179608 | russell | 2009-03-03 07:53:52 -0600 (Tue, 03 Mar 2009)
- | 9 lines Make it easier to detect an improper call to
- ast_read(). When you call ast_waitfor() on a channel, the index
- into the channel fds array that holds the file descriptor that
- poll() determines has input available is stored in fdno. This
- patch clears out this value after a call to ast_read() and also
- reports errors if ast_read() is called without an fdno set. From
- a discussion on the asterisk-dev list. ........ ................
-
-2009-03-03 00:03 +0000 [r179538] Jeff Peeler <jpeeler@digium.com>
-
- * main/channel.c, /: Merged revisions 179537 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r179537 | jpeeler | 2009-03-02 18:01:51 -0600 (Mon, 02 Mar 2009)
- | 21 lines Merged revisions 179536 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r179536 | jpeeler | 2009-03-02 17:54:39 -0600 (Mon, 02 Mar 2009)
- | 15 lines Fix bridging regression from commit 176701 This fixes
- a bad regression where the bridge would exit after an attended
- transfer was made. The problem was due to nexteventts getting set
- after the masquerade which caused the bridge to return
- AST_BRIDGE_COMPLETE. (closes issue #14315) Reported by:
- tim_ringenbach ........ ................
-
-2009-03-02 23:38 +0000 [r179534] Russell Bryant <russell@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 179533 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r179533 | russell | 2009-03-02 17:36:38 -0600 (Mon, 02 Mar 2009)
- | 48 lines Merged revisions 179532 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r179532 | russell | 2009-03-02 17:34:13 -0600 (Mon, 02 Mar 2009)
- | 40 lines Move ast_waitfor() down to avoid the results of the
- API call becoming stale. This call to ast_waitfor() was being
- done way too soon in this section of code. Specifically, there
- was code in between the call to waitfor and the code that uses
- the result that puts the channel in autoservice. By putting the
- channel in autoservice, the previous results of ast_waitfor()
- become meaningless, as the autoservice thread will do it's own
- ast_waitfor() and ast_read() on the channel. So, when we came
- back out of autoservice and eventually hit the block of code that
- calls ast_read() on the channel, there may not actually be any
- input on the channel available. Even though the previous call to
- ast_waitfor() in app_meetme said there was input, the autoservice
- thread has since serviced the channel for some period of time.
- This bug manifested itself while dvossel was doing some testing
- of MeetMe in Asterisk trunk. He was using the timerfd timing
- module. When the code hit ast_read() erroneously, it determined
- that it must have been called because of input on the timer fd,
- as chan->fdno was set to AST_TIMING_FD, since that was the cause
- of the last legitimate call to ast_read() done by autoservice. In
- this test, an IAX2 channel was calling into the MeetMe
- conference. It was _much_ more likely to be seen with an IAX2
- channel because of the way audio is handled. Every audio frame
- that comes in results in a call to ast_queue_frame(), which then
- uses ast_timer_enable_continuous() to notify the channel thread
- that a frame is waiting to be handled. So, the chances of
- ast_waitfor() indicating that a channel needs servicing due to a
- timer event on an IAX2 event is very high. Finally, it is
- interesting to note that if a different timing interface was
- being used, this bug would probably not be noticed. When
- ast_read() is called and erroneously thinks that there is a timer
- event to handle, it calls the ast_timer_ack() function. The
- pthread and dahdi timing modules handle the ack() function being
- called when there is no event by simply ignoring it. In the case
- of the timerfd module, it results in a read() on the timer fd
- that will block forever, as there is no data to read. This caused
- Asterisk to lock up very quickly. Thanks to dvossel and
- mmichelson for the fun debugging session. :-) ........
- ................
-
-2009-03-02 23:15 +0000 [r179473] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 151464 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r151464 |
- mmichelson | 2008-10-21 18:54:41 -0500 (Tue, 21 Oct 2008) | 11
- lines Make the sip_standard_port function more granular by
- allowing separate type and port arguments. This is necessary
- because when building our From and Contact headers, we need to be
- absolutely sure that we are placing our source port there and not
- the peer's source port. (closes issue #12761) Reported by:
- asbestoshead Patches: patch-chan-sip-contact-port.txt uploaded by
- asbestoshead (license 455) ........
-
-2009-03-02 23:11 +0000 [r179470] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/app.c: Merged revisions 179469 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r179469 | tilghman | 2009-03-02 17:10:18 -0600 (Mon, 02 Mar 2009)
- | 17 lines Merged revisions 179468 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r179468 | tilghman | 2009-03-02 17:09:01 -0600 (Mon, 02 Mar 2009)
- | 10 lines When ending a recording with silence detection,
- remember to reduce the duration. The end of the recording is
- correspondingly trimmed, but the duration was not trimmed by the
- number of seconds trimmed, so the saved duration was necessarily
- longer than the actual soundfile duration. (closes issue #14406)
- Reported by: sasargen Patches: 20090226__bug14406.diff.txt
- uploaded by tilghman (license 14) Tested by: sasargen ........
- ................
-
-2009-03-02 23:02 +0000 [r179463] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /: Merged revisions 179462 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r179462 | russell | 2009-03-02 17:00:30 -0600 (Mon, 02 Mar 2009)
- | 16 lines Merged revisions 179461 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r179461 | russell | 2009-03-02 16:58:18 -0600 (Mon, 02 Mar 2009)
- | 8 lines Ensure that only one thread is calling ast_settimeout()
- on a channel at a time. For example, with an IAX2 channel, you
- can have both the channel thread and the chan_iax2 processing
- threads calling this function, and doing so twice at the same
- time is a bad thing. (Found in a debugging session with dvossel
- and mmichelson) ........ ................
-
-2009-03-02 20:17 +0000 [r179402] Jason Parker <jparker@digium.com>
-
- * /, main/editline/configure, main/editline/np/unvis.c,
- main/editline/sys.h, main/editline/configure.in: Merged revisions
- 179396 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r179396 | qwell | 2009-03-02 14:16:51 -0600 (Mon, 02 Mar 2009) |
- 9 lines Merged revisions 179395 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r179395 | qwell | 2009-03-02 14:14:57 -0600 (Mon, 02 Mar 2009) |
- 1 line Remove several silly warnings in editline. One about a
- broken preprocessor directive, and another about strlcpy/strlcat.
- (closes issue #14264) Reported by: dimas ........
- ................
-
-2009-03-02 17:58 +0000 [r179360-179363] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_stack.c: KeepAlive application no longer exists, so fix
- gosub implementation to not use it. (closes issue #14571)
- Reported by: zktech Patches: 20090302__bug14571.diff.txt uploaded
- by tilghman (license 14) Tested by: tilghman
-
- * cdr/cdr_sqlite3_custom.c: If cdr registration somehow succeeds
- without a config file, don't crash. (closes issue #14563)
- Reported by: alerios
-
-2009-03-01 22:07 +0000 [r179220-179222] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Add error checking when updating the "paused"
- field of a realtime queue member. This code already existed in
- trunk and 1.6.1, but was not in 1.6.0 prior to this commit.
- (closes issue #14338) Reported by: fiddur Patches: 14338.patch
- uploaded by mmichelson (license 60) Tested by: fiddur
-
- * /, channels/chan_sip.c: Merged revisions 179219 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r179219 |
- mmichelson | 2009-03-01 15:45:08 -0600 (Sun, 01 Mar 2009) | 18
- lines Properly free memory and remove scheduler entries when a
- transmission failure occurs. Previously, only the "data" field of
- the sip_pkt created during __sip_reliable_xmit was freed when
- XMIT_FAILURE was returned by __sip_xmit. When retrans_pkt was
- called, this inevitably resulted in the reading and writing of
- freed memory. XMIT_FAILURE is a condition meaning that we don't
- want to attempt resending the packet at all. The proper action to
- take is to remove the scheduler entry we just created, free the
- packet's data as well as the packet itself, and unlink it from
- the list of packets on the sip_pvt structure. (closes issue
- #14455) Reported by: Nick_Lewis Patches: 14455.patch uploaded by
- mmichelson (license 60) Tested by: Nick_Lewis ........
-
-2009-02-27 21:33 +0000 [r179162] Tilghman Lesher <tlesher@digium.com>
-
- * cdr/cdr_sqlite3_custom.c, /: Merged revisions 179161 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r179161 | tilghman | 2009-02-27 15:32:13 -0600 (Fri, 27 Feb 2009)
- | 3 lines If config file is blank, don't load module. (Closes
- issue #14563) ........
-
-2009-02-27 19:05 +0000 [r179058] Jason Parker <jparker@digium.com>
-
- * /, doc/tex/channelvariables.tex: Merged revisions 179057 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r179057 | qwell | 2009-02-27 13:04:57 -0600 (Fri, 27 Feb
- 2009) | 8 lines Update documentation for DIALEDTIME and
- ANSWEREDTIME variables. (closes issue #14566) Reported by:
- klaus3000 Patches: ANSWEREDTIME-1.4-patch.txt uploaded by
- klaus3000 (license 65) ANSWEREDTIME-trunk-patch.txt uploaded by
- klaus3000 (license 65) ........
-
-2009-02-27 03:52 +0000 [r178987] Steve Murphy <murf@digium.com>
-
- * configs/features.conf.sample, /, main/features.c: Merged
- revisions 178986 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r178986 | murf | 2009-02-26 20:45:58 -0700 (Thu, 26 Feb 2009) |
- 26 lines Merged revisions 178956 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 In this
- case, it's just a matter of reducing the default timeouts from
- 2000 to 1000 msec, as the max def feature digit timeout is no
- longer halved. ........ r178956 | murf | 2009-02-26 14:27:32
- -0700 (Thu, 26 Feb 2009) | 18 lines This change moves the default
- feature digit timeout to 1000 ms from the previous default of
- 500. As per bug 14515, a dev discussion arrived at a "mediated
- concensus" of a default feature digit timeout of 1.0 sec. Some
- voted for 1300; ctooley thought 1500 for distracted phone users
- in phone booths; kpfleming put his foot down at 1.0 sec. Users
- who found the previous default max delay of 250 msec perfect, are
- welcome to override the new default. Notice that I said that 250
- msec was the default; wait a minute, you might say, the config
- file said it was 500 msec!; well, because of the bug fix for
- 14515, we found that 500 msec was actually enforcing a max of
- 250. The bug fix would restore 500 msec, but we felt even that
- was a bit tight for most users... 2000 msec was pushed earlier by
- mmichelson, so that reduces to 1000 msec after the bug fix.
- Enjoy! ........ ................
-
-2009-02-26 17:50 +0000 [r178874] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 178871 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r178871 | dvossel | 2009-02-26 11:46:12 -0600 (Thu, 26 Feb 2009)
- | 6 lines IAX2 prune realtime, minor tweak to last fix A return
- statement was missing which caused unexpected cli output. issue
- #14479 ........
-
-2009-02-26 17:29 +0000 [r178866] Steve Murphy <murf@digium.com>
-
- * /, main/features.c: Merged revisions 178828 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r178828 | murf | 2009-02-26 10:22:11 -0700 (Thu, 26 Feb 2009) |
- 34 lines Merged revisions 178804 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r178804 | murf | 2009-02-26 10:09:03 -0700 (Thu, 26 Feb 2009) |
- 28 lines This patch prevents the feature detection timeout from
- being cut in half. Because the ast_channel_bridge() call will
- return 0 and pass a frame pointer for both DTMF_BEGIN and
- DTMF_END, the feature_timer field in hte config struct is getting
- decremented twice, which effectively cuts the digittimeout in
- half. I added conditions to the if statement to only let DTMF_END
- frames to flow thru, which solved the problem. Also, when the
- frame pointer is null, let control flow thru-- this usually
- happens on timeouts. I added a comment to the code to explain
- what's going on and why. Many thanks to sodom for reporting this
- problem. Personnally, it always seemed like something was wrong
- with the featuredigittimeout, but I never could quite decide
- what... and was too busy to investigate. This bug forced the
- issue, and now we know. Sodom had other issues in 14515, but I
- couldn't reproduce them. If he still has problems, and wants to
- get them solved, he is welcome to reopen 14515. (closes issue
- #14515) Reported by: sodom Patches: 14515.patch uploaded by murf
- (license 17) Tested by: murf, sodom ........ ................
-
-2009-02-26 16:01 +0000 [r178768] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 178767 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r178767 | dvossel | 2009-02-26 09:50:22 -0600 (Thu, 26 Feb 2009)
- | 8 lines IAX2 prune realtime fix Iax2 prune realtime had issues.
- If "iax2 prune realtime all" was called, it would appear like the
- command was successful, but in reality nothing happened. This is
- because the reload that was supposed to take place checks the
- config files, sees no changes, and does nothing. If there had
- been a change in the the config file, the realtime users would
- have been marked for deletion and everything would have been
- fine. Now prune_users() and prune_peers() are called instead of
- reload_config() to prune all users/peers that are realtime. These
- functions remove all users/peers with the rtfriend and delme
- flags set. iax2_prune_realtime() also lacked the code to properly
- delete a single friend. For example. if iax2 prune realtime
- <friend> was called, only the peer instance would be removed. The
- user would still remain. (closes issue #14479) Reported by:
- mousepad99 Review: http://reviewboard.digium.com/r/176/ ........
-
-2009-02-25 12:46 +0000 [r178510] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c, /: Merged revisions 178509 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r178509 | russell | 2009-02-25 06:45:30 -0600 (Wed, 25 Feb 2009)
- | 10 lines Merged revisions 178508 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r178508 | russell | 2009-02-25 06:43:36 -0600 (Wed, 25 Feb 2009)
- | 2 lines Update the copyright year for the main page of the
- doxygen documentation. ........ ................
-
-2009-02-24 23:28 +0000 [r178382-178447] Tilghman Lesher <tlesher@digium.com>
-
- * configs/extensions.conf.sample, /: Merged revisions 178446 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r178446 | tilghman | 2009-02-24 17:27:23 -0600
- (Tue, 24 Feb 2009) | 12 lines Merged revisions 178445 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r178445 | tilghman | 2009-02-24 17:25:24 -0600 (Tue, 24 Feb 2009)
- | 5 lines Add section about the #exec command in configuration
- files. (closes issue #14540) Reported by: jtodd Patch by: jtodd,
- with additional notes by tilghman (license 14) ........
- ................
-
- * main/asterisk.c, /: Merged revisions 178381 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r178381 |
- tilghman | 2009-02-24 14:52:44 -0600 (Tue, 24 Feb 2009) | 2 lines
- Apparently, a void cast doesn't override warn_unused_result.
- ........
-
-2009-02-24 20:43 +0000 [r178378] Russell Bryant <russell@digium.com>
-
- * main/rtp.c, /: Merged revisions 178374 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r178374 | russell | 2009-02-24 14:39:57 -0600 (Tue, 24 Feb 2009)
- | 14 lines Merged revisions 178373 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r178373 | russell | 2009-02-24 14:36:19 -0600 (Tue, 24 Feb 2009)
- | 6 lines Only set dtmfcount on BEGIN, and ensure it gets reset
- to 0 properly. (issue #14460) Reported by: moliveras Tested by:
- russell ........ ................
-
-2009-02-24 20:40 +0000 [r178343-178376] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c, /: Merged revisions 178375 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r178375 |
- tilghman | 2009-02-24 14:40:02 -0600 (Tue, 24 Feb 2009) | 2 lines
- The 3 possible errors with pipe(2) are all impossible in this
- situation. ........
-
- * main/asterisk.c, /, utils/astcanary.c: Merged revisions 178342
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r178342 | tilghman | 2009-02-24 14:06:48 -0600 (Tue, 24
- Feb 2009) | 2 lines Use a SIGPIPE to kill the process, instead of
- depending upon the astcanary process being inherited by init.
- ........
-
-2009-02-24 18:05 +0000 [r178306] Terry Wilson <twilson@digium.com>
-
- * apps/app_dahdiras.c: Change include order to make compile on
- Centos 5 with DAHDI If BIT_TYPES_DEFINED gets defined before
- linux/types.h is included, the __s32 type doesn't get defined
-
-2009-02-24 17:53 +0000 [r178304] Tilghman Lesher <tlesher@digium.com>
-
- * /, utils/astcanary.c: Merged revisions 178303 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r178303 |
- tilghman | 2009-02-24 11:51:36 -0600 (Tue, 24 Feb 2009) | 7 lines
- Cause astcanary to exit if Asterisk exits abnormally and doesn't
- kill astcanary. Also, add some documentation supporting the use
- of astcanary. (closes issue #14538) Reported by: KNK Patches:
- asterisk-1.6.x-astcanary.diff uploaded by KNK (license 545)
- ........
-
-2009-02-24 15:20 +0000 [r178224] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 178213 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r178213 | file | 2009-02-24 11:18:38 -0400 (Tue, 24 Feb 2009) |
- 16 lines Merged revisions 178205 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r178205 | file | 2009-02-24 11:16:07 -0400 (Tue, 24 Feb 2009) | 9
- lines Skip check for extension when subscribing for MWI. Since
- the remote side is not actually subscribing to a specific
- extension when subscribing for MWI just skip the check to see if
- the extension exists. They can't use it to specify the mailbox
- either since we require configuration of that in sip.conf (closes
- issue #14531) Reported by: festr ........ ................
-
-2009-02-23 23:17 +0000 [r178145] Russell Bryant <russell@digium.com>
-
- * main/rtp.c, /: Merged revisions 178142 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r178142 | russell | 2009-02-23 17:11:37 -0600 (Mon, 23 Feb 2009)
- | 22 lines Merged revisions 178141 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r178141 | russell | 2009-02-23 17:09:01 -0600 (Mon, 23 Feb 2009)
- | 14 lines Fix infinite DTMF when a BEGIN is received without an
- END. This commit is related to rev 175124 of 1.4 where a previous
- attempt was made to fix this problem. The problem with the
- previous patch was that the inserted code needed to go _before_
- setting the lastrxts to the current timestamp. Because those were
- the same, the dtmfcount variable was never decremented, and so
- the END was never sent. In passing, I removed the dtmfsamples
- variable which was completed unused. I also removed a redundant
- setting of the lastrxts variable. (closes issue #14460) Reported
- by: moliveras ........ ................
-
-2009-02-23 Leif Madsen <lmadsen@digium.com>
-
- * Released 1.6.0.6
-
-2009-02-13 Leif Madsen <lmadsen@digium.com>
-
- * Released 1.6.0.6-rc1
-
-2009-02-13 16:43 +0000 [r175550] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_record.c: Merged revisions 175549 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r175549 |
- file | 2009-02-13 12:41:15 -0400 (Fri, 13 Feb 2009) | 4 lines Add
- an option to keep the recorded file upon hangup. (closes issue
- #14341) Reported by: fnordian ........
-
-2009-02-12 21:41 +0000 [r175369] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 175368 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r175368 |
- russell | 2009-02-12 15:41:01 -0600 (Thu, 12 Feb 2009) | 2 lines
- Remove useless string copy, and make sscanf safe again ........
-
-2009-02-12 21:27 +0000 [r175347] Tilghman Lesher <tlesher@digium.com>
-
- * main/udptl.c, /: Merged revisions 175334 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r175334 | tilghman | 2009-02-12 15:25:14 -0600 (Thu, 12 Feb 2009)
- | 16 lines Merged revisions 175311 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r175311 | tilghman | 2009-02-12 15:19:40 -0600 (Thu, 12 Feb 2009)
- | 9 lines Fix crashes when receiving certain T.38 packets. Also,
- increase the maximum size of T.38 packets and warn users when
- they try to set the limits above those maximums. (closes issue
- #13050) Reported by: schern Patches: 20090212__bug13050.diff.txt
- uploaded by Corydon76 (license 14) Tested by: schern ........
- ................
-
-2009-02-12 20:59 +0000 [r175299-175301] Jeff Peeler <jpeeler@digium.com>
-
- * main/features.c: Fix mistake in merging conflict from 175299.
-
- * /, main/features.c: Merged revisions 175298 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r175298 | jpeeler | 2009-02-12 14:48:56 -0600 (Thu, 12 Feb 2009)
- | 15 lines Merged revisions 175294 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r175294 | jpeeler | 2009-02-12 14:34:36 -0600 (Thu, 12 Feb 2009)
- | 9 lines Fix ParkedCall event information for From field in the
- case of a blind transfer If the parker information can not be
- obtained from the peer, try and see if the BLINDTRANSFER channel
- variable has been set. Previously, a blind transfer to the
- ParkAndAnnounce app would return nothing for the From. Closes
- AST-189 ........ ................
-
-2009-02-12 20:46 +0000 [r175256-175296] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 175295 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r175295 |
- russell | 2009-02-12 14:45:47 -0600 (Thu, 12 Feb 2009) | 2 lines
- Avoid using ast_strdupa() in a loop. ........
-
- * build_tools/cflags.xml, /: Merged revisions 175255 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r175255 | russell | 2009-02-12 13:11:08 -0600 (Thu, 12 Feb 2009)
- | 4 lines Don't enable something by default that has a dependency
- on something _not_ enabled by default. menuselect was not happy
- with this. ........
-
-2009-02-12 18:00 +0000 [r175189] Jeff Peeler <jpeeler@digium.com>
-
- * /, main/features.c: Merged revisions 175188 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r175188 | jpeeler | 2009-02-12 12:00:11 -0600 (Thu, 12 Feb 2009)
- | 12 lines Merged revisions 175187 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r175187 | jpeeler | 2009-02-12 11:57:10 -0600 (Thu, 12 Feb 2009)
- | 6 lines Fix crash in event of failed attempt to transfer to
- parking The peer may not necessarily exist, such as in the case
- of a transfer to ParkAndAnnounce. In this case don't try to play
- a sound to it. ........ ................
-
-2009-02-12 17:03 +0000 [r175126] Russell Bryant <russell@digium.com>
-
- * main/rtp.c, /: Merged revisions 175125 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r175125 | russell | 2009-02-12 10:57:25 -0600 (Thu, 12 Feb 2009)
- | 35 lines Merged revisions 175124 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r175124 | russell | 2009-02-12 10:51:13 -0600 (Thu, 12 Feb 2009)
- | 27 lines Don't send DTMF for infinite time if we do not receive
- an END event. I thought that this was going to end up being a
- pretty gnarly fix, but it turns out that there was actually
- already a configuration option in rtp.conf, dtmftimeout, that was
- intended to handle this situation. However, in between Asterisk
- 1.2 and Asterisk 1.4, the code that processed the option got
- lost. So, this commit brings it back to life. The default timeout
- is 3 seconds. However, it is worth noting that having this be
- configurable at all is not really the recommended behavior in RFC
- 2833. From Section 3.5 of RFC 2833: Limiting the time period of
- extending the tone is necessary to avoid that a tone "gets
- stuck". Regardless of the algorithm used, the tone SHOULD NOT be
- extended by more than three packet interarrival times. A slight
- extension of tone durations and shortening of pauses is generally
- harmless. Three seconds will pretty much _always_ be far more
- than three packet interarrival times. However, that behavior is
- not required, so I'm going to leave it with our legacy behavior
- for now. Code from svn/asterisk/team/russell/issue_14460 (closes
- issue #14460) Reported by: moliveras ........ ................
-
-2009-02-12 16:33 +0000 [r175122] Mark Michelson <mmichelson@digium.com>
-
- * main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions
- 175121 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r175121 |
- mmichelson | 2009-02-12 10:28:06 -0600 (Thu, 12 Feb 2009) | 11
- lines Make lock information for ao2_trylock be more useful and
- gnarly Core show locks information involving an ao2_trylock did
- not show the function that called ao2_trylock, but would instead
- show ao2_trylock as the source of the lock. This is not useful
- when trying to debug locking issues. One bizarre note is that
- this logic is already in 1.4 but somehow did not get merged to
- trunk or the 1.6.X branches. ........
-
-2009-02-12 14:27 +0000 [r175059-175090] Philippe Sultan <philippe.sultan@gmail.com>
-
- * /, channels/chan_gtalk.c: Merged revisions 175089 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r175089 | phsultan | 2009-02-12 15:25:03 +0100 (Thu, 12 Feb 2009)
- | 6 lines Issue a warning message if our candidate's IP is the
- loopback address. (closes issue #13985) Reported by: jcovert
- Tested by: phsultan ........
-
- * /, channels/chan_gtalk.c: Merged revisions 175058 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r175058 | phsultan | 2009-02-12 11:31:36 +0100
- (Thu, 12 Feb 2009) | 20 lines Merged revisions 175029 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r175029 | phsultan | 2009-02-12 11:16:21 +0100 (Thu, 12 Feb 2009)
- | 12 lines Set the initiator attribute to lowercase in our
- replies when receiving calls. This attribute contains a JID that
- identifies the initiator of the GoogleTalk voice session. The
- GoogleTalk client discards Asterisk's replies if the initiator
- attribute contains uppercase characters. (closes issue #13984)
- Reported by: jcovert Patches: chan_gtalk.2.patch uploaded by
- jcovert (license 551) Tested by: jcovert ........
- ................
-
-2009-02-11 23:04 +0000 [r174765-174949] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 174948 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r174948 | mmichelson | 2009-02-11 17:03:08 -0600 (Wed, 11 Feb
- 2009) | 35 lines Fix odd "thank you" sound playing behavior in
- app_queue.c If someone has configured the queue to play an
- position or holdtime announcement, then it is odd and potentially
- unexpected to hear a "Thank you for your patience" sound when no
- position or holdtime was actually announced. This fixes the
- announcement so that the "thanks" sound is only played in the
- case that a position or holdtime was actually announced. There is
- a way that the "thank you" sound can be played without a position
- or holdtime, and that is to set announce-frequency to a value but
- keep announce-position and announce-holdtime both turned off.
- (closes issue #14227) Reported by: caspy Patches: 14227_v3.patch
- uploaded by putnopvut (license 60) Tested by: caspy
- ................
-
- * apps/app_dial.c, main/channel.c, main/pbx.c, /,
- apps/app_dictate.c, apps/app_waitforsilence.c,
- include/asterisk/channel.h: Merged revisions 174945 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r174945 | mmichelson | 2009-02-11 16:41:01 -0600 (Wed, 11 Feb
- 2009) | 29 lines Fix 'd' option for app_dial and add new option
- to Answer application The 'd' option would not work for channel
- types which use RTP to transport DTMF digits. The only way to
- allow for this to work was to answer the channel if we saw that
- this option was enabled. I realized that this may cause issues
- with CDRs, specifically with giving false dispositions and answer
- times. I therefore modified ast_answer to take another parameter
- which would tell if the CDR should be marked answered. I also
- extended this to the Answer application so that the channel may
- be answered but not CDRified if desired. I also modified
- app_dictate and app_waitforsilence to only answer the channel if
- it is not already up, to help not allow for faulty CDR answer
- times. All of these changes are going into Asterisk trunk. For
- 1.6.0 and 1.6.1, however, all the changes except for the change
- to the Answer application will go in since we do not introduce
- new features into stable branches (closes issue #14164) Reported
- by: DennisD Patches: 14164.patch uploaded by putnopvut (license
- 60) Tested by: putnopvut Review:
- http://reviewboard.digium.com/r/145 ........
-
- * apps/app_chanspy.c, /: Merged revisions 174805 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r174805 |
- mmichelson | 2009-02-10 17:17:03 -0600 (Tue, 10 Feb 2009) | 11
- lines Fix potential for stack overflows in app_chanspy.c When
- using the 'g' or 'e' options, the stack allocations that were
- used could cause a stack overflow if a spyer stayed on the line
- long enough without actually successfully spying on anyone. The
- problem has been corrected by using static buffers and copying
- the contents of the appropriate strings into them instead of
- using functions like alloca or ast_strdupa ........
-
- * main/manager.c, /: Merged revisions 174764 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r174764 |
- mmichelson | 2009-02-10 15:45:14 -0600 (Tue, 10 Feb 2009) | 21
- lines Fix an fd leak that would occur in HTTP AMI sessions The
- explanation behind this fix is a bit complicated, and I've
- already typed it up in the code as a huge comment inside of
- manager.c, so I'll give the abridged version here. We needed a
- way to separate action-specific data from session-specific data.
- Unfortunately, the only way to maintain API compatibility and to
- not have to change every single manager action was to rename the
- current mansession structure and wrap it inside a new mansession
- structure which actually contains action- specific data. (closes
- issue #14364) Reported by: awk Patches: 14364_better.patch
- uploaded by putnopvut (license 60) Tested by: putnopvut Review:
- http://reviewboard.digium.com/r/148/ ........
-
-2009-02-10 20:16 +0000 [r174711] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 174710 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r174710 |
- file | 2009-02-10 16:15:43 -0400 (Tue, 10 Feb 2009) | 4 lines
- Only decrease inringing count if above zero. (issue #13238)
- Reported by: kowalma ........
-
-2009-02-10 18:19 +0000 [r174596] Matthew Nicholson <mnicholson@digium.com>
-
- * /, main/jitterbuf.c: Merged revisions 174584 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r174584 | mnicholson | 2009-02-10 12:16:31 -0600 (Tue, 10 Feb
- 2009) | 25 lines Merged revisions 174583 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r174583 | mnicholson | 2009-02-10 11:52:42 -0600 (Tue, 10 Feb
- 2009) | 18 lines Improve behavior of jitterbuffer when
- maxjitterbuffer is set. This change improves the way the
- jitterbuffer handles maxjitterbuffer and dramatically reduces the
- number of frames dropped when maxjitterbuffer is exceeded. In the
- previous jitterbuffer, when maxjitterbuffer was exceeded, all new
- frames were dropped until the jitterbuffer is empty. This change
- modifies the code to only drop frames until maxjitterbuffer is no
- longer exceeded. Also, previously when maxjitterbuffer was
- exceeded, dropped frames were not tracked causing stats for
- dropped frames to be incorrect, this change also addresses that
- problem. (closes issue #14044) Patches: bug14044-1.diff uploaded
- by mnicholson (license 96) Tested by: mnicholson Review:
- http://reviewboard.digium.com/r/144/ ........ ................
-
-2009-02-10 15:39 +0000 [r174544] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 174543 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r174543 |
- file | 2009-02-10 11:37:07 -0400 (Tue, 10 Feb 2009) | 6 lines
- Make the logic for inuse and inringing manipluation match that of
- 1.4. The old broken logic would reset the values back to 0 during
- certain scenarios causing the wrong state to be reported. (closes
- issue #14399) Reported by: caspy (issue #13238) Reported by:
- kowalma ........
-
-2009-02-10 05:06 +0000 [r174439] Steve Murphy <murf@digium.com>
-
- * apps/app_rpt.c: For some strange reason, I didn't think 1.6.0
- needed this fix. I was wrong. Here it is.
-
-2009-02-09 17:28 +0000 [r174322-174328] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 174327 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r174327 |
- mmichelson | 2009-02-09 11:27:32 -0600 (Mon, 09 Feb 2009) | 3
- lines Fix something I messed up in the merge I just did ........
-
- * /, channels/chan_sip.c: Merged revisions 174301 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r174301 | mmichelson | 2009-02-09 11:20:55 -0600 (Mon, 09 Feb
- 2009) | 20 lines Merged revisions 174282 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r174282 | mmichelson | 2009-02-09 11:11:05 -0600 (Mon, 09 Feb
- 2009) | 12 lines Don't do an SRV lookup if a port is specified
- RFC 3263 says to do A record lookups on a hostname if a port has
- been specified, so that's what we're going to do. See section
- 4.2. (closes issue #14419) Reported by: klaus3000 Patches:
- patch_chan_sip_nosrvifport_1.4.23.txt uploaded by klaus3000
- (license 65) ........ ................
-
-2009-02-09 14:50 +0000 [r174220] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_musiconhold.c: Merged revisions 174219 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r174219 | file | 2009-02-09 10:49:24 -0400 (Mon,
- 09 Feb 2009) | 11 lines Merged revisions 174218 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r174218 | file | 2009-02-09 10:48:21 -0400 (Mon, 09 Feb 2009) | 4
- lines Don't overwrite our pointer to the music class when music
- on hold stops. We will use this if it starts again to see if we
- can resume the music where it left off. (closes issue #14407)
- Reported by: mostyn ........ ................
-
-2009-02-07 16:17 +0000 [r174151] Russell Bryant <russell@digium.com>
-
- * /, res/snmp/agent.c: Merged revisions 174149 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r174149 | russell | 2009-02-07 10:16:50 -0600 (Sat, 07 Feb 2009)
- | 10 lines Merged revisions 174148 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r174148 | russell | 2009-02-07 10:15:07 -0600 (Sat, 07 Feb 2009)
- | 2 lines Fix a race condition that could cause a crash. ........
- ................
-
-2009-02-06 23:59 +0000 [r174085] Dwayne M. Hubbard <dhubbard@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 174084 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r174084 | dhubbard | 2009-02-06 17:51:56 -0600 (Fri, 06 Feb 2009)
- | 13 lines Merged revisions 174082 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r174082 | dhubbard | 2009-02-06 17:36:03 -0600 (Fri, 06 Feb 2009)
- | 5 lines check ast_strlen_zero() before calling ast_strdupa() in
- sip_uri_headers_cmp() and sip_uri_params_cmp() The reporter
- didn't actually upload a properly-formed patch, instead a
- modified chan_sip.c file was uploaded. I created a patch to
- determine the changes, then modified the suggested changes to
- create a proper fix. The summary above is a complete description
- of the changes. (closes issue #13547) Reported by: tecnoxarxa
- Patches: chan_sip.c.gz uploaded by tecnoxarxa (license 258)
- Tested by: tecnoxarxa ........ ................
- ------------------------------------------------------------------------
-
-2009-02-06 19:29 +0000 [r173986-174042] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 174041 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r174041 | file | 2009-02-06 15:28:53 -0400 (Fri, 06 Feb 2009) | 4
- lines Don't subscribe to a mailbox on pseudo channels. It is
- futile. This solves an issue where duplicated pseudo channels
- would cause a crash because the first one would unsubscribe and
- the next one would also try to unsubscribe the same subscription.
- (closes issue #14322) Reported by: amessina ........
-
- * /, channels/chan_sip.c: Merged revisions 173974 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r173974 | file | 2009-02-06 13:18:35 -0400 (Fri, 06 Feb 2009) |
- 15 lines Merged revisions 173967-173968 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r173967 | file | 2009-02-06 13:14:15 -0400 (Fri, 06 Feb 2009) | 4
- lines Some clients do not put the call-id for replaces at the
- beginning, so support it being anywhere in the string. (closes
- issue #14350) Reported by: fhackenberger ........ r173968 | file
- | 2009-02-06 13:15:01 -0400 (Fri, 06 Feb 2009) | 2 lines Remove a
- debug message I put in by accident. ........ ................
-
-2009-02-06 16:33 +0000 [r173963] Matthew Nicholson <mnicholson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 173952 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r173952 | mnicholson | 2009-02-06 10:28:19 -0600 (Fri, 06 Feb
- 2009) | 14 lines Merged revisions 173917 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r173917 | mnicholson | 2009-02-06 10:20:23 -0600 (Fri, 06 Feb
- 2009) | 7 lines Limit the addition of the Contact header in SIP
- responses according to various SIP RFCs. (closes issue #13602)
- Reported by: hjourdain Tested by: mnicholson ........
- ................
-
-2009-02-05 23:51 +0000 [r173774-173777] Mark Michelson <mmichelson@digium.com>
-
- * configs/extensions.conf.sample, /: Merged revisions 173776 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r173776 | mmichelson | 2009-02-05 17:48:48 -0600 (Thu,
- 05 Feb 2009) | 14 lines Update extensions.conf.sample to be
- correct. In trunk, the only necessary change pointed out was that
- the call to ChanIsAvail uses an option that has been removed. For
- the 1.6.1 branch, however, it appears that the sample file is
- badly in need of updating since there are |'s used all over the
- place there. My tentative plan is just to copy trunk's sample
- config file to those branches since the info there is most
- up-to-date and should be correct for use in 1.6.1 Thanks to macli
- in #asterisk-dev for bringing this up ........
-
- * apps/app_voicemail.c, /: Merged revisions 173773 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r173773 | mmichelson | 2009-02-05 17:28:19 -0600 (Thu, 05 Feb
- 2009) | 7 lines Properly set "seen" and "unseen" flags when
- moving messages from the new to the old folder when using IMAP
- for voicemail storage (closes issue #13905) Reported by: jaroth
- Patches: foldermove_v2.patch uploaded by jaroth (license 50)
- ........
-
-2009-02-05 21:04 +0000 [r173698] Jeff Peeler <jpeeler@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 173697 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r173697 | jpeeler | 2009-02-05 15:00:26 -0600
- (Thu, 05 Feb 2009) | 18 lines Merged revisions 173696 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r173696 | jpeeler | 2009-02-05 14:47:51 -0600 (Thu, 05 Feb 2009)
- | 12 lines Add new configuration option to make shared IMAP
- mailboxes function as expected. The new option is "imapvmshareid"
- which is an ID to tag multiple mailboxes using the same IMAP
- storage location to function as one mailbox. This allows all
- messages to be retrieved for any user in the group. The patch
- alters the 'X-Asterisk-VM-Extension' header that is responsible
- for matching voicemails for a given user. (closes issue #13673)
- Reported by: howardwilkinson ........ ................
-
-2009-02-05 20:34 +0000 [r173590-173694] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 173693 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r173693 | mmichelson | 2009-02-05 14:30:45 -0600 (Thu, 05 Feb
- 2009) | 20 lines Merged revisions 173692 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r173692 | mmichelson | 2009-02-05 14:29:09 -0600 (Thu, 05 Feb
- 2009) | 12 lines Fix situations where queue members could be
- autopaused unexpectedly Specifically, this patch prevents us from
- autopausing members when we receive a busy or congestion frame
- from them. (closes issue #14376) Reported by: fiddur Patches:
- 14376.patch uploaded by putnopvut (license 60) Tested by: fiddur
- ........ ................
-
- * apps/app_mixmonitor.c, /: Merged revisions 173593 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r173593 | mmichelson | 2009-02-05 12:48:55 -0600
- (Thu, 05 Feb 2009) | 11 lines Merged revisions 173592 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r173592 | mmichelson | 2009-02-05 12:47:24 -0600 (Thu, 05 Feb
- 2009) | 3 lines Add some missing cleanup to app_mixmonitor
- ........ ................
-
- * apps/app_mixmonitor.c, /: Merged revisions 173589 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r173589 | mmichelson | 2009-02-05 12:34:06 -0600
- (Thu, 05 Feb 2009) | 33 lines Merged revisions 173559 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r173559 | mmichelson | 2009-02-05 11:34:33 -0600 (Thu, 05 Feb
- 2009) | 25 lines Fix a problem where a channel pointer becomes
- invalid due to masquerading or hanging up. app_mixmonitor runs
- its own thread to monitor the channel's activity and write the
- mixed audio to a file. Since this thread runs independently of
- the channel, it is possible that the mixmonitor thread's channel
- pointer will point to freed memory when the channel either is
- masqueraded or hangs up (technically, both cases are hangups, but
- we need to handle the cases slightly differently). The solution
- for this is to employ a datastore, which has the nice benefit of
- allowing us to hook into channel masquerades and hangups and
- update our pointer as necessary. If this looks familiar, this
- same technique is employed in app_chanspy. app_chanspy is a bit
- more involved since it does a lot more operations on the channel
- that is being spied upon. app_mixmonitor does have an extra touch
- that app_chanspy doesn't have, though. Since there is a thread
- race between the channel's thread and the mixmonitor thread on a
- hangup, we em- ploy a condition-and-boolean combination to ensure
- that the channel thread finishes with our structure before the
- mixmonitor thread attempts to free it. No crashes! (closes issue
- #14374) Reported by: aragon Patches: 14374.patch uploaded by
- putnopvut (license 60) Tested by: aragon, putnopvut ........
- ................
-
-2009-02-05 16:23 +0000 [r173554] Jeff Peeler <jpeeler@digium.com>
-
- * build_tools/menuselect-deps.in: fix WORKING_FORK detection
-
-2009-02-05 00:11 +0000 [r173548] Tilghman Lesher <tlesher@digium.com>
-
- * build_tools/menuselect-deps.in: regenerate with bootstrap.sh
-
-2009-02-04 23:44 +0000 [r173546-173547] Jeff Peeler <jpeeler@digium.com>
-
- * /: I messed up and accidentally reverted the trunk-merged prop
- before committing 173546. Added it manually.
-
- * main/features.c: Merged revisions 173500 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r173500 | jpeeler | 2009-02-04 15:17:53 -0600 (Wed, 04 Feb 2009)
- | 23 lines Merged revisions 173211 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r173211 | jpeeler | 2009-02-03 15:57:01 -0600 (Tue, 03 Feb 2009)
- | 17 lines Parking attempts made to one end of a bridge no longer
- will hang up due to a parking failure. Parking attempts made
- using either one-touch, or doing either a blind or assisted
- transfer to the parking extension now keep up the bridge instead
- of hanging up the attempted parked party. Normal causes for the
- parking attempt to fail includes the specific specified extension
- (via PARKINGEXTEN) not being available or if all the parking
- spaces are currently in use. To avoid having to reverse a
- masquerade park_space_reserve was made to provide foresight if a
- parking attempt will succeed and if so reserve the parking space.
- (closes issue #13494) Reported by: mdu113 Reviewed by Russell:
- http://reviewboard.digium.com/r/133/ ........ ................
-
-2009-02-04 22:23 +0000 [r173534] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 173507 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r173507 |
- mmichelson | 2009-02-04 16:16:19 -0600 (Wed, 04 Feb 2009) | 7
- lines Fix some areas where the incorrect interface was passed to
- ast_device_state I swear it feels like I already did this once...
- (closes issue #14359) Reported by: francesco_r ........
-
-2009-02-04 18:55 +0000 [r173460] Tilghman Lesher <tlesher@digium.com>
-
- * main/tcptls.c, /: Merged revisions 173458 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r173458 |
- tilghman | 2009-02-04 12:48:06 -0600 (Wed, 04 Feb 2009) | 9 lines
- When using a socket as a FILE *, the stdio functions will
- sometimes try to do an fseek() on the stream, which is an invalid
- operation for a socket. Turning off buffering explicitly lets the
- stdio functions know they cannot do this, thus avoiding a
- potential error. (closes issue #14400) Reported by: fnordian
- Patches: tcptls.patch uploaded by fnordian (license 110) ........
-
-2009-02-04 17:46 +0000 [r173355-173398] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_chanspy.c, /: Merged revisions 173397 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r173397 | mmichelson | 2009-02-04 11:45:14 -0600 (Wed, 04 Feb
- 2009) | 11 lines Merged revisions 173396 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r173396 | mmichelson | 2009-02-04 11:44:48 -0600 (Wed, 04 Feb
- 2009) | 3 lines Revert my previous change because it was stupid
- ........ ................
-
- * apps/app_chanspy.c, /: Merged revisions 173393 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r173393 | mmichelson | 2009-02-04 11:41:02 -0600 (Wed, 04 Feb
- 2009) | 11 lines Merged revisions 173392 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r173392 | mmichelson | 2009-02-04 11:40:29 -0600 (Wed, 04 Feb
- 2009) | 3 lines Add a missing unlock. Extremely unlikely to ever
- matter, but it's needed. ........ ................
-
- * /, main/file.c: Merged revisions 173354 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r173354 |
- mmichelson | 2009-02-04 09:30:12 -0600 (Wed, 04 Feb 2009) | 30
- lines Fix a problem where file playback would cause fds to remain
- open forever The problem came from the fact that a frame read
- from a format interpreter was not freed. Adding a call to
- ast_frfree fixed this. The explanation for why this caused the
- problem is a bit complex, but here goes: There was a problem in
- all versions of Asterisk where the embedded frame of a filestream
- structure was referenced after the filestream was freed. This was
- fixed by adding reference counting to the filestream structure.
- The refcount would increase every time that a filestream's frame
- pointer was pointing to an actual frame of data. When the frame
- was freed, the refcount would decrease. Once the refcount reached
- 0, the filestream was freed, and as part of the operation, the
- open files were closed as well. Thus it becomes more clear why a
- missing ast_frfree would cause a reference leak and cause the
- files to not be closed. You may ask then if there was a frame
- leak before this patch. The answer to that is actually no! The
- filestream code was "smart" enough to know that since the frame
- we received came from a format interpreter, the frame had no
- malloced data and thus didn't need to be freed. Now, however,
- there is cleanup that needs to be done when we finish with the
- frame, so we do need to call ast_frfree on the frame to be sure
- that the refcount for the filestream is decremented
- appropriately. (closes issue #14384) Reported by: fiddur Patches:
- 14384.patch uploaded by putnopvut (license 60) Tested by: fiddur,
- putnopvut ........
-
-2009-02-04 00:45 +0000 [r173312] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c, /: Merged revisions 173311 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r173311 |
- tilghman | 2009-02-03 18:43:52 -0600 (Tue, 03 Feb 2009) | 10
- lines Ensure that commas placed in the middle of extension
- character classes do not interfere with correct parsing of the
- extension. Also, if an unterminated character class DOES make its
- way into the pbx core (through some other method), ensure that it
- does not crash Asterisk. (closes issue #14362) Reported by:
- Nick_Lewis Patches: 20090129__bug14362.diff.txt uploaded by
- Corydon76 (license 14) Tested by: Corydon76 ........
-
-2009-02-03 23:41 +0000 [r173250] David Vossel <dvossel@digium.com>
-
- * channels/chan_iax2.c: Fixes issue with IAX2 transfer not handing
- of calls. Fixes issue with IAX2 transfers not taking place. As it
- was, a call that was being transfered would never be handed off
- correctly to the call ends because of how call numbers were
- stored in a hash table. The hash table, "iax_peercallno_pvt",
- storing all the current call numbers did not take into account
- the complications associated with transferring a call, so a
- separate hash table was required. This second hash table
- "iax_transfercallno_pvt" handles calls being transfered, once the
- call transfer is complete the call is removed from the transfer
- hash table and added to the peer hash table resuming normal
- operations. Addition functions were created to handle storing,
- removing, and comparing items in the iax_transfercallno_pvt
- table. (issue #13468) Review:
- http://reviewboard.digium.com/r/140/
-
-2009-02-03 00:26 +0000 [r173111] Tilghman Lesher <tlesher@digium.com>
-
- * configs/extensions.conf.sample, /: Merged revisions 173104 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r173104 | tilghman | 2009-02-02 18:24:52 -0600
- (Mon, 02 Feb 2009) | 12 lines Merged revisions 173070 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r173070 | tilghman | 2009-02-02 18:15:59 -0600 (Mon, 02 Feb 2009)
- | 5 lines Add warning to standard config, that globals may be
- overridden by other dialplan configuration files. (closes issue
- #14388) Reported by: macli ........ ................
-
-2009-02-02 23:59 +0000 [r173068] Terry Wilson <twilson@digium.com>
-
- * /, main/features.c: Merged revisions 173067 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r173067 | twilson | 2009-02-02 17:57:25 -0600 (Mon, 02 Feb 2009)
- | 9 lines Merged revisions 173066 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r173066 | twilson | 2009-02-02 17:48:06 -0600 (Mon, 02 Feb 2009)
- | 2 lines Fix a feature inheritance bug I added after code review
- ........ ................
-
-2009-02-02 18:15 +0000 [r172896] Leif Madsen <lmadsen@digium.com>
-
- * /, configs/res_ldap.conf.sample: Merged revisions 172894 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r172894 | lmadsen | 2009-02-02 13:13:40 -0500 (Mon, 02
- Feb 2009) | 7 lines Update the res_ldap.conf file with a better
- working example. (closes issue #13861) Reported by: scramatte
- Patches: __20080110-res_ldap.conf-2.patch uploaded by blitzrage
- (license 10) Tested by: jcovert ........
-
-2009-02-01 02:45 +0000 [r172707-172742] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 172741 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r172741 | tilghman | 2009-01-31 20:44:23 -0600 (Sat, 31 Jan 2009)
- | 4 lines Blank argument crashes Asterisk (closes issue #14377)
- Reported by: amorsen ........
-
- * /, funcs/func_strings.c: Merged revisions 172706 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r172706 | tilghman | 2009-01-31 10:40:59 -0600 (Sat, 31 Jan 2009)
- | 7 lines Don't increment the loop, now that incrementing is
- taken care of by the decoder function. (closes issue #14363)
- Reported by: andrew53 Patches: func_strings_filter.patch uploaded
- by andrew53 (license 519) ........
-
-2009-01-31 00:06 +0000 [r172635-172637] Terry Wilson <twilson@digium.com>
-
- * configs/features.conf.sample, /: Merged revisions 172581 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r172581 | twilson | 2009-01-30 15:50:03 -0600 (Fri, 30
- Jan 2009) | 2 lines Remove incorret line from sample config
- ........
-
- * configs/features.conf.sample, apps/app_dial.c,
- main/global_datastores.c, /, main/features.c,
- include/asterisk/global_datastores.h, CHANGES: Merged revisions
- 172580 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r172580 | twilson | 2009-01-30 15:29:12 -0600 (Fri, 30 Jan 2009)
- | 44 lines Merged revisions 172517 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009)
- | 37 lines Fix feature inheritance with builtin features When
- using builtin features like parking and transfers, the
- AST_FEATURE_* flags would not be set correctly for all instances
- when either performing a builtin attended transfer, or parking a
- call and getting the timeout callback. Also, there was no way on
- a per-call basis to specify what features someone should have on
- picking up a parked call (since that doesn't involve the Dial()
- command). There was a global option for setting whether or not
- all users who pickup a parked call should have
- AST_FEATURE_REDIRECT set, but nothing for DISCONNECT, AUTOMON, or
- PARKCALL. This patch: 1) adds the BRIDGE_FEATURES dialplan
- variable which can be set either in the dialplan or with setvar
- in channels that support it. This variable can be set to any
- combination of 't', 'k', 'w', and 'h' (case insensitive matching
- of the equivalent dial options), to set what features should be
- activated on this channel. The patch moves the setting of the
- features datastores into the bridging code instead of app_dial to
- help facilitate this. 2) adds global options parkedcallparking,
- parkedcallhangup, and parkedcallrecording to be similar to the
- parkedcalltransfers option for globally setting features. 3) has
- builtin_atxfer call builtin_parkcall if being transfered to the
- parking extension since tracking everything through multiple
- masquerades, etc. is difficult and error-prone 4) attempts to fix
- all cases of return calls from parking and completed builtin
- transfers not having the correct permissions (closes issue
- #14274) Reported by: aragon Patches:
- fix_feature_inheritence.diff.txt uploaded by otherwiseguy
- (license 396) Tested by: aragon, otherwiseguy Review
- http://reviewboard.digium.com/r/138/ ........ ................
-
-2009-01-30 22:23 +0000 [r172604] Mark Michelson <mmichelson@digium.com>
-
- * /, include/asterisk/channel.h: Merged revisions 172598 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r172598 | mmichelson | 2009-01-30 16:22:04 -0600 (Fri,
- 30 Jan 2009) | 3 lines Fix redefinition of flag in channel.h
- ........
-
-2009-01-29 23:47 +0000 [r172503] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c, apps/app_nbscat.c, /, autoconf/ast_func_fork.m4,
- apps/app_festival.c, build_tools/menuselect-deps.in, configure,
- apps/app_dahdiras.c, apps/app_mp3.c, res/res_agi.c,
- apps/app_externalivr.c, apps/app_ices.c, res/res_musiconhold.c:
- Merged revisions 172441 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r172441 | tilghman | 2009-01-29 17:15:40 -0600 (Thu, 29 Jan 2009)
- | 16 lines Merged revisions 172438 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r172438 | tilghman | 2009-01-29 16:54:29 -0600 (Thu, 29 Jan 2009)
- | 9 lines Lose the CAP_NET_ADMIN at every fork, instead of at
- startup. Otherwise, if Asterisk runs as a non-root user and the
- administrator does a 'restart now', Asterisk loses the ability to
- set QOS on packets. (closes issue #14004) Reported by: nemo
- Patches: 20090105__bug14004.diff.txt uploaded by Corydon76
- (license 14) Tested by: Corydon76 ........ ................
-
-2009-01-29 21:35 +0000 [r172434] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged
- revisions 172400 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r172400 |
- rmudgett | 2009-01-29 14:38:34 -0600 (Thu, 29 Jan 2009) | 12
- lines channels/chan_dahdi.c * Added doxygen comments to the major
- dahdi structures. * Fixed PRI and SS7 using an incorrect string
- value if the extension delimiter is not present in the Dial()
- function. * Fixed SS7 not checking if the dialed extension is at
- least as long as the stripmsd option. * Fixed PRI not handling
- unknown TON/NPI prefix letters correctly. * Fixed some
- uninitialized string variables on FXS ports.
- configs/chan_dahdi.conf.sample * Updated some documentation.
- ........
-
-2009-01-29 16:49 +0000 [r172316] Tilghman Lesher <tlesher@digium.com>
-
- * configs/func_odbc.conf.sample, /: Merged revisions 172315 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r172315 | tilghman | 2009-01-29 10:48:25 -0600 (Thu, 29
- Jan 2009) | 2 lines Better document mode=multirow, based upon a
- conversation with Jared. ........
-
-2009-01-29 13:51 +0000 [r172273] Leif Madsen <lmadsen@digium.com>
-
- * contrib/scripts/realtime_pgsql.sql, /: Merged revisions 172271
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r172271 | lmadsen | 2009-01-29 08:47:27 -0500 (Thu, 29
- Jan 2009) | 5 lines The realtime_pgsql.sql script is missing a
- couple of fields. closes issue #14339) Reported by: fiddur
- Patches: realtime_pgsql.sql.diff uploaded by fiddur (license 678)
- ........
-
-2009-01-29 09:56 +0000 [r172217] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 172173 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r172173 | oej | 2009-01-29 10:18:01 +0100 (Tor, 29 Jan 2009) | 24
- lines Merged revisions 172169 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r172169 | oej | 2009-01-29 09:48:18 +0100 (Tor, 29 Jan 2009) | 16
- lines Make sure that we always add the hangupcause headers. In
- some cases, the owner was disconnected before we checked for the
- cause. This patch implements a temporary storage in the pvt and
- use that instead. The code is based on ideas from code from
- Adomjan in issue #13385 (Add support for Reason: header) Thanks
- to Klaus Darillion for testing! (closes issue #14294) related to
- issue #13385 Reported by: klaus3000 and adomjan Patches:
- bug14294b.diff uploaded by oej (license 306) Based on
- 20080829_chan_sip.c-q850reason_header.patch uploaded by adomjan
- (license 487) Tested by: oej, klaus3000 ........ ................
-
-2009-01-28 20:41 +0000 [r172065] Steve Murphy <murf@digium.com>
-
- * apps/app_channelredirect.c, main/pbx.c, main/manager.c, /,
- main/features.c, include/asterisk/channel.h: Merged revisions
- 172063 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r172063 | murf | 2009-01-28 13:31:06 -0700 (Wed, 28 Jan 2009) |
- 52 lines Merged revisions 172030 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r172030 | murf | 2009-01-28 11:51:16 -0700 (Wed, 28 Jan 2009) |
- 46 lines This patch fixes h-exten running misbehavior in
- manager-redirected situations. What it does: 1. A new Flag value
- is defined in include/asterisk/channel.h,
- AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the
- bridge hangup exten code not to run the h-exten there (nor
- publish the bridge cdr there). It will done at the pbx-loop level
- instead. 2. In the manager Redirect code, I set this flag on the
- channel if the channel has a non-null pbx pointer. I did the same
- for the second (chan2) channel, which gets run if name2 is set...
- and the first succeeds. 3. I restored the ending of the cdr for
- the pbx loop h-exten running code. Don't know why it was removed
- in the first place. 4. The first attempt at the fix for this bug
- was to place code directly in the async_goto routine, which was
- called from a large number of places, and could affect a large
- number of cases, so I tested that fix against a fair number of
- transfer scenarios, both with and without the patch. In the
- process, I saw that putting the fix in async_goto seemed not to
- affect any of the blind or attended scenarios, but still, I was
- was highly concerned that some other scenarios I had not tested
- might be negatively impacted, so I refined the patch to its
- current scope, and jmls tested both. In the process, tho, I saw
- that blind xfers in one situation, when the one-touch blind-xfer
- feature is used by the peer, we got strange h-exten behavior. So,
- I inserted code to swap CDRs and to set the HANGUP_DONT field, to
- get uniform behavior. 5. I added code to the bridge to obey the
- HANGUP_DONT flag, skipping both publishing the bridge CDR, and
- running the h-exten; they will be done at the pbx-loop (higher)
- level instead. 6. I removed all the debug logs from the patch
- before committing. 7. I moved the AUTOLOOP set/reset in the
- h-exten code in res_features so it's only done if the h-exten is
- going to be run. A very minor performance improvement, but
- technically correct. (closes issue #14241) Reported by: jmls
- Patches: 14241_redirect_no_bridgeCDR_or_h_exten_via_transfer
- uploaded by murf (license 17) Tested by: murf, jmls ........
- ................
-
-2009-01-28 17:28 +0000 [r171965] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 171964 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r171964 | tilghman | 2009-01-28 11:27:40 -0600
- (Wed, 28 Jan 2009) | 9 lines Merged revisions 171963 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r171963 | tilghman | 2009-01-28 11:25:18 -0600 (Wed, 28
- Jan 2009) | 2 lines Clarify log message (suggested by manxpower
- on #asterisk-dev) ........ ................
-
-2009-01-28 13:18 +0000 [r171846] Olle Johansson <oej@edvina.net>
-
- * /, configs/sip.conf.sample: Merged revisions 171838 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r171838 | oej | 2009-01-28 14:11:44 +0100 (Ons,
- 28 Jan 2009) | 10 lines Merged revisions 171837 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r171837 | oej | 2009-01-28 14:07:27 +0100 (Ons, 28 Jan 2009) | 2
- lines Add a better explanation of the difference between the
- device namespace and the dialplan for newbies. ........
- ................
-
-2009-01-27 22:00 +0000 [r171619-171692] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 171691 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r171691 | mmichelson | 2009-01-27 15:58:39 -0600
- (Tue, 27 Jan 2009) | 47 lines Merged revisions 171689 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r171689 | mmichelson | 2009-01-27 15:55:08 -0600 (Tue, 27 Jan
- 2009) | 39 lines Fix devicestate problems for "always-on" agent
- channels A revision to chan_agent attempted to "inherit" the
- device state of the underlying channel in order to report the
- device state of an agent channel more accurately. The problem
- with the logic here is that it makes no sense to use this for
- always-on agents. If the agent is logged in, then to the
- underlying channel, the agent will always appear to be "in use,"
- no matter if the agent is on a call or not. The reason is that to
- the underlying channel, the channel is currently in use on a call
- to the AgentLogin application. The most common cause that I found
- for this issue to occur was for a SIP channel to be the
- underlying channel type for an Agent channel. If the SIP phone
- re-registers, then the registration will cause the device state
- core to query the device state of the SIP channel. Since the SIP
- channel is in use, the Agent channel would also inherit this
- status. Once the agent channel was set to "in use" there was no
- way that the device state could change on that channel unless the
- agent logged out. The solution for this problem is a bit
- different in 1.4 than it is in the other branches. In 1.4, there
- will be a one-line fix to make sure that only callback agents
- will inherit device state from their underlying channel type. For
- the other branches of Asterisk, since callback support has been
- removed, there is also no need for device state inheritance in
- chan_agent, so I will simply be removing it from the code. In
- addition, the 1.4 source is getting a new comment to help the
- next person who edits chan_agent.c. I'm adding a comment that a
- agent_pvt's loginchan field may be used to determine if the agent
- is a callback agent or not. (closes issue #14173) Reported by:
- nathan Patches: 14173.patch uploaded by putnopvut (license 60)
- Tested by: nathan, aramirez ........ ................
-
- * /, main/slinfactory.c: Merged revisions 171622 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r171622 | mmichelson | 2009-01-27 14:11:30 -0600 (Tue, 27 Jan
- 2009) | 26 lines Merged revisions 171621 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r171621 | mmichelson | 2009-01-27 14:06:01 -0600 (Tue, 27 Jan
- 2009) | 18 lines Prevent a crash from occurring when a jitter
- buffer interpolated frame is removed from a slinfactory
- slinfactory used the "samples" field of an ast_frame in order to
- determine the amount of data contained within the frame. In
- certain cases, such as jitter buffer interpolated frames, the
- frame would have a non-zero value for "samples" but have NULL
- "data" This caused a problem when a memcpy call in
- ast_slinfactory_read would attempt to access invalid memory. The
- solution in use here is to never feed frames into the slinfactory
- if they have NULL "data" (closes issue #13116) Reported by:
- aragon Patches: 13116.diff uploaded by putnopvut (license 60)
- ........ ................
-
- * /, apps/app_queue.c: Merged revisions 171618 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r171618 |
- mmichelson | 2009-01-27 13:30:54 -0600 (Tue, 27 Jan 2009) | 24
- lines Fix queue crashes that would occur after the calling
- channel was masqueraded. The data passed to the
- end_bridge_callback was assumed to be data which was still
- stack'd. The problem was that with some call features, attended
- transfers in particular, a new bridge thread is started once the
- feature completes, meaning that when the end_bridge_callback is
- called, the end_bridge_callback_data was invalid. To fix this
- problem, there are two measures taken 1. Instead of pointing to
- stacked data, we now used heap-allocated data for passing to the
- end_bridge_callback in app_queue 2. Since bridges can end
- multiple times on a single logical call, we wait until the final
- bridge is broken to actually set any queue variables. This is
- accomplished through reference-counting and the use of an
- end_bridge_callback_data_fixup function in app_queue.c (closes
- issue #14260) Reported by: ccesario Patches: 14260.patch uploaded
- by putnopvut (license 60) Tested by: ccesario ........
-
-2009-01-27 16:15 +0000 [r171594-171595] Matthew Fredrickson <creslin@digium.com>
-
- * main/ast_expr2.c, main/ast_expr2.h: Revert some changes that
- shouldn't have made it in
-
- * main/ast_expr2.c, channels/chan_dahdi.c, main/ast_expr2.h: Make
- sure we do not go into alarm on PTMP links with non persistent
- D-channels
-
-2009-01-27 15:13 +0000 [r171529] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 171528 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r171528 | oej | 2009-01-27 16:00:19 +0100 (Tis, 27 Jan 2009) | 23
- lines Solving the same issue, but a bit different in trunk...
- Merged revisions 171527 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r171527 | oej | 2009-01-27 15:33:20 +0100 (Tis, 27 Jan 2009) | 13
- lines Use the same branch tag in CANCEL as in INVITE Originally
- putnopvut implemented some changes in revision 142079 that
- according to the bug report seemed to have worked then, but
- somehow fails now. I guess code, as humans, get old and forget
- stuff. Anyway, this bug caused CANCEL not to work with picky
- systems. Thanks Fredrik for pointing out where the bug in the SIP
- messaging was. (closes issue #14346) Reported by: oej Patches:
- bug14346.diff uploaded by oej (license 306) Tested by: oej
- ........ ................
-
-2009-01-26 14:02 +0000 [r171327] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 171326 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r171326 | oej | 2009-01-26 14:44:40 +0100 (MÃ¥n, 26 Jan 2009) |
- 17 lines Merged revisions 171264 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r171264 | oej | 2009-01-26 13:51:53 +0100 (MÃ¥n, 26 Jan 2009) | 9
- lines Don't retransmit 401 on REGISTER requests when
- alwaysauthreject=yes (closes issue #14284) Reported by: klaus3000
- Patches: patch_chan_sip_unreliable_1.4.23_14284.txt uploaded by
- klaus3000 (license 65) Tested by: klaus3000 ........
- ................
-
-2009-01-26 00:03 +0000 [r171189] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_oss.c, /: Merged revisions 171188 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r171188 | tilghman | 2009-01-25 17:58:00 -0600 (Sun, 25 Jan 2009)
- | 13 lines Merged revisions 171187 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r171187 | tilghman | 2009-01-25 17:44:01 -0600 (Sun, 25 Jan 2009)
- | 6 lines Correctly track the hookstate (closes issue #13686)
- Reported by: itiliti Patches: 20081013__bug13686.diff.txt
- uploaded by Corydon76 (license 14) ........ ................
-
-2009-01-25 13:38 +0000 [r170981] Sean Bright <sean.bright@gmail.com>
-
- * /, apps/app_page.c: Merged revisions 170980 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r170980 | seanbright | 2009-01-25 08:35:48 -0500 (Sun, 25 Jan
- 2009) | 16 lines Merged revisions 170979 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r170979 | seanbright | 2009-01-25 08:33:20 -0500 (Sun, 25 Jan
- 2009) | 9 lines Resolve a logic error that was causing Page() to
- crash when more than one channel was specified. (closes issue
- #14308) Reported by: bluefox Patches: 20090124__bug14308.diff.txt
- uploaded by seanbright (license 71) Tested by: kc0bvu ........
- ................
-
-2009-01-25 02:50 +0000 [r170944] Russell Bryant <russell@digium.com>
-
- * include/asterisk/utils.h, /: Merged revisions 170943 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r170943 | russell | 2009-01-24 20:49:30 -0600 (Sat, 24 Jan 2009)
- | 6 lines Change ARRAY_LEN() to be more C++ safe. When the second
- part of this macro is written as 0[a] instead of a[0], it will
- force a failure if the macro is used on a C++ object that
- overloads the [] operator. ........
-
-2009-01-24 13:56 +0000 [r170838] Tilghman Lesher <tlesher@digium.com>
-
- * configs/res_odbc.conf.sample, /: Merged revisions 170837 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r170837 | tilghman | 2009-01-24 07:55:53 -0600
- (Sat, 24 Jan 2009) | 9 lines Merged revisions 170836 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r170836 | tilghman | 2009-01-24 07:55:02 -0600 (Sat, 24
- Jan 2009) | 2 lines Remove superfluous implementation note
- (closes issue #14319) ........ ................
-
-2009-01-23 23:52 +0000 [r170830] Richard Mudgett <rmudgett@digium.com>
-
- * /, doc/tex/Makefile: Merged revisions 170794 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r170794 |
- rmudgett | 2009-01-23 17:10:34 -0600 (Fri, 23 Jan 2009) | 1 line
- Fix asterisk.pdf generation if branch name has an underscore in
- it. ........
-
-2009-01-23 22:59 +0000 [r170791] Russell Bryant <russell@digium.com>
-
- * /, doc/tex/Makefile: Merged revisions 170790 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r170790 |
- russell | 2009-01-23 16:58:37 -0600 (Fri, 23 Jan 2009) | 2 lines
- Don't blow up if a branch name has an underscore in it ........
-
-2009-01-23 20:56 +0000 [r170685-170721] Mark Michelson <mmichelson@digium.com>
-
- * configs/res_odbc.conf.sample, /: Merged revisions 170720 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r170720 | mmichelson | 2009-01-23 14:56:07 -0600
- (Fri, 23 Jan 2009) | 16 lines Merged revisions 170719 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r170719 | mmichelson | 2009-01-23 14:55:26 -0600 (Fri, 23 Jan
- 2009) | 8 lines Add notes to the idlecheck explanation in
- res_odbc.conf.sample (closes issue #14319) Reported by: klaus3000
- Patches: patch_idlecheck_res_odbc.conf.sample.txt uploaded by
- klaus3000 (license 65) ........ ................
-
- * contrib/i18n.testsuite.conf, /: Merged revisions 170677 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r170677 | mmichelson | 2009-01-23 14:23:00 -0600
- (Fri, 23 Jan 2009) | 22 lines Merged revisions 170671 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r170671 | mmichelson | 2009-01-23 14:21:51 -0600 (Fri, 23 Jan
- 2009) | 14 lines Update contrib/i18n.testsuite.conf to not use
- deprecated syntax * Convert Wait,1 to Wait(1) * Convert
- SetLanguage to Set(CHANNEL(language)) * Use 'n' for all
- priorities beyond the first Also added test for Chinese numbers,
- too. (closes issue #14320) Reported by: dant Patches:
- i18n.testsuite.conf.issue14320.v2.diff uploaded by dant (license
- 670) ........ ................
-
-2009-01-23 20:19 +0000 [r170659] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /: Merged revisions 170652 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r170652 | file | 2009-01-23 16:18:05 -0400 (Fri, 23 Jan 2009) |
- 11 lines Merged revisions 170648 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r170648 | file | 2009-01-23 16:16:39 -0400 (Fri, 23 Jan 2009) | 4
- lines When a channel is answered make sure any indications
- currently playing stop. Usually the phone would do this but if
- the channel was already answered then they are being generated by
- Asterisk and we darn well need to stop them. (closes issue
- #14249) Reported by: RadicAlish ........ ................
-
-2009-01-23 Tilghman Lesher <tlesher@digium.com>
-
- * Released 1.6.0.5
-
- * channels/chan_iax2.c: Regression fixes for security fix AST-2009-001
-
-2009-01-06 Tilghman Lesher <tlesher@digium.com>
-
- * Released 1.6.0.3
-
- * channels/chan_iax2.c: Security fix AST-2009-001
-
-2008-12-03 Tilghman Lesher <tlesher@digium.com>
-
- * Released 1.6.0.3-rc1
-
-2008-12-03 14:13 +0000 [r160482] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 160481 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r160481 | tilghman | 2008-12-03 08:11:53 -0600 (Wed, 03 Dec 2008)
- | 14 lines Merged revisions 160480 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r160480 | tilghman | 2008-12-03 08:09:35 -0600 (Wed, 03 Dec 2008)
- | 7 lines 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. ........ ................
-
-2008-12-03 00:53 +0000 [r160427] Sean Bright <sean.bright@gmail.com>
-
- * Makefile: Fix some 'make menuselect' breakage introduced by
- recent merges.
-
-2008-12-02 23:22 +0000 [r160386-160393] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 156388 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r156388 | tilghman | 2008-11-12 15:34:51 -0600 (Wed, 12 Nov 2008)
- | 12 lines Merged revisions 156386 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r156386 | tilghman | 2008-11-12 15:18:57 -0600 (Wed, 12 Nov 2008)
- | 5 lines When using call limits under 1 second, infinite call
- lengths are allowed, instead. (closes issue #13851) Reported by:
- ruddy ........ ................
-
- * /, apps/app_meetme.c: Merged revisions 156290 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r156290 | jpeeler | 2008-11-12 13:11:15 -0600 (Wed, 12 Nov 2008)
- | 11 lines Merged revisions 156289 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r156289 | jpeeler | 2008-11-12 13:10:12 -0600 (Wed, 12 Nov 2008)
- | 3 lines For whatever reason, gcc only warned me about the
- possible use of an uninitialized variable when compiling 1.6.1.
- ........ ................
-
- * /, apps/app_meetme.c: Merged revisions 156228 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r156228 | jpeeler | 2008-11-12 12:32:46 -0600 (Wed, 12 Nov 2008)
- | 16 lines Merged revisions 156178 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r156178 | jpeeler | 2008-11-12 11:53:44 -0600 (Wed, 12 Nov 2008)
- | 8 lines (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/ ........ ................
-
- * main/astobj2.c, main/asterisk.c, apps/app_while.c,
- apps/app_dial.c, main/pbx.c, channels/chan_misdn.c,
- main/manager.c, /, apps/app_meetme.c, channels/chan_sip.c,
- channels/chan_skinny.c, include/asterisk/astobj2.h,
- channels/chan_agent.c, channels/chan_h323.c,
- channels/chan_iax2.c: Merged revisions
- 152969,153122,154264,154268,154366,155399,155863,156166,156295,156690,156756,158066,158082,158540,158602,159276
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r152969 | tilghman | 2008-10-30 15:35:46 -0500
- (Thu, 30 Oct 2008) | 10 lines Merged revisions 152958 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r152958 | tilghman | 2008-10-30 15:33:28 -0500 (Thu, 30 Oct 2008)
- | 3 lines Cannot join detached threads. See
- http://www.opengroup.org/onlinepubs/000095399/functions/pthread_join.html
- (Closes issue #13400) ........ ................ r153122 |
- tilghman | 2008-10-31 11:35:21 -0500 (Fri, 31 Oct 2008) | 10
- lines Merged revisions 153114 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r153114 | tilghman | 2008-10-31 11:30:32 -0500 (Fri, 31 Oct 2008)
- | 3 lines Turn off qualify on uncached realtime peers. (Closes
- issue #13383) ........ ................ r154264 | tilghman |
- 2008-11-04 12:59:48 -0600 (Tue, 04 Nov 2008) | 10 lines Recorded
- merge of revisions 154263 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r154263 | tilghman | 2008-11-04 12:58:05 -0600 (Tue, 04 Nov 2008)
- | 3 lines Make the monitor thread non-detached, so it can be
- joined (suggested by Russell on -dev list). ........
- ................ r154268 | rmudgett | 2008-11-04 13:07:26 -0600
- (Tue, 04 Nov 2008) | 11 lines Merged revisions 154266 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r154266 | rmudgett | 2008-11-04 13:01:08 -0600 (Tue, 04 Nov 2008)
- | 4 lines JIRA ABE-1703 mISDN sets the channel to the wrong state
- when it receives the indication AST_CONTROL_RINGING. ........
- ................ r154366 | tilghman | 2008-11-04 14:51:18 -0600
- (Tue, 04 Nov 2008) | 16 lines Merged revisions 154365 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r154365 | tilghman | 2008-11-04 14:49:33 -0600 (Tue, 04 Nov 2008)
- | 9 lines 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 ........
- ................ r155399 | tilghman | 2008-11-07 16:28:58 -0600
- (Fri, 07 Nov 2008) | 14 lines Merged revisions 155398 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r155398 | tilghman | 2008-11-07 16:27:32 -0600 (Fri, 07 Nov 2008)
- | 7 lines Clarify error message. (closes issue #13809) Reported
- by: denke Patches: 20081104__bug13809.diff.txt uploaded by
- Corydon76 (license 14) Tested by: denke ........ ................
- r155863 | mmichelson | 2008-11-10 15:14:44 -0600 (Mon, 10 Nov
- 2008) | 22 lines Merged revisions 155861 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r155861 | mmichelson | 2008-11-10 15:07:39 -0600 (Mon, 10 Nov
- 2008) | 14 lines 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 ........
- ................ r156166 | russell | 2008-11-12 11:38:20 -0600
- (Wed, 12 Nov 2008) | 15 lines Merged revisions 156164 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r156164 | russell | 2008-11-12 11:29:52 -0600 (Wed, 12 Nov 2008)
- | 7 lines 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.) ........
- ................ r156295 | tilghman | 2008-11-12 13:28:22 -0600
- (Wed, 12 Nov 2008) | 13 lines Merged revisions 156294 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r156294 | tilghman | 2008-11-12 13:26:45 -0600 (Wed, 12 Nov 2008)
- | 6 lines 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) ........
- ................ r156690 | tilghman | 2008-11-13 15:30:41 -0600
- (Thu, 13 Nov 2008) | 14 lines Merged revisions 156688 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r156688 | tilghman | 2008-11-13 15:24:00 -0600 (Thu, 13 Nov 2008)
- | 7 lines 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)
- ........ ................ r156756 | tilghman | 2008-11-13
- 18:43:13 -0600 (Thu, 13 Nov 2008) | 13 lines Merged revisions
- 156755 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r156755 | tilghman | 2008-11-13 18:41:37 -0600 (Thu, 13 Nov 2008)
- | 6 lines 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 ........ ................ r158066 | mmichelson |
- 2008-11-20 11:39:06 -0600 (Thu, 20 Nov 2008) | 20 lines Merged
- revisions 158053 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r158053 | mmichelson | 2008-11-20 11:33:06 -0600 (Thu, 20 Nov
- 2008) | 12 lines 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 ........
- ................ r158082 | mmichelson | 2008-11-20 11:54:31 -0600
- (Thu, 20 Nov 2008) | 24 lines Merged revisions 158071 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r158071 | mmichelson | 2008-11-20 11:48:42 -0600 (Thu, 20 Nov
- 2008) | 16 lines 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 ........ ................
- r158540 | russell | 2008-11-21 16:12:37 -0600 (Fri, 21 Nov 2008)
- | 10 lines Merged revisions 158539 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r158539 | russell | 2008-11-21 16:05:55 -0600 (Fri, 21 Nov 2008)
- | 2 lines When compiling with DEBUG_THREADS, report the real
- file/func/line for ao2_lock/ao2_unlock ........ ................
- r158602 | tilghman | 2008-11-21 17:14:11 -0600 (Fri, 21 Nov 2008)
- | 12 lines Merged revisions 158600 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r158600 | tilghman | 2008-11-21 17:07:46 -0600 (Fri, 21 Nov 2008)
- | 5 lines 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) ........ ................ r159276 | tilghman | 2008-11-25
- 15:57:59 -0600 (Tue, 25 Nov 2008) | 14 lines Merged revisions
- 159269 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r159269 | tilghman | 2008-11-25 15:56:48 -0600 (Tue, 25 Nov 2008)
- | 7 lines 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 ........
- ................
-
- * configs/features.conf.sample, apps/app_voicemail.c,
- apps/app_dial.c, channels/chan_dahdi.c, channels/chan_local.c, /,
- channels/chan_sip.c, apps/app_queue.c: Merged revisions
- 152216,152287,152369,152467,152569,152605 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r152216 | tilghman | 2008-10-27 16:34:04 -0500 (Mon, 27 Oct 2008)
- | 13 lines Merged revisions 152215 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r152215 | tilghman | 2008-10-27 16:32:00 -0500 (Mon, 27 Oct 2008)
- | 6 lines 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) ........ ................ r152287 | jpeeler |
- 2008-10-27 18:31:39 -0500 (Mon, 27 Oct 2008) | 10 lines Merged
- revisions 152286 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r152286 | jpeeler | 2008-10-27 18:28:49 -0500 (Mon, 27 Oct 2008)
- | 2 lines Buffer policy setting for half is not needed. ........
- ................ r152369 | tilghman | 2008-10-28 12:07:39 -0500
- (Tue, 28 Oct 2008) | 15 lines Merged revisions 152368 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r152368 | tilghman | 2008-10-28 12:04:56 -0500 (Tue, 28 Oct 2008)
- | 8 lines 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 ........ ................ r152467 |
- tilghman | 2008-10-28 17:33:40 -0500 (Tue, 28 Oct 2008) | 10
- lines Merged revisions 152463 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r152463 | tilghman | 2008-10-28 17:32:34 -0500 (Tue, 28 Oct 2008)
- | 3 lines Quoting in the wrong direction (Fixes AST-107) ........
- ................ r152569 | russell | 2008-10-29 00:34:26 -0500
- (Wed, 29 Oct 2008) | 15 lines Merged revisions 152539 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r152539 | russell | 2008-10-29 00:23:51 -0500 (Wed, 29 Oct 2008)
- | 7 lines Fix an incorrect usage of sizeof() (closes issue
- #13795) Reported by: andrew53 Patches: chan_sip_sizeof.patch
- uploaded by andrew53 (license 519) ........ ................
- r152605 | murf | 2008-10-29 00:47:13 -0500 (Wed, 29 Oct 2008) |
- 22 lines Merged revisions 152538 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r152538 | murf | 2008-10-28 23:19:04 -0600 (Tue, 28 Oct 2008) |
- 14 lines 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_speech_utils.c, apps/app_voicemail.c, Makefile,
- channels/chan_dahdi.c, /, channels/chan_sip.c,
- include/asterisk/audiohook.h, apps/app_waitforsilence.c,
- main/features.c, main/audiohook.c, apps/app_queue.c: Merged
- revisions
- 147518,147689,148000,148112,148268,148917,148988,149062,149131,149201,149205,149208
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r147518 | file | 2008-10-08 09:53:51 -0500 (Wed,
- 08 Oct 2008) | 9 lines Merged revisions 147517 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r147517 | file | 2008-10-08 11:51:42 -0300 (Wed, 08 Oct 2008) | 2
- lines If we receive DTMF make sure that the state of the speech
- structure goes back to being not ready. (issue #LUMENVOX-8)
- ........ ................ r147689 | kpfleming | 2008-10-08
- 17:26:55 -0500 (Wed, 08 Oct 2008) | 9 lines Merged revisions
- 147681 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r147681 | kpfleming | 2008-10-08 17:22:09 -0500 (Wed, 08 Oct
- 2008) | 3 lines 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)
- ........ ................ r148000 | tilghman | 2008-10-09
- 14:39:34 -0500 (Thu, 09 Oct 2008) | 11 lines Merged revisions
- 147997 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r147997 | tilghman | 2008-10-09 14:38:33 -0500 (Thu, 09 Oct 2008)
- | 4 lines When blank, callerid name and number should display
- "unknown caller" in voicemail emails. (Closes issue #13643)
- ........ ................ r148112 | mmichelson | 2008-10-09
- 18:15:33 -0500 (Thu, 09 Oct 2008) | 26 lines Merged revisions
- 146026 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r146026 | murf | 2008-10-03 12:12:54 -0500 (Fri, 03 Oct 2008) |
- 18 lines (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! ........ ................ r148268 | tilghman |
- 2008-10-10 11:31:31 -0500 (Fri, 10 Oct 2008) | 14 lines Merged
- revisions 148257 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r148257 | tilghman | 2008-10-10 11:25:31 -0500 (Fri, 10 Oct 2008)
- | 7 lines 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 ........ ................
- r148917 | tilghman | 2008-10-14 12:46:48 -0500 (Tue, 14 Oct 2008)
- | 11 lines Merged revisions 148916 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r148916 | tilghman | 2008-10-14 12:41:08 -0500 (Tue, 14 Oct 2008)
- | 4 lines Ensure that mail headers are 7-bit clean, even when
- UTF-8 characters are used in headers like 'Subject' and 'To'.
- Closes AST-107. ........ ................ r148988 | tilghman |
- 2008-10-14 14:03:44 -0500 (Tue, 14 Oct 2008) | 9 lines Merged
- revisions 148987 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r148987 | tilghman | 2008-10-14 14:03:08 -0500 (Tue, 14 Oct 2008)
- | 2 lines Some compilers warn, some don't. Fixing. ........
- ................ r149062 | tilghman | 2008-10-14 15:16:48 -0500
- (Tue, 14 Oct 2008) | 13 lines Merged revisions 149061 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r149061 | tilghman | 2008-10-14 15:09:06 -0500 (Tue, 14 Oct 2008)
- | 6 lines 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 ........ ................
- r149131 | mmichelson | 2008-10-14 16:08:48 -0500 (Tue, 14 Oct
- 2008) | 15 lines Merged revisions 149130 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r149130 | mmichelson | 2008-10-14 15:49:02 -0500 (Tue, 14 Oct
- 2008) | 7 lines Don't allow reserved characters to be used in
- register lines in sip.conf. (closes issue #13570) Reported by:
- putnopvut ........ ................ r149201 | mmichelson |
- 2008-10-14 17:41:13 -0500 (Tue, 14 Oct 2008) | 20 lines Merged
- revisions 149200 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r149200 | mmichelson | 2008-10-14 17:40:42 -0500 (Tue, 14 Oct
- 2008) | 12 lines 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) ........ ................ r149205 |
- mmichelson | 2008-10-14 18:04:44 -0500 (Tue, 14 Oct 2008) | 20
- lines Merged revisions 149204 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r149204 | mmichelson | 2008-10-14 18:00:01 -0500 (Tue, 14 Oct
- 2008) | 12 lines 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
- ........ ................ r149208 | mmichelson | 2008-10-14
- 18:15:04 -0500 (Tue, 14 Oct 2008) | 17 lines Merged revisions
- 149207 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r149207 | mmichelson | 2008-10-14 18:10:26 -0500 (Tue, 14 Oct
- 2008) | 9 lines 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) ........ ................
-
- * channels/misdn/isdn_lib.c, Makefile, channels/chan_dahdi.c,
- channels/chan_misdn.c, main/manager.c, /: Merged revisions
- 115313,121770,123272,139624,140205,144257 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r115313 | tilghman | 2008-05-05 15:22:08 -0500 (Mon, 05 May 2008)
- | 10 lines Merged revisions 115312 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r115312 | tilghman | 2008-05-05 15:17:55 -0500 (Mon, 05 May 2008)
- | 2 lines Reverse order, such that user configs override default
- selections ........ ................ r121770 | crichter |
- 2008-06-11 06:52:18 -0500 (Wed, 11 Jun 2008) | 9 lines Merged
- revisions 121751 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r121751 | crichter | 2008-06-11 11:28:04 +0200 (Mi, 11 Jun 2008)
- | 1 line fixed issue with previous commit, the find_free_channel
- test for channels which where inuse was broken. ........
- ................ r123272 | russell | 2008-06-17 10:52:13 -0500
- (Tue, 17 Jun 2008) | 12 lines Merged revisions 123271 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r123271 | russell | 2008-06-17 10:48:31 -0500 (Tue, 17 Jun 2008)
- | 4 lines 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. ........ ................ r139624 | jpeeler |
- 2008-08-22 16:57:32 -0500 (Fri, 22 Aug 2008) | 13 lines Merged
- revisions 139621 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r139621 | jpeeler | 2008-08-22 16:36:13 -0500 (Fri, 22 Aug 2008)
- | 5 lines (closes issue #13359) Reported by: Laureano Patches:
- originate_channel_check.patch uploaded by Laureano (license 265)
- ........ ................ r140205 | jpeeler | 2008-08-26 13:48:55
- -0500 (Tue, 26 Aug 2008) | 17 lines Merged revisions 140056 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r140056 | jpeeler | 2008-08-26 10:57:02 -0500 (Tue, 26 Aug 2008)
- | 9 lines (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. ........ ................ r144257 |
- crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008) | 9 lines
- Merged revisions 144238 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r144238 | crichter | 2008-09-24 10:20:52 +0200 (Mi, 24 Sep 2008)
- | 1 line improved helptext of misdn_set_opt. ........
- ................
-
-2008-12-02 18:05 +0000 [r160326-160337] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 160333 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r160333 | jpeeler | 2008-12-02 12:04:51 -0600 (Tue, 02 Dec 2008)
- | 1 line remove duplicate comment that I accidentally merged
- ........
-
- * channels/chan_dahdi.c, /: Merged revisions 160319 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r160319 | jpeeler | 2008-12-02 12:00:24 -0600 (Tue, 02 Dec 2008)
- | 7 lines (closes issue #13786) Reported by: tzafrir Readding
- DAHDI_CHECK_HOOKSTATE define that was removed in r134260 which
- fixes not being able to make outgoing calls on some FXO adapters:
- http://lists.digium.com/pipermail/asterisk-users/2008-November/thread.html#221553
- ........
-
-2008-12-02 18:01 +0000 [r160228-160322] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 160308 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r160308 | tilghman | 2008-12-02 11:56:24 -0600 (Tue, 02 Dec 2008)
- | 17 lines Merged revisions 160297 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r160297 | tilghman | 2008-12-02 11:42:09 -0600 (Tue, 02 Dec 2008)
- | 10 lines 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) ........ ................
-
- * include/asterisk/stringfields.h, apps/app_voicemail.c,
- main/cli.c, main/pbx.c, main/frame.c, /,
- channels/chan_features.c: Merged revisions 160208 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r160208 | tilghman | 2008-12-01 18:37:21 -0600
- (Mon, 01 Dec 2008) | 10 lines Merged revisions 160207 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r160207 | tilghman | 2008-12-01 18:25:16 -0600 (Mon, 01 Dec 2008)
- | 3 lines Ensure that Asterisk builds with --enable-dev-mode,
- even on the latest gcc and glibc. ........ ................
-
-2008-12-01 23:41 +0000 [r160173] Sean Bright <sean.bright@gmail.com>
-
- * channels/chan_phone.c, main/manager.c, /, utils/smsq.c: Merged
- revisions 160170-160172 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r160170 | seanbright | 2008-12-01 18:08:24 -0500 (Mon, 01 Dec
- 2008) | 1 line Pay attention to the return value of system(),
- even if we basically ignore it. ................ r160171 |
- seanbright | 2008-12-01 18:18:48 -0500 (Mon, 01 Dec 2008) | 1
- line Silence a build warning. (chan_phone.c:810: warning: value
- computed is not used) ................ r160172 | seanbright |
- 2008-12-01 18:37:49 -0500 (Mon, 01 Dec 2008) | 10 lines Merged
- revisions 159976 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r159976 | mvanbaak | 2008-12-01 11:08:36 -0500 (Mon, 01 Dec 2008)
- | 3 lines 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 Tilghman Lesher <tlesher@digium.com>
-
- * Released 1.6.0.2
-
-2008-12-01 21:45 +0000 [r160100] Tilghman Lesher <tlesher@digium.com>
-
- * /, configure, configure.ac: Merged revisions 160097 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r160097 | tilghman | 2008-12-01 15:23:37 -0600 (Mon, 01 Dec 2008)
- | 2 lines Use AST_EXT_LIB_SETUP before using AST_EXT_LIB_CHECK or
- bad things happen. ........
-
-2008-12-01 21:07 +0000 [r160096] Sean Bright <sean.bright@gmail.com>
-
- * include/asterisk.h, /: Merged revisions 154919 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r154919 |
- seanbright | 2008-11-05 17:01:22 -0500 (Wed, 05 Nov 2008) | 2
- lines Fix a problem found while building res_snmp. ........
-
-2008-12-01 17:39 +0000 [r160005] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 160004 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r160004 | russell | 2008-12-01 11:34:31 -0600
- (Mon, 01 Dec 2008) | 14 lines Merged revisions 160003 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r160003 | russell | 2008-12-01 11:27:30 -0600 (Mon, 01 Dec 2008)
- | 6 lines 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 16:04 +0000 [r159974] Michiel van Baak <michiel@vanbaak.info>
-
- * main/manager.c, /: Merged revisions 159898 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r159898 | mvanbaak | 2008-12-01 15:09:59 +0100 (Mon, 01 Dec 2008)
- | 11 lines Merged revisions 159897 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r159897 | mvanbaak | 2008-12-01 15:05:41 +0100 (Mon, 01 Dec 2008)
- | 4 lines 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. ........ ................
-
-2008-12-01 14:56 +0000 [r159915] Russell Bryant <russell@digium.com>
-
- * .cleancount, /: Merged revisions 159911 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r159911 | russell | 2008-12-01 08:56:10 -0600 (Mon, 01 Dec 2008)
- | 10 lines Merged revisions 159900 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r159900 | russell | 2008-12-01 08:52:56 -0600 (Mon, 01 Dec 2008)
- | 2 lines Force a "make clean" to avoid a bizarre build issue ...
- ........ ................
-
-2008-11-29 18:37 +0000 [r159855] Kevin P. Fleming <kpfleming@digium.com>
-
- * utils/conf2ael.c, cdr/cdr_tds.c, 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, main/event.c,
- include/asterisk/strings.h, utils/extconf.c, makeopts.in,
- include/asterisk/stringfields.h, utils/check_expr.c,
- channels/chan_vpb.cc, /, main/utils.c, res/res_config_sqlite.c,
- utils/frame.c, channels/misdn_config.c, include/asterisk/astmm.h,
- include/asterisk/compat.h, configure, channels/misdn/ie.c,
- include/asterisk/module.h, main/features.c, main/dns.c,
- funcs/Makefile, include/asterisk/devicestate.h,
- include/asterisk/utils.h, channels/chan_sip.c, main/Makefile,
- include/asterisk/dundi.h, include/asterisk/enum.h, configure.ac,
- channels/chan_agent.c, utils/astman.c, include/asterisk/cli.h,
- include/asterisk/channel.h, include/jitterbuf.h,
- include/asterisk/manager.h: Merged revisions 159818 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r159818 | kpfleming | 2008-11-29 11:57:39 -0600 (Sat, 29 Nov
- 2008) | 18 lines incorporates r159808 from branches/1.4:
- ------------------------------------------------------------------------
- r159808 | kpfleming | 2008-11-29 10:58:29 -0600 (Sat, 29 Nov
- 2008) | 7 lines 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
- ------------------------------------------------------------------------
- in addition: move some format attributes from main/utils.c to the
- header files they belong in, and fix up references to the
- relevant functions based on new compiler warnings ........
-
-2008-11-26 19:58 +0000 [r159558] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 159554 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r159554 |
- mmichelson | 2008-11-26 13:57:11 -0600 (Wed, 26 Nov 2008) | 19
- lines Add some necessary hangup commands in the case that
- forwarding a call fails 1) Hang up the original destination if
- the local channel cannot be requested. 2) Hang up the local
- channel (in addition to the original destination) if ast_call
- fails when calling the newly created local channel. This prevents
- channels from sticking around forever in the case of a botched
- call forward (e.g. to an extension which does not exist). (closes
- issue #13764) Reported by: davidw Patches: 13764_v2.patch
- uploaded by putnopvut (license 60) Tested by: putnopvut, davidw
- ........
-
-2008-11-26 19:18 +0000 [r159536] Kevin P. Fleming <kpfleming@digium.com>
-
- * agi/Makefile, utils/Makefile, /, Makefile.moddir_rules,
- Makefile.rules: Merged revisions 159534 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r159534 | kpfleming | 2008-11-26 13:08:56 -0600 (Wed, 26 Nov
- 2008) | 11 lines Merged revisions 159476 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r159476 | kpfleming | 2008-11-26 12:36:24 -0600 (Wed, 26 Nov
- 2008) | 7 lines simplify (and slightly bug-fix) the recent
- developer-oriented COMPILE_DOUBLE mode ensure that 'make clean'
- removes dependency files for .i files that are created in
- COMPILE_DOUBLE mode ........ ................
-
-2008-11-26 18:40 +0000 [r159478] Tilghman Lesher <tlesher@digium.com>
-
- * main/udptl.c, /: Merged revisions 159475 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r159475 |
- tilghman | 2008-11-26 12:33:04 -0600 (Wed, 26 Nov 2008) | 7 lines
- If the config file does not exist, then the first use crashes
- Asterisk. (closes issue #13848) Reported by: klaus3000 Patches:
- udptl.c.patch uploaded by eliel (license 64) Tested by: blitzrage
- ........
-
-2008-11-26 15:01 +0000 [r159439] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_agent.c: Merged revisions 159437 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r159437 |
- mmichelson | 2008-11-26 08:58:17 -0600 (Wed, 26 Nov 2008) | 10
- lines Don't allow for configuration options to overwrite options
- set via channel variables on a reload. (closes issue #13921)
- Reported by: davidw Patches: 13921.patch uploaded by putnopvut
- (license 60) Tested by: davidw ........
-
-2008-11-25 23:09 +0000 [r159374] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, /, channels/chan_iax2.c: Merged revisions 159360 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r159360 | murf | 2008-11-25 16:03:01 -0700 (Tue,
- 25 Nov 2008) | 23 lines Merged revisions 159316 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r159316 | murf | 2008-11-25 15:41:10 -0700 (Tue, 25 Nov 2008) |
- 15 lines (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 22:28 +0000 [r159314] Mark Michelson <mmichelson@digium.com>
-
- * main/channel.c: I don't care what anyone says, this change is
- going into 1.6.0. Otherwise, the simple act of logging an agent
- in spams the CLI with warning messages about failed reads of the
- alertpipe.
-
-2008-11-25 21:43 +0000 [r159248] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 159247 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r159247 | tilghman | 2008-11-25 15:42:42 -0600
- (Tue, 25 Nov 2008) | 21 lines Merged revisions 159246 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r159246 | tilghman | 2008-11-25 15:40:28 -0600
- (Tue, 25 Nov 2008) | 14 lines 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 16:21 +0000 [r159024-159094] Terry Wilson <twilson@digium.com>
-
- * /, apps/app_festival.c: Merged revisions 159093 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r159093 |
- twilson | 2008-11-25 10:18:53 -0600 (Tue, 25 Nov 2008) | 2 lines
- Add missing variable declaration for PPC code ........
-
- * channels/chan_usbradio.c, /: Merged revisions 158992 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r158992 | twilson | 2008-11-24 21:49:30 -0600 (Mon, 24 Nov 2008)
- | 2 lines Make chan_usbradio compile under dev mode ........
-
-2008-11-21 22:40 +0000 [r158545] Steve Murphy <murf@digium.com>
-
- * /, main/features.c: Merged revisions 158484 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r158484 | murf | 2008-11-21 14:47:16 -0700 (Fri, 21 Nov 2008) |
- 19 lines Merged revisions 158483 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r158483 | murf | 2008-11-21 14:19:47 -0700 (Fri, 21 Nov 2008) |
- 11 lines (closes issue #13871) Reported by: mdu113 This one is
- totally my fault. The code doesn't even create a bridge CDR 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 22:13 +0000 [r158542] Russell Bryant <russell@digium.com>
-
- * main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions
- 158540 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r158540 | russell | 2008-11-21 16:12:37 -0600 (Fri, 21 Nov 2008)
- | 10 lines Merged revisions 158539 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r158539 | russell | 2008-11-21 16:05:55 -0600 (Fri, 21 Nov 2008)
- | 2 lines When compiling with DEBUG_THREADS, report the real
- file/func/line for ao2_lock/ao2_unlock ........ ................
-
-2008-11-21 20:44 +0000 [r158451] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, UPGRADE-1.2.txt, UPGRADE-1.4.txt, UPGRADE.txt,
- UPGRADE-1.6.txt, CHANGES: Merged revisions 158449 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r158449 | kpfleming | 2008-11-21 14:42:37 -0600 (Fri, 21 Nov
- 2008) | 3 lines as suggested by jtodd, document the purposes of
- the CHANGES and UPGRADE files ........
-
-2008-11-21 17:12 +0000 [r158376] Terry Wilson <twilson@digium.com>
-
- * cdr/cdr_csv.c, /: Merged revisions 158374 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r158374 |
- twilson | 2008-11-21 11:08:16 -0600 (Fri, 21 Nov 2008) | 8 lines
- Reloading the config and having no changes still initialized some
- settings to 0. Initialize settings after doing all of the cfg
- checks. (closes issue #13942) Reported by: davidw Patches:
- cdr_diff.txt uploaded by otherwiseguy (license 396) Tested by:
- davidw ........
-
-2008-11-21 01:23 +0000 [r158231-158267] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 158265-158266 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r158265 | mmichelson | 2008-11-20 19:14:20 -0600 (Thu,
- 20 Nov 2008) | 4 lines Use some magic constants to get the right
- size for this sscanf statement. Thanks Richard! ........ r158266
- | mmichelson | 2008-11-20 19:22:18 -0600 (Thu, 20 Nov 2008) | 3
- lines Use a more expressive constant for a 64-bit scanned int
- ........
-
- * /, channels/chan_sip.c: Merged revisions 158262 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r158262 |
- mmichelson | 2008-11-20 18:59:23 -0600 (Thu, 20 Nov 2008) | 6
- lines Fix the build for 32-bit systems. %lu is only 32-bits on
- 32-bit systems, so we need to use %llu instead. Of course %llu is
- 128-bits on 64-bit systems, so we have to cast to unsigned long
- long. No harm, but it's sure annoying. ........
-
- * /, channels/chan_sip.c: Merged revisions 158230 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r158230 |
- mmichelson | 2008-11-20 17:12:50 -0600 (Thu, 20 Nov 2008) | 20
- lines Change the remote user agent session version variable from
- an int to a uint64_t. This prevents potential comparison problems
- from happening if the version string exceeds INT_MAX. This was an
- apparent problem for one user who could not properly place a call
- on hold since the version in the SDP of the re-INVITE to place
- the call on hold greatly exceeded INT_MAX. This also aligns with
- RFC 2327 better since it recommends using an NTP timestamp for
- the version (which is a 64-bit number). (closes issue #13531)
- Reported by: sgofferj Patches: 13531.patch uploaded by putnopvut
- (license 60) Tested by: sgofferj ........
-
-2008-11-20 19:42 +0000 [r158190] Sean Bright <sean.bright@gmail.com>
-
- * res/ael/pval.c, /: Merged revisions 158188 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r158188 |
- seanbright | 2008-11-20 14:41:23 -0500 (Thu, 20 Nov 2008) | 10
- lines Fix one case where the application argument was not
- converted from a pipe to a comma. This was causing problems with
- switch statements with empty expressions. (closes issue #13901)
- Reported by: smurfix Patches: 20081118_bug13901.diff uploaded by
- seanbright (license 71) Tested by: seanbright Reviewed by: murf
- ........
-
-2008-11-20 00:12 +0000 [r157738-157976] 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, res/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, Makefile.rules, res/snmp,
- pbx/Makefile, res/Makefile: Merged revisions 157974 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r157974 | kpfleming | 2008-11-19 18:08:12 -0600
- (Wed, 19 Nov 2008) | 13 lines Merged revisions 157859 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r157859 | kpfleming | 2008-11-19 15:34:47 -0600 (Wed, 19 Nov
- 2008) | 7 lines 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 ........ ................
-
- * /, res/res_agi.c: Merged revisions 157743 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r157743 |
- kpfleming | 2008-11-19 07:45:48 -0600 (Wed, 19 Nov 2008) | 1 line
- correct small bug introduced during API conversion ........
-
- * apps/app_stack.c, include/asterisk/agi.h, /, channels/chan_sip.c,
- res/res_agi.c, UPGRADE.txt, UPGRADE-1.6.txt (added): Merged
- revisions 157706 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r157706 |
- kpfleming | 2008-11-19 06:42:19 -0600 (Wed, 19 Nov 2008) | 5
- lines make some corrections to the ast_agi_register_multiple(),
- ast_agi_unregister_multiple() and ast_agi_fdprintf() API calls to
- be consistent with API guidelines also, move UPGRADE.txt to
- UPGRADE-1.6.txt and make the new UPGRADE.txt contain information
- about upgrading between Asterisk 1.6 releases ........
-
-2008-11-19 00:33 +0000 [r157601] Sean Bright <sean.bright@gmail.com>
-
- * Makefile, /, build_tools/make_version, configure, configure.ac,
- build_tools/make_buildopts_h, makeopts.in: Merged revisions
- 157600 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r157600 |
- seanbright | 2008-11-18 19:27:45 -0500 (Tue, 18 Nov 2008) | 10
- lines Fix a few build problems on Solaris (and check for an md5
- utility in configure instead of the icky loop I was doing
- before). (closes issue #13842) Reported by: snuffy Patches:
- bug13842_20081106.diff uploaded by snuffy (license 35) 13842.diff
- uploaded by seanbright (license 71) Tested by: snuffy ........
-
-2008-11-18 22:59 +0000 [r157307-157541] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: Merged revisions 157512 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r157512 | mmichelson | 2008-11-18 16:54:08 -0600 (Tue, 18 Nov
- 2008) | 21 lines Merged revisions 157503 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r157503 | mmichelson | 2008-11-18 16:47:57 -0600 (Tue, 18 Nov
- 2008) | 13 lines 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 ........ ................
-
- * channels/chan_sip.c: Once again, Russell to the rescue. Use the
- builtin astobj1 lock of the sip_peer and sip_user instead of
- adding a new one
-
- * /, channels/chan_sip.c: Merged revisions 157496 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r157496 |
- mmichelson | 2008-11-18 15:59:24 -0600 (Tue, 18 Nov 2008) | 6
- lines Based on Russell's advice on the asterisk-dev list, I have
- changed from using a global lock in update_call_counter to using
- the locks within the sip_pvt and sip_peer structures instead.
- ........
-
- * /, channels/chan_sip.c: Merged revisions 157427 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r157427 |
- mmichelson | 2008-11-18 14:23:58 -0600 (Tue, 18 Nov 2008) | 13
- lines * Add a lock to be used in the update_call_counter
- function. * Revert logic to mirror 1.4's in the sense that it
- will not allow the call counter to dip below 0. These two
- measures prevent potential races that could cause a SIP peer to
- appear to be busy forever. (closes issue #13668) Reported by: mjc
- Patches: hintfix_trunk_rev152649.patch uploaded by wolfelectronic
- (license 586) ........
-
- * apps/app_dial.c, channels/chan_local.c, /, main/features.c,
- include/asterisk/channel.h, apps/app_followme.c: Merged revisions
- 157306 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r157306 | mmichelson | 2008-11-18 12:31:08 -0600 (Tue, 18 Nov
- 2008) | 20 lines Merged revisions 157305 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r157305 | mmichelson | 2008-11-18 12:25:55 -0600 (Tue, 18 Nov
- 2008) | 12 lines 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. ........ ................
-
-2008-11-18 18:10 +0000 [r157303] Steve Murphy <murf@digium.com>
-
- * main/config.c, /: Merged revisions 157302 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r157302 |
- murf | 2008-11-18 11:07:55 -0700 (Tue, 18 Nov 2008) | 18 lines
- (closes issue #13420) Reported by: alex70 Patches:
- 13420.13539.patch uploaded by murf (license 17) Tested by: murf,
- awk This fixes two problems: a spurious linefeed insertion
- probably left over from pre-precomment times. Only generated when
- category had no previous comments. The other problem: Insertions
- could get the line-numbering out of whack and generate negative
- line numbers, causing chunks of line numbers to be emitted, on
- the scale of the number of lines up to that point in the file. In
- such cases, abort the looping, and all is well. ........
-
-2008-11-15 19:47 +0000 [r157107-157165] Kevin P. Fleming <kpfleming@digium.com>
-
- * Makefile, /, Makefile.moddir_rules, Makefile.rules: Merged
- revisions 157164 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r157164 | kpfleming | 2008-11-15 20:45:19 +0100 (Sat, 15 Nov
- 2008) | 13 lines Merged revisions 157162-157163 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r157162 | kpfleming | 2008-11-15 20:24:24 +0100 (Sat, 15 Nov
- 2008) | 1 line dist-clean should remove dependency information
- files as well ........ r157163 | kpfleming | 2008-11-15 20:31:03
- +0100 (Sat, 15 Nov 2008) | 1 line 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 ........ ................
-
- * /, contrib/asterisk-ng-doxygen: Merged revisions 157105 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r157105 | kpfleming | 2008-11-15 19:00:32 +0100 (Sat, 15
- Nov 2008) | 13 lines 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 17:03 +0000 [r156912] Tilghman Lesher <tlesher@digium.com>
-
- * main/manager.c, /: Merged revisions 156911 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r156911 |
- tilghman | 2008-11-14 11:02:00 -0600 (Fri, 14 Nov 2008) | 4 lines
- Ping is missing the standard double-newline after the event.
- (closes issue #13903) Reported by: kebl0155 ........
-
-2008-11-14 16:55 +0000 [r156818-156889] Mark Michelson <mmichelson@digium.com>
-
- * include/asterisk/strings.h, apps/app_queue.c: This is the 1.6.0
- version of revision 156883 of trunk. This is different in that it
- preserves the case-sensitiveness of processing queues from
- configuration. closes issue #13703
-
- * apps/app_voicemail.c, /: Merged revisions 156817 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r156817 | mmichelson | 2008-11-14 09:20:03 -0600
- (Fri, 14 Nov 2008) | 18 lines Merged revisions 156816 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r156816 | mmichelson | 2008-11-14 09:18:59 -0600 (Fri, 14 Nov
- 2008) | 10 lines 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! ........ ................
-
-2008-11-13 19:26 +0000 [r156652-156653] Brandon Kruse <bkruse@digium.com>
-
- * main/manager.c: Update to Coding Guidelines
-
- * main/manager.c, /: Merged revisions 156017 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r156017 |
- pari | 2008-11-11 17:02:43 -0600 (Tue, 11 Nov 2008) | 5 lines
- Patch by Ryan Brindley -- Make sure that manager refuses any
- duplicate 'new category' requests in updateconfig (closes issue
- #13539) ........
-
-2008-11-12 19:56 +0000 [r156319] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, /: Merged revisions 156299 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r156299 | murf | 2008-11-12 12:47:29 -0700 (Wed, 12 Nov 2008) |
- 26 lines Merged revisions 156297 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r156297 | murf | 2008-11-12 12:36:16 -0700 (Wed, 12 Nov 2008) |
- 18 lines 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 18:57 +0000 [r156251] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 156243 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r156243 | tilghman | 2008-11-12 12:55:18 -0600
- (Wed, 12 Nov 2008) | 18 lines Merged revisions 156229 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r156229 | tilghman | 2008-11-12 12:39:21 -0600 (Wed, 12 Nov 2008)
- | 11 lines 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 17:47 +0000 [r156170] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 156169 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r156169 | mmichelson | 2008-11-12 11:41:56 -0600 (Wed, 12 Nov
- 2008) | 15 lines Merged revisions 156167 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r156167 | mmichelson | 2008-11-12 11:38:33 -0600 (Wed, 12 Nov
- 2008) | 7 lines 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. ........ ................
-
-2008-11-11 21:28 +0000 [r156012] Russell Bryant <russell@digium.com>
-
- * apps/app_directory.c: Don't blow up if we get NULL when trying to
- parse out the full name field (fixed for Jared in the training
- room)
-
-2008-11-11 20:04 +0000 [r156007] Michiel van Baak <michiel@vanbaak.info>
-
- * /: remove prop that shouldn't be here
-
-2008-11-11 19:49 +0000 [r155815-156004] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_realtime.c: Merged revisions 155862 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r155862 |
- tilghman | 2008-11-10 15:12:28 -0600 (Mon, 10 Nov 2008) | 5 lines
- Make documentation of update method match documentation and
- update update2 method to match. Reported by: atis, via -dev
- mailing list. Fixed by: me ........
-
- * doc/valgrind.txt, /: Merged revisions 155804 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r155803 |
- tilghman | 2008-11-10 14:49:59 -0600 (Mon, 10 Nov 2008) | 1 line
- I got tired of saying this in every single bugnote referring to
- this file. ........
-
-2008-11-09 01:34 +0000 [r155555] Sean Bright <sean.bright@gmail.com>
-
- * apps/app_dial.c, /, main/features.c, include/asterisk/channel.h,
- apps/app_followme.c, apps/app_queue.c: Merged revisions 155554
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r155554 | seanbright | 2008-11-08 20:27:00 -0500
- (Sat, 08 Nov 2008) | 14 lines Merged revisions 155553 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r155553 | seanbright | 2008-11-08 20:08:07 -0500 (Sat, 08 Nov
- 2008) | 6 lines 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 23:42 +0000 [r155361-155468] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 155467 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r155467 |
- mmichelson | 2008-11-07 17:41:44 -0600 (Fri, 07 Nov 2008) | 12
- lines Set the invite state to INV_CANCELLED in a place that makes
- more sense. Where it was set before, it was impossible to
- actually delay sending a CANCEL if we had not yet received a
- provisional response to an INVITE. (closes issue #13626) Reported
- by: atis Patches: 13626.patch uploaded by putnopvut (license 60)
- Tested by: atis ........
-
- * /, configs/voicemail.conf.sample: Merged revisions 155360 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r155360 | mmichelson | 2008-11-07 15:14:49 -0600 (Fri,
- 07 Nov 2008) | 8 lines Remove one more instance of the sample
- configuration lying about what's possible. The tz cannot be set
- in a context like this. It can only be set in the general section
- or per-mailbox. Thanks to sasargen on #asterisk-dev for pointing
- this out ........
-
-2008-11-06 22:50 +0000 [r155123] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, res/ael/ael_lex.c, utils/extconf.c, res/ael/ael.flex: Merged
- revisions 155121 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r155121 |
- kpfleming | 2008-11-06 16:49:19 -0600 (Thu, 06 Nov 2008) | 3
- lines don't blindly assume that Darwin and Cygwin need
- GLOB_ABORTED defined; only define it if it is not already defined
- ........
-
-2008-11-06 19:47 +0000 [r155013] Mark Michelson <mmichelson@digium.com>
-
- * /, configs/voicemail.conf.sample: Merged revisions 155012 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r155012 | mmichelson | 2008-11-06 13:46:53 -0600
- (Thu, 06 Nov 2008) | 16 lines Merged revisions 155011 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r155011 | mmichelson | 2008-11-06 13:45:52 -0600 (Thu, 06 Nov
- 2008) | 8 lines 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. ........ ................
-
-2008-11-03 22:30 +0000 [r154062-154081] Tilghman Lesher <tlesher@digium.com>
-
- * /: Recorded merge of revisions 154072 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r154072 | tilghman | 2008-11-03 16:28:12 -0600 (Mon, 03 Nov 2008)
- | 12 lines Merged revisions 154066 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r154066 | tilghman | 2008-11-03 16:27:10 -0600 (Mon, 03 Nov 2008)
- | 5 lines 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, /: Merged revisions 154060 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r154060 | tilghman | 2008-11-03 15:48:21 -0600 (Mon, 03 Nov 2008)
- | 3 lines Remove the potential for a division by zero error.
- (Closes issue #13810) ........
-
-2008-11-03 00:53 +0000 [r153743-153746] Kevin P. Fleming <kpfleming@digium.com>
-
- * /: record revisions that were manually merged
-
- * apps/app_stack.c, include/asterisk/agi.h, configure,
- include/asterisk/autoconfig.h.in, autoconf/ast_gcc_attribute.m4,
- configure.ac, include/asterisk/compiler.h: Merge revision 153709
- from trunk
- ------------------------------------------------------------------------
- r153709 | kpfleming | 2008-11-02 17:34:39 -0600 (Sun, 02 Nov
- 2008) | 3 lines instead of trying to forcibly load res_agi when
- app_stack is loaded (even if the administrator didn't want it
- loaded), use GCC weak symbols to determine whether it was loaded
- already or not; if it was loaded, then use it.
- ------------------------------------------------------------------------
-
- * channels/chan_oss.c, agi/eagi-sphinx-test.c, res/ael/ael_lex.c,
- channels/chan_h323.c, main/file.c, apps/app_sms.c,
- pbx/pbx_dundi.c, res/ael/ael.flex, pbx/pbx_config.c,
- apps/app_chanspy.c, apps/app_stack.c, utils/streamplayer.c,
- main/asterisk.c, apps/app_voicemail.c, utils/muted.c,
- apps/app_authenticate.c, res/res_phoneprov.c, main/utils.c,
- res/res_musiconhold.c, formats/format_wav_gsm.c,
- res/res_jabber.c, channels/chan_iax2.c, utils/frame.c,
- utils/stereorize.c, main/channel.c, channels/chan_dahdi.c,
- main/manager.c, res/ael/ael.tab.c, funcs/func_odbc.c,
- main/ast_expr2f.c, res/res_agi.c, main/logger.c, main/http.c,
- formats/format_gsm.c, apps/app_adsiprog.c, apps/app_dial.c,
- channels/chan_sip.c, formats/format_wav.c, apps/app_festival.c,
- main/db1-ast/hash/hash_page.c, res/ael/ael.y, res/res_crypto.c,
- agi/eagi-test.c, utils/astman.c, pbx/pbx_lua.c,
- formats/format_ogg_vorbis.c, utils/astcanary.c, apps/app_queue.c:
- port gcc 4.3.x warning fixes from trunk to this branch
-
-2008-10-31 21:49 +0000 [r153265] Terry Wilson <twilson@digium.com>
-
- * apps/app_dial.c, /, main/features.c, include/asterisk/channel.h,
- apps/app_followme.c, apps/app_queue.c: Merged revisions 153181
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r153181 | twilson | 2008-10-31 13:55:33 -0500 (Fri, 31
- Oct 2008) | 5 lines 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 21:00 +0000 [r152994] Sean Bright <sean.bright@gmail.com>
-
- * /, bootstrap.sh: Merged revisions 152993 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r152993 | seanbright | 2008-10-30 16:59:17 -0400 (Thu, 30 Oct
- 2008) | 10 lines Merged revisions 152992 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r152992 | seanbright | 2008-10-30 16:58:24 -0400 (Thu, 30 Oct
- 2008) | 2 lines The -I argument to aclocal needs a space before
- the include directory name. ........ ................
-
-2008-10-30 16:54 +0000 [r152813] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/cdr.c, /: Merged revisions 152812 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r152812 | kpfleming | 2008-10-30 11:54:29 -0500 (Thu, 30 Oct
- 2008) | 9 lines Merged revisions 152811 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r152811 | kpfleming | 2008-10-30 11:53:48 -0500 (Thu, 30 Oct
- 2008) | 3 lines instead of comparing the string pointer to 0,
- let's compare the value that was actually parsed out of the
- string (found by sparse) ........ ................
-
-2008-10-30 04:28 +0000 [r152772] Tilghman Lesher <tlesher@digium.com>
-
- * configs/extensions.conf.sample, /: Merged revisions 152765 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r152765 | tilghman | 2008-10-29 23:26:34 -0500 (Wed, 29
- Oct 2008) | 5 lines Set up an example stdexten that preserves the
- original context and extension in the CDR. (Related to issue
- #13799) Reported by: davidw ........
-
-2008-10-29 20:54 +0000 [r152647] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_directory.c: Merged revisions 152646 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r152646 | mmichelson | 2008-10-29 15:53:53 -0500 (Wed, 29 Oct
- 2008) | 9 lines If there was no named defined in a voicemail.conf
- mailbox entry, then app_directory would crash when attempting to
- read that entry from the file. We now check for the NULL or empty
- string properly so that there will be no crash. (closes issue
- #13804) Reported by: bluecrow76 ........
-
-2008-10-29 20:13 +0000 [r152644] Terry Wilson <twilson@digium.com>
-
- * apps/app_queue.c: Small modification to putnopvut's patch to fix
- this issue. Thanks for all the help, putnopvut! (closes issue
- #12884) Reported by: bcnit Patches: 12884v4-1.6.0-branch.patch
- uploaded by otherwiseguy (license 396) Tested by: otherwiseguy
-
-2008-10-28 21:39 +0000 [r152443] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_mgcp.c, /: Merged revisions 152442 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r152442 | tilghman | 2008-10-28 16:38:26 -0500 (Tue, 28 Oct 2008)
- | 7 lines Only re-add the io port if it was closed, otherwise
- reload causes a memory leak. (closes issue #13785) Reported by:
- eliel Patches: chan_mgcp.c.patch uploaded by eliel (license 64)
- ........
-
-2008-10-27 16:33 +0000 [r152157] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_stack.c, /: Merged revisions 152134 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r152134 |
- tilghman | 2008-10-27 11:24:11 -0500 (Mon, 27 Oct 2008) | 4 lines
- Oops, only delete the ARG variables once upon release. The
- following section would have removed them again (removing
- variables from 2 stack frames, instead of just one). ........
-
-2008-10-26 20:26 +0000 [r152062] Sean Bright <sean.bright@gmail.com>
-
- * /, funcs/func_strings.c: Merged revisions 152060 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r152060 | seanbright | 2008-10-26 16:25:08 -0400
- (Sun, 26 Oct 2008) | 15 lines Merged revisions 152059 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r152059 | seanbright | 2008-10-26 16:23:36 -0400 (Sun, 26 Oct
- 2008) | 7 lines 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 ........ ................
-
-2008-10-23 16:12 +0000 [r151765] Russell Bryant <russell@digium.com>
-
- * channels/chan_sip.c: Fix some memory leaks. These issues are
- 1.6.0 specific. - Freeing the peer got accidentally removed from
- the peer's destructor. It is still needed for astobj, but not for
- astobj2. - Fix some places that called find_user or find_peer,
- but did not release the reference that was returned. (closes
- issue #13331) Reported by: sergee Patches:
- chan_sip-3leaks-16-r151244.diff uploaded by sergee (license 138)
- Tested by: sergee
-
-2008-10-20 05:03 +0000 [r151244] Kevin P. Fleming <kpfleming@digium.com>
-
- * autoconf (added), autoconf/ast_check_pwlib.m4,
- autoconf/acx_pthread.m4, autoconf/ast_func_fork.m4, configure,
- autoconf/ast_gcc_attribute.m4, bootstrap.sh,
- autoconf/ast_check_gnu_make.m4, autoconf/ast_ext_lib.m4,
- autoconf/ast_prog_ld.m4, autoconf/ast_c_compile_check.m4,
- autoconf/ast_c_define_check.m4, autoconf/ast_prog_egrep.m4,
- autoconf/ast_ext_tool_check.m4, autoconf/ast_check_mandatory.m4,
- /, autoconf/ast_check_openh323.m4, autoconf/ast_prog_ld_gnu.m4,
- configure.ac, acinclude.m4 (removed), autoconf/ast_prog_sed.m4:
- Merged revisions 151242-151243 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r151242 | kpfleming | 2008-10-20 07:59:04 +0300 (Mon, 20 Oct
- 2008) | 9 lines Merged revisions 151240 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r151240 | kpfleming | 2008-10-20 07:45:56 +0300 (Mon, 20 Oct
- 2008) | 3 lines 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 ........ ................ r151243 | kpfleming |
- 2008-10-20 08:00:56 +0300 (Mon, 20 Oct 2008) | 9 lines Merged
- revisions 151241 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r151241 | kpfleming | 2008-10-20 07:57:33 +0300 (Mon, 20 Oct
- 2008) | 2 lines rename this macro to properly reflect what it
- does ........ ................
-
-2008-10-18 02:35 +0000 [r150854] BJ Weschke <bweschke@btwtech.com>
-
- * main/manager.c, /: Merged revisions 150817 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r150817 |
- bweschke | 2008-10-17 22:18:33 -0400 (Fri, 17 Oct 2008) | 8 lines
- 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 00:19 +0000 [r150308-150313] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: Instead of merging commit 150307 to 1.6.0, I
- had meant to block it in 1.6.1...time to go home :)
-
- * /, channels/chan_sip.c: Merged revisions 150307 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r150307 |
- mmichelson | 2008-10-16 19:13:35 -0500 (Thu, 16 Oct 2008) | 14
- lines After a long discussion on #asterisk-bugs, it seems kind of
- odd that a channel would be named after the port on which it came
- in on. For endpoints that always include ":5060" as part of the
- From: header, it will mean that you have a ton of channels with
- names like "SIP/5060-3ea38a8b." I am boldly moving forward with
- this change in trunk, but I'm not touching other branches with
- this one since this definitely would qualify as a behavior
- change. If there is a problem with this commit, and I haven't
- seen the obvious reason why you'd want to name the channel after
- the port from which the call originated, then please feel free to
- revert this ........
-
-2008-10-16 16:10 +0000 [r150126] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_misdn.c, /: Merged revisions 150125 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r150125 | rmudgett | 2008-10-16 11:04:45 -0500
- (Thu, 16 Oct 2008) | 9 lines Merged revisions 150124 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r150124 | rmudgett | 2008-10-16 10:56:06 -0500 (Thu, 16
- Oct 2008) | 1 line Fix memory leak found by customer ........
- ................
-
-2008-10-15 20:18 +0000 [r149757] BJ Weschke <bweschke@btwtech.com>
-
- * configs/agents.conf.sample, /: Merged revisions 149756 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r149756 | bweschke | 2008-10-15 16:14:20 -0400
- (Wed, 15 Oct 2008) | 10 lines Merged revisions 149683 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r149683 | bweschke | 2008-10-15 14:28:54 -0400 (Wed, 15 Oct 2008)
- | 4 lines 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 11:29 +0000 [r149495] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 149487 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r149487 | kpfleming | 2008-10-15 13:26:36 +0200 (Wed, 15 Oct
- 2008) | 9 lines Merged revisions 149452 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r149452 | kpfleming | 2008-10-15 12:30:40 +0200 (Wed, 15 Oct
- 2008) | 3 lines fix some problems when parsing SIP messages that
- have the maximum number of headers or body lines that we support
- ........ ................
-
-2008-10-14 17:39 +0000 [r148914] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_local.c, /: Merged revisions 148913 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r148913 | mmichelson | 2008-10-14 12:38:06 -0500
- (Tue, 14 Oct 2008) | 17 lines Merged revisions 148912 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r148912 | mmichelson | 2008-10-14 12:33:38 -0500 (Tue, 14 Oct
- 2008) | 9 lines 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 10:34 +0000 [r148613-148739] Kevin P. Fleming <kpfleming@digium.com>
-
- * Makefile, /: Merged revisions 148738 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r148738 | kpfleming | 2008-10-14 12:33:14 +0200 (Tue, 14 Oct
- 2008) | 9 lines Merged revisions 148736 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r148736 | kpfleming | 2008-10-14 12:30:54 +0200 (Tue, 14 Oct
- 2008) | 3 lines 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: Merged revisions 148612 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r148612 | kpfleming | 2008-10-14 03:06:45 -0500 (Tue, 14 Oct
- 2008) | 9 lines Merged revisions 148611 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r148611 | kpfleming | 2008-10-14 02:54:41 -0500 (Tue, 14 Oct
- 2008) | 3 lines it would be nice if this message printing code
- had actually been tested before it was committed... ........
- ................
-
-2008-10-10 21:18 +0000 [r148374] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 148373 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r148373 |
- mmichelson | 2008-10-10 16:18:10 -0500 (Fri, 10 Oct 2008) | 8
- lines Make sure that the inUse and inRinging fields for a sip
- peer cannot go below zero. This is a regression from 1.4 and so
- it will be applied to 1.6.0 as well. (closes issue #13668)
- Reported by: mjc ........
-
-2008-10-10 01:25 +0000 [r148201-148204] Sean Bright <sean.bright@gmail.com>
-
- * res/res_config_sqlite.c, apps/app_voicemail.c,
- include/asterisk.h, /, main/tdd.c, main/cryptostub.c: Merged
- revisions 148200 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r148200 |
- seanbright | 2008-10-09 20:42:13 -0400 (Thu, 09 Oct 2008) | 12
- lines Don't include logger.h in asterisk.h by default as it is
- causing problems building app_voicemail. Instead, include it
- where it is needed. This turned out to be a relatively minor
- issue because other headers include logger.h as well. Need to
- test -addons before merging this back to 1.6.0. (closes issue
- #13605) Reported by: tomo1657 Patches: 13605_seanbright.diff
- uploaded by seanbright (license 71) Tested by: mmichelson
- ........
-
- * apps/app_rpt.c: Somehow we got conflict markers checked in! Might
- need a 1.6.0.1 sooner than we'd like.
-
-2008-10-09 23:31 +0000 [r148147] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 148144 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r148144 | mmichelson | 2008-10-09 18:30:47 -0500 (Thu, 09 Oct
- 2008) | 10 lines Read the callerid in the correct order and make
- sure to read the Urgent flag value from the IMAP headers. (closes
- issue #13652) Reported by: jaroth Patches: imapheaders.patch
- uploaded by jaroth (license 50) ........
-
-2008-10-09 23:26 +0000 [r148124] Tilghman Lesher <tlesher@digium.com>
-
- * /, configs/res_ldap.conf.sample: Merged revisions 148120 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r148120 | tilghman | 2008-10-09 18:25:53 -0500 (Thu, 09
- Oct 2008) | 6 lines Fix example schema (closes issue #12860)
- Reported by: flyn Patches: res_ldap.conf.patch uploaded by flyn
- (license 503) ........
-
-2008-10-09 17:51 +0000 [r147900] Michiel van Baak <michiel@vanbaak.info>
-
- * include/asterisk/endian.h, /: Merged revisions 147899 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r147899 | mvanbaak | 2008-10-09 19:48:53 +0200 (Thu, 09
- Oct 2008) | 5 lines only include this for OpenBSD. At least
- FreeBSD is borked when including it (closes issue #13649)
- Reported by: ys ........
-
-2008-10-09 17:47 +0000 [r147897] Tilghman Lesher <tlesher@digium.com>
-
- * configs/extensions.conf.sample, /: Merged revisions 147896 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r147896 | tilghman | 2008-10-09 12:46:15 -0500 (Thu, 09
- Oct 2008) | 4 lines Remove "second form" of extensions, as it no
- longer applies. Also, cleanup the grammar, formatting, and
- introduce several clarifications to the text. (Closes issue
- #13654) ........
-
-2008-10-09 14:56 +0000 [r147809] Steve Murphy <murf@digium.com>
-
- * main/astobj2.c, channels/chan_oss.c, main/config.c, main/rtp.c,
- main/cli.c, configure, channels/console_gui.c, utils/extconf.c,
- main/pbx.c, include/asterisk.h, doc/CODING-GUIDELINES, /,
- include/asterisk/autoconfig.h.in, main/translate.c,
- channels/vcodecs.c, configure.ac, channels/console_video.c,
- channels/chan_iax2.c: Merged revisions 147807 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r147807 |
- murf | 2008-10-09 08:17:33 -0600 (Thu, 09 Oct 2008) | 15 lines
- (closes issue #13557) Reported by: nickpeirson Patches:
- pbx.c.patch uploaded by nickpeirson (license 579)
- replace_bzero+bcopy.patch uploaded by nickpeirson (license 579)
- Tested by: nickpeirson, murf 1. replaced all refs to bzero and
- bcopy to memset and memmove instead. 2. added a note to the
- CODING-GUIDELINES 3. add two macros to asterisk.h to prevent
- bzero, bcopy from creeping back into the source 4. removed bzero
- from configure, configure.ac, autoconfig.h.in ........
-
-2008-10-08 12:16 +0000 [r147458] Russell Bryant <russell@digium.com>
-
- * configs/chan_dahdi.conf.sample: Remove the sample configuration
- for configuration sections in chan_dahdi.conf. This code was not
- merged into 1.6.0. Reported by: angler (closes AST-119)
-
-2008-10-08 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.6.0.1 released.
-
- * configs/chan_dahdi.conf.sample: Remove mention of configuration
- sections for defining channels in chan_dahdi.conf. This code
- is in 1.6.1, and was not merged into 1.6.0.
-
-2008-10-01 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.6.0 released.
-
-2008-09-09 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.6.0-rc6 released.
-
-2008-09-09 15:44 +0000 [r142065] Russell Bryant <russell@digium.com>
-
- * /, main/features.c: Merged revisions 142064 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r142064 | russell | 2008-09-09 10:44:10 -0500 (Tue, 09 Sep 2008)
- | 13 lines Merged revisions 142063 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r142063 | russell | 2008-09-09 10:40:24 -0500 (Tue, 09 Sep 2008)
- | 5 lines 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) ........ ................
-
-2008-09-09 12:34 +0000 [r141996-141999] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_oss.c, /: Merged revisions 141995 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r141995 |
- mmichelson | 2008-09-09 05:20:58 -0500 (Tue, 09 Sep 2008) | 8
- lines Fix a memory leak in chan_oss (closes issue #13311)
- Reported by: eliel Patches: chan_oss.c.patch uploaded by eliel
- (license 64) ........
-
-2008-09-09 01:49 +0000 [r141950] Russell Bryant <russell@digium.com>
-
- * main/channel.c, /: Merged revisions 141949 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r141949 |
- russell | 2008-09-08 20:47:56 -0500 (Mon, 08 Sep 2008) | 9 lines
- Modify ast_answer() to not hold the channel lock while calling
- ast_safe_sleep() or when calling ast_waitfor(). These are
- inappropriate times to hold the channel lock. This is what has
- caused "could not get the channel lock" messages from chan_sip
- and has likely caused a negative impact on performance results of
- SIP in Asterisk 1.6. Thanks to file for pointing out this section
- of code. (closes issue #13287) (closes issue #13115) ........
-
-2008-09-08 21:07 +0000 [r141808] Russell Bryant <russell@digium.com>
-
- * main/pbx.c, /: Merged revisions 141807 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r141807 | russell | 2008-09-08 16:05:01 -0500 (Mon, 08 Sep 2008)
- | 15 lines Merged revisions 141806 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r141806 | russell | 2008-09-08 16:02:36 -0500 (Mon, 08 Sep 2008)
- | 7 lines 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 1.6.0-rc5 released.
-
-2008-09-08 20:19 +0000 [r141746] Jason Parker <jparker@digium.com>
-
- * Makefile, /, redhat (removed): Merged revisions 141745 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r141745 | qwell | 2008-09-08 15:18:17 -0500
- (Mon, 08 Sep 2008) | 16 lines Merged revisions 141741 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r141741 | qwell | 2008-09-08 15:15:42 -0500 (Mon, 08 Sep 2008) |
- 8 lines 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 17:14 +0000 [r141683] Sean Bright <sean.bright@gmail.com>
-
- * /, build_tools/make_buildopts_h: Merged revisions 141682 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r141682 | seanbright | 2008-09-08 13:13:04 -0400 (Mon,
- 08 Sep 2008) | 9 lines Quote the arguments to grep so that sh on
- various platforms doesn't choke on the special characters (like
- ^). (closes issue #13417) Reported by: dougm Patches:
- 13417.make_buildopts_h.patch uploaded by seanbright (license 71)
- Tested by: dougm ........
-
-2008-09-06 20:21 +0000 [r141567] Steve Murphy <murf@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 141566 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r141566 | murf | 2008-09-06 14:19:50 -0600 (Sat, 06 Sep 2008) | 9
- lines Merged revisions 141565 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r141565 | murf | 2008-09-06 14:13:16 -0600 (Sat, 06 Sep 2008) | 1
- line 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 15:40 +0000 [r141505-141508] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_agi.c: Merged revisions 141504 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r141504 | tilghman | 2008-09-06 10:26:45 -0500 (Sat, 06 Sep 2008)
- | 12 lines Merged revisions 141503 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r141503 | tilghman | 2008-09-06 10:23:42 -0500 (Sat, 06 Sep 2008)
- | 4 lines Reverting behavior change (AGI should not exit non-zero
- on SUCCESS) (closes issue #13434) Reported by: francesco_r
- ........ ................
-
-2008-09-05 22:06 +0000 [r141368-141426] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 141367 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r141367 | mmichelson | 2008-09-05 16:12:09 -0500
- (Fri, 05 Sep 2008) | 15 lines Merged revisions 141366 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r141366 | mmichelson | 2008-09-05 16:10:32 -0500 (Fri, 05 Sep
- 2008) | 7 lines 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 ........ ................
-
-2008-09-05 14:24 +0000 [r141116-141158] Steve Murphy <murf@digium.com>
-
- * main/channel.c, /: Merged revisions 141157 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r141157 | murf | 2008-09-05 08:18:43 -0600 (Fri, 05 Sep 2008) | 9
- lines Merged revisions 141156 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r141156 | murf | 2008-09-05 08:15:43 -0600 (Fri, 05 Sep 2008) | 1
- line 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,
- pbx/ael/ael-test/ael-vtest25 (added), res/ael/ael_lex.c,
- pbx/ael/ael-test/ref.ael-test6, res/ael/ael.flex: Merged
- revisions 141115 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r141115 | murf | 2008-09-04 17:31:41 -0600 (Thu, 04 Sep 2008) |
- 78 lines Merged revisions 141094 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r141094 | murf | 2008-09-04 17:15:07 -0600 (Thu, 04 Sep 2008) |
- 70 lines (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 18:35 +0000 [r141086] Jeff Peeler <jpeeler@digium.com>
-
- * /, main/features.c, res/res_agi.c: Merged revisions 141039 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r141039 | jpeeler | 2008-09-04 12:27:56 -0500
- (Thu, 04 Sep 2008) | 15 lines Merged revisions 141028 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r141028 | jpeeler | 2008-09-04 12:00:29 -0500 (Thu, 04 Sep 2008)
- | 7 lines (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 20:18 +0000 [r140976] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 140975 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r140975 |
- mmichelson | 2008-09-03 15:16:12 -0500 (Wed, 03 Sep 2008) | 4
- lines Fix some locking order issues in app_queue. This was
- brought up by atis on IRC a while ago. ........
-
-2008-09-03 Russell Bryant <russell@digium.com>
-
- * Asterisk 1.6.0-rc4 released.
-
-2008-09-03 14:17 +0000 [r140825-140827] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, /: Merged revisions 140749 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r140749 | murf | 2008-09-02 17:44:04 -0600 (Tue, 02 Sep 2008) |
- 11 lines Merged revisions 140747 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r140747 | murf | 2008-09-02 17:36:56 -0600 (Tue, 02 Sep 2008) | 1
- line 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. For trunk, I am going one step further.
- These messages were pretty worthless even for debug, so I'm
- completely removing them. ........ ................
-
- * main/channel.c, /: Merged revisions 140692 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r140692 | murf | 2008-09-02 16:55:12 -0600 (Tue, 02 Sep 2008) |
- 13 lines Merged revisions 140690 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r140690 | murf | 2008-09-02 16:40:13 -0600 (Tue, 02 Sep 2008) | 1
- line 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. Hmmm. It doesn't hush the useless warnings, but it
- does allow control of posting via the detach and post routines,
- for those possible situations, where you'd want to post
- single-channel cdrs. ........ ................
-
- * main/channel.c, main/pbx.c, /: Merged revisions 140691 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r140691 | murf | 2008-09-02 16:50:59 -0600 (Tue,
- 02 Sep 2008) | 22 lines Merged revisions 140670 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r140670 | murf | 2008-09-02 16:15:57 -0600 (Tue, 02 Sep 2008) |
- 14 lines (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. ........ ................
-
-2008-09-03 13:27 +0000 [r140818] Russell Bryant <russell@digium.com>
-
- * main/poll.c, /: Merged revisions 140817 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r140817 | russell | 2008-09-03 08:26:43 -0500 (Wed, 03 Sep 2008)
- | 12 lines Merged revisions 140816 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r140816 | russell | 2008-09-03 08:24:35 -0500 (Wed, 03 Sep 2008)
- | 4 lines Don't freak out if the poll emulation receives NULL for
- the pollfds array (closes issue #13307) Reported by: jcovert
- ........ ................
-
-2008-09-02 18:17 +0000 [r140607] Sean Bright <sean.bright@gmail.com>
-
- * /, channels/chan_iax2.c: Merged revisions 140606 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r140606 | seanbright | 2008-09-02 14:15:54 -0400
- (Tue, 02 Sep 2008) | 16 lines Merged revisions 140605 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r140605 | seanbright | 2008-09-02 14:14:57 -0400 (Tue, 02 Sep
- 2008) | 8 lines 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) ........ ................
-
-2008-09-02 15:12 +0000 [r140564-140567] Russell Bryant <russell@digium.com>
-
- * apps/app_jack.c, /, codecs/codec_resample.c: Merged revisions
- 140566 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r140566 |
- russell | 2008-09-02 10:11:53 -0500 (Tue, 02 Sep 2008) | 2 lines
- Update instructions for getting libresample ........
-
-2008-08-27 20:15 +0000 [r140302-140304] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: Revert commit 140302. Should not be merging
- changes like that into a release-candidate branch
-
- * channels/chan_sip.c: Merged revisions 140301 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r140301 | mmichelson | 2008-08-27 15:11:22 -0500 (Wed, 27 Aug
- 2008) | 19 lines Merged revisions 140299 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r140299 | mmichelson | 2008-08-27 14:49:20 -0500 (Wed, 27 Aug
- 2008) | 11 lines 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 18:12 +0000 [r140170] Russell Bryant <russell@digium.com>
-
- * Makefile, /: Merged revisions 140169 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r140169 |
- russell | 2008-08-26 13:11:49 -0500 (Tue, 26 Aug 2008) | 4 lines
- Fix building menuselect-tree with PRINT_DIR set. We _must_ use
- the --quiet flag here, or else some arbitrary text will end up in
- the resulting menuselect-tree file and things will explode.
- ........
-
-2008-08-25 21:33 +0000 [r139918] Sean Bright <sean.bright@gmail.com>
-
- * build_tools/get_moduleinfo, /, build_tools/get_makeopts: Merged
- revisions 139915 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r139915 | seanbright | 2008-08-25 17:32:10 -0400 (Mon, 25 Aug
- 2008) | 17 lines Merged revisions 139909 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r139909 | seanbright | 2008-08-25 17:31:03 -0400 (Mon, 25 Aug
- 2008) | 9 lines 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 21:05 +0000 [r139872] Terry Wilson <twilson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 139870 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r139870 | twilson | 2008-08-25 15:59:58 -0500 (Mon, 25 Aug 2008)
- | 10 lines Merged revisions 139869 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r139869 | twilson | 2008-08-25 15:46:10 -0500 (Mon, 25 Aug 2008)
- | 2 lines Make SIPADDHEADER() propagate indefinitely ........
- ................
-
-2008-08-25 16:00 +0000 [r139774] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, /, main/features.c: Merged revisions 139770 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r139770 | murf | 2008-08-25 09:54:18 -0600 (Mon,
- 25 Aug 2008) | 17 lines Merged revisions 139764 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r139764 | murf | 2008-08-25 09:33:14 -0600 (Mon, 25 Aug 2008) | 9
- lines 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! ........
- ................
-
-2008-08-24 16:30 +0000 [r139705-139708] Tilghman Lesher <tlesher@digium.com>
-
- * /, cdr/cdr_pgsql.c: Merged revisions 139707 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r139707 |
- tilghman | 2008-08-24 11:26:48 -0500 (Sun, 24 Aug 2008) | 2 lines
- Memory leak ........
-
-2008-08-22 22:35 +0000 [r139628-139671] Steve Murphy <murf@digium.com>
-
- * /, main/features.c: Merged revisions 139662 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r139662 | murf | 2008-08-22 16:32:35 -0600 (Fri, 22 Aug 2008) |
- 14 lines Merged revisions 139635 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r139635 | murf | 2008-08-22 16:24:02 -0600 (Fri, 22 Aug 2008) | 6
- lines 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. ........
- ................
-
- * apps/app_dial.c, main/pbx.c, /, main/features.c: Merged revisions
- 139627 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r139627 | murf | 2008-08-22 16:03:13 -0600 (Fri, 22 Aug 2008) |
- 59 lines Merged revisions 139347 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r139347 | murf | 2008-08-21 17:03:50 -0600 (Thu, 21 Aug 2008) |
- 47 lines (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! ........ I also made a little fix to
- the app_dial's 'e' option, that is related to my updates.
- ................
-
-2008-08-22 20:21 +0000 [r139458-139564] Mark Michelson <mmichelson@digium.com>
-
- * include/asterisk/threadstorage.h, /: Merged revisions 139554 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r139554 | mmichelson | 2008-08-22 14:45:41 -0500
- (Fri, 22 Aug 2008) | 16 lines Merged revisions 139553 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r139553 | mmichelson | 2008-08-22 14:45:19 -0500 (Fri, 22 Aug
- 2008) | 8 lines Fix compilation when DEBUG_THREAD_LOCALS is
- selected (closes issue #13298) Reported by: snuffy Patches:
- bug13298_20080822.diff uploaded by snuffy (license 35) ........
- ................
-
- * /, channels/chan_iax2.c: Merged revisions 139469 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r139469 | mmichelson | 2008-08-22 12:25:12 -0500
- (Fri, 22 Aug 2008) | 11 lines Merged revisions 139466 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r139466 | mmichelson | 2008-08-22 12:24:47 -0500 (Fri, 22 Aug
- 2008) | 3 lines Fix the build. Thanks, mvanbaak! ........
- ................
-
- * /, channels/chan_iax2.c: Merged revisions 139457 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r139457 | mmichelson | 2008-08-22 11:58:21 -0500
- (Fri, 22 Aug 2008) | 15 lines Merged revisions 139456 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r139456 | mmichelson | 2008-08-22 11:57:38 -0500 (Fri, 22 Aug
- 2008) | 7 lines Prevent a deadlock in chan_iax2 resulting from
- incorrect locking order between iax2_pvt and ast_channel
- structures. AST-13 ........ ................
-
-2008-08-21 23:46 +0000 [r139400] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 139391 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r139391 | jpeeler | 2008-08-21 18:41:50 -0500
- (Thu, 21 Aug 2008) | 11 lines Merged revisions 139387 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r139387 | jpeeler | 2008-08-21 18:39:31 -0500 (Thu, 21 Aug 2008)
- | 3 lines 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 10:02 +0000 [r139282] Philippe Sultan <philippe.sultan@gmail.com>
-
- * /, channels/chan_gtalk.c: Merged revisions 139281 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r139281 | phsultan | 2008-08-21 11:55:31 +0200 (Thu, 21 Aug 2008)
- | 5 lines Fix two memory leaks in chan_gtalk, thanks Eliel!
- (closes issue #13310) Reported by: eliel Patches:
- chan_gtalk.c.patch uploaded by eliel (license 64) ........
-
-2008-08-20 Kevin P. Fleming <kpfleming@digium.com>
-
- * Asterisk 1.6.0-rc3 released.
-
-2008-08-20 22:17 +0000 [r139216] Russell Bryant <russell@digium.com>
-
- * apps/app_chanspy.c, /: Merged revisions 139215 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r139215 | russell | 2008-08-20 17:16:36 -0500 (Wed, 20 Aug 2008)
- | 19 lines Merged revisions 139213 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r139213 | russell | 2008-08-20 17:14:35 -0500 (Wed, 20 Aug 2008)
- | 11 lines 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 20:12 +0000 [r139155] 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 17:30 +0000 [r139104] Steve Murphy <murf@digium.com>
-
- * main/cdr.c, /: Merged revisions 139083 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r139083 | murf | 2008-08-20 11:25:07 -0600 (Wed, 20 Aug 2008) |
- 20 lines Merged revisions 139074 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r139074 | murf | 2008-08-20 11:14:55 -0600 (Wed, 20 Aug 2008) |
- 12 lines (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 15:39 +0000 [r138889-139017] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 139016 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r139016 | mmichelson | 2008-08-20 10:38:47 -0500 (Wed, 20 Aug
- 2008) | 14 lines Merged revisions 139015 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r139015 | mmichelson | 2008-08-20 10:37:56 -0500 (Wed, 20 Aug
- 2008) | 6 lines sip_read should properly handle a NULL return
- from sip_rtp_read. (closes issue #13257) Reported by: travishein
- ........ ................
-
- * apps/app_chanspy.c: Manually add revision 138887 from trunk to
- the 1.6.0 branch. I had misunderstood the policy for when to
- merge to 1.6.0 since it moved to rc status.
-
-2008-08-19 16:38 +0000 [r138846-138847] Steve Murphy <murf@digium.com>
-
- * utils/conf2ael.c, /, res/ael/ael.tab.c, res/ael/ael.y,
- res/ael/ael.tab.h, utils/ael_main.c: Merged revisions 138845 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r138845 | murf | 2008-08-19 10:31:24 -0600 (Tue, 19 Aug
- 2008) | 1 line Oops. put a decl in a generated file. My bad, but
- fixed now. ........
-
- * main/pbx.c, /, res/ael/ael.tab.c, res/ael/ael.y,
- res/ael/ael.tab.h: Merged revisions 138815 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r138815 |
- murf | 2008-08-19 09:59:12 -0600 (Tue, 19 Aug 2008) | 19 lines
- These changes are in regards to bug 13249, where users are being
- surprised by the changes made to the Set app in trunk/1.6.x, as
- they come from the 1.4 world. They are only bitten if they write
- their AEL dialplan in the 1.4 world, and then carry it over to a
- trunk/1.6.x installation where a "make samples" was executed, or
- where they hand-edited the asterisk.conf file and added the
- [compat] category with app_set = 1.6 (or higher). (this commit
- does not totally solve 13249, at least not yet) The change
- involves issueing a single warning while the AEL file is loading,
- if: 1. app_set is present in the config file, and set to 1.6 or
- higher. 2. there are double quotes in an assignment statement (eg
- x = "hi there";) 3. the warning was not already issued. The
- standalone app, aelparse, does not (yet) issue this warning. I'd
- have to have it read in the asterisk.conf file, and that's a bit
- of hassle. I'll add it if users request it, tho. ........
-
-2008-08-19 00:15 +0000 [r138776-138781] Sean Bright <sean.bright@gmail.com>
-
- * /, channels/chan_sip.c: Merged revisions 138778-138780 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r138778 | seanbright | 2008-08-18 20:08:27 -0400 (Mon,
- 18 Aug 2008) | 1 line While we're at it, make this machine
- parseable too. ........ r138779 | seanbright | 2008-08-18
- 20:09:38 -0400 (Mon, 18 Aug 2008) | 1 line And remove code we
- don't need anymore. ........ r138780 | seanbright | 2008-08-18
- 20:10:56 -0400 (Mon, 18 Aug 2008) | 1 line Let it compile now,
- too (woops) ........
-
- * /, channels/chan_sip.c: Merged revisions 138775 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r138775 |
- seanbright | 2008-08-18 19:42:36 -0400 (Mon, 18 Aug 2008) | 3
- lines Change event header to RegistrationTime to be more
- consistent (and avoid breaking existing frameworks). Pointed out
- by Laureano on #asterisk-dev. ........
-
-2008-08-18 20:23 +0000 [r138688-138695] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 138687 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r138687 | mmichelson | 2008-08-18 15:04:10 -0500 (Mon, 18 Aug
- 2008) | 18 lines Merged revisions 138685 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r138685 | mmichelson | 2008-08-18 15:01:14 -0500 (Mon, 18 Aug
- 2008) | 10 lines 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 15:54 +0000 [r138632] Jason Parker <jparker@digium.com>
-
- * Makefile, /: Merged revisions 138631 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r138631 |
- qwell | 2008-08-18 10:54:07 -0500 (Mon, 18 Aug 2008) | 1 line
- Remove option that isn't valid here. ........
-
-2008-08-18 02:14 +0000 [r138519] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 138518 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r138518 | jpeeler | 2008-08-17 21:13:04 -0500 (Sun, 17 Aug 2008)
- | 1 line add missing define for SS7 in dahdi_restart ........
-
-2008-08-17 14:14 +0000 [r138443-138483] Sean Bright <sean.bright@gmail.com>
-
- * /, main/features.c: Merged revisions 138482 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r138482 |
- seanbright | 2008-08-17 10:12:11 -0400 (Sun, 17 Aug 2008) | 6
- lines Move Uniqueid to the end of the event for those that rely
- on the position of the name/value pairs, pointed out by
- snuffy-home on #asterisk-commits. For those of you who rely on
- the position of name/value pairs in manager events... stop...
- that is why associative arrays were invented. ........
-
- * /, main/features.c: Merged revisions 138479 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r138479 |
- seanbright | 2008-08-17 09:51:08 -0400 (Sun, 17 Aug 2008) | 7
- lines Add Uniqueid header to ParkedCall manager event. (closes
- issue #13323) Reported by: srt Patches:
- 13323_unique_id_for_parkedcalls_event.diff uploaded by srt
- (license 378) ........
-
- * main/rtp.c, /: Merged revisions 138476 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r138476 |
- seanbright | 2008-08-17 09:40:36 -0400 (Sun, 17 Aug 2008) | 7
- lines Add missing colons to RTCPReceived and RTCPSent manager
- events. (closes issue #13319) Reported by: srt Patches:
- 13319_rtcp_manager_event_headers.diff uploaded by srt (license
- 378) ........
-
- * /, channels/chan_iax2.c: Merged revisions 138473 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r138473 | seanbright | 2008-08-17 09:31:54 -0400 (Sun, 17 Aug
- 2008) | 7 lines Fix the output of the JitterBufStats manager
- event. (closes issue #13324) Reported by: srt Patches:
- 13324_missing_nl_in_jitterbufstats_event_2.diff uploaded by srt
- (license 378) ........
-
- * configs/cdr_tds.conf.sample, /: Merged revisions 138442 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r138442 | seanbright | 2008-08-16 12:40:43 -0400 (Sat,
- 16 Aug 2008) | 4 lines Since it's introduction in revision 3497,
- cdr_tds has *never* read the port configuration option from
- cdr_tds.conf. So go ahead and remove it from the sample config.
- ........
-
-2008-08-16 13:07 +0000 [r138410-138413] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 138412 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r138412 | tilghman | 2008-08-16 08:07:08 -0500 (Sat, 16 Aug 2008)
- | 2 lines Fix compilation warnings (found with dev-mode) ........
-
-2008-08-16 01:14 +0000 [r138333-138362] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 138361 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r138361 | jpeeler | 2008-08-15 20:13:26 -0500
- (Fri, 15 Aug 2008) | 9 lines Merged revisions 138360 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r138360 | jpeeler | 2008-08-15 20:12:18 -0500 (Fri, 15
- Aug 2008) | 1 line fixes use count to properly decrement if an
- active dahdi channel is destroyed allowing module to be unloaded
- ........ ................
-
- * channels/chan_dahdi.c, /: Merged revisions 138311 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r138311 | jpeeler | 2008-08-15 18:46:09 -0500
- (Fri, 15 Aug 2008) | 20 lines Merged revisions
- 138119,138151,138238 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r138119 | jpeeler | 2008-08-15 14:21:51 -0500 (Fri, 15 Aug 2008)
- | 4 lines 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) ........ r138151 | jpeeler |
- 2008-08-15 14:41:29 -0500 (Fri, 15 Aug 2008) | 1 line declared
- static mutexes using AST_MUTEX_DEFINE_STATIC macro ........
- r138238 | jpeeler | 2008-08-15 16:28:26 -0500 (Fri, 15 Aug 2008)
- | 1 line initialize condition variable ss_thread_complete using
- ast_cond_init ........ ................
-
-2008-08-15 23:03 +0000 [r138207-138262] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
- 138260 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r138260 | tilghman | 2008-08-15 17:54:57 -0500 (Fri, 15 Aug 2008)
- | 16 lines Merged revisions 138258 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r138258 | tilghman | 2008-08-15 17:33:42 -0500 (Fri, 15 Aug 2008)
- | 8 lines 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 ........
- ................
-
- * configs/extensions.conf.sample, main/pbx.c, /: Merged revisions
- 138206 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r138206 |
- tilghman | 2008-08-15 15:35:24 -0500 (Fri, 15 Aug 2008) | 4 lines
- Remove deprecated syntax from sample config file (closes issue
- #13314) Reported by: kue ........
-
-2008-08-15 20:20 +0000 [r138156-138157] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c: rename all zfd instances in chan_dahdi to
- dfd to match 1.4 (left over from DAHDI transition)
-
-2008-08-15 15:12 +0000 [r138029] Russell Bryant <russell@digium.com>
-
- * main/autoservice.c, /: Merged revisions 138028 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r138028 | russell | 2008-08-15 10:09:46 -0500 (Fri, 15 Aug 2008)
- | 17 lines Merged revisions 138027 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r138027 | russell | 2008-08-15 10:07:16 -0500 (Fri, 15 Aug 2008)
- | 9 lines 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 15:04 +0000 [r138025] Tilghman Lesher <tlesher@digium.com>
-
- * /, funcs/func_strings.c: Merged revisions 138024 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r138024 | tilghman | 2008-08-15 10:03:32 -0500
- (Fri, 15 Aug 2008) | 16 lines Merged revisions 138023 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r138023 | tilghman | 2008-08-15 09:51:12 -0500 (Fri, 15 Aug 2008)
- | 8 lines 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 ........ ................
-
-2008-08-14 22:43 +0000 [r137988] Russell Bryant <russell@digium.com>
-
- * /, doc/tex/Makefile: Merged revisions 137987 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r137987 |
- russell | 2008-08-14 17:43:15 -0500 (Thu, 14 Aug 2008) | 2 lines
- Fix a bashism that causes an error when trying to build the pdf
- on ubuntu ........
-
-2008-08-14 18:48 +0000 [r137934] Sean Bright <sean.bright@gmail.com>
-
- * cdr/cdr_sqlite3_custom.c, /: Merged revisions 137933 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r137933 | seanbright | 2008-08-14 14:47:28 -0400 (Thu, 14 Aug
- 2008) | 8 lines Fix memory leak in cdr_sqlite3_custom. (closes
- issue #13304) Reported by: eliel Patches: sqlite.patch uploaded
- by eliel (license 64) (Slightly modified by me) ........
-
-2008-08-14 17:01 +0000 [r137849-137852] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 137848 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r137848 | tilghman | 2008-08-14 11:52:43 -0500
- (Thu, 14 Aug 2008) | 17 lines Merged revisions 137847 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r137847 | tilghman | 2008-08-14 11:47:30 -0500 (Thu, 14 Aug 2008)
- | 9 lines 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 Jason Parker <jparker@digium.com>
-
- * Asterisk 1.6.0-rc2 released.
-
-2008-08-14 15:37 +0000 [r137814] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 137812 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r137812 |
- qwell | 2008-08-14 10:32:16 -0500 (Thu, 14 Aug 2008) | 8 lines
- Make sure we set the socket port, so we don't try to use <ip
- address>:0. (closes issue #13255) Reported by: falves11 Patches:
- 13255-socketport.diff uploaded by qwell (license 4) Tested by:
- falves11 ........
-
-2008-08-14 15:20 +0000 [r137783] Russell Bryant <russell@digium.com>
-
- * /, configs/sip.conf.sample: Merged revisions 137732 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r137732 | russell | 2008-08-14 09:15:50 -0500
- (Thu, 14 Aug 2008) | 12 lines Merged revisions 137731 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r137731 | russell | 2008-08-14 09:05:23 -0500 (Thu, 14 Aug 2008)
- | 4 lines 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 15:05 +0000 [r137781] Sean Bright <sean.bright@gmail.com>
-
- * cdr/cdr_tds.c, /: Merged revisions 137780 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r137780 |
- seanbright | 2008-08-14 11:03:03 -0400 (Thu, 14 Aug 2008) | 8
- lines If we detect that we are no longer connected, try to
- reconnect a few times before giving up. This relies on the
- timeout settings in the freetds.conf file and, unfortunately, on
- a recent version of FreeTDS (0.82 or newer). I either need to
- change the current execs to be non-blocking (which I do not want
- to do) or we have to force people to run with the latest and
- greatest of FreeTDS. I'm on the fence... ........
-
-2008-08-14 02:04 +0000 [r137681] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, Zaptel-to-DAHDI.txt: Merged revisions 137680 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r137680 | kpfleming | 2008-08-13 21:03:47 -0500 (Wed, 13 Aug
- 2008) | 9 lines Merged revisions 137679 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r137679 | kpfleming | 2008-08-13 21:03:04 -0500 (Wed, 13 Aug
- 2008) | 1 line forgot one module name that changed ........
- ................
-
-2008-08-13 Kevin P. Fleming <kpfleming@digium.com>
-
- * Asterisk 1.6.0-rc1 released.
-
-2008-08-13 23:00 +0000 [r137631-137641] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, build_tools/prep_tarball: Merged revisions 137640 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r137640 | kpfleming | 2008-08-13 18:00:37 -0500 (Wed, 13 Aug
- 2008) | 1 line make this script actually work ........
-
- * /, Zaptel-to-DAHDI.txt (added), UPGRADE.txt: Merged revisions
- 137627 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r137627 | kpfleming | 2008-08-13 17:33:32 -0500 (Wed, 13 Aug
- 2008) | 9 lines Merged revisions 137530 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r137530 | kpfleming | 2008-08-13 15:46:19 -0500 (Wed, 13 Aug
- 2008) | 1 line add document describing what users will need to be
- aware of when upgrading to this version and using DAHDI ........
- ................
-
-2008-08-13 21:09 +0000 [r137497-137533] Jason Parker <jparker@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 137532 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r137532 |
- qwell | 2008-08-13 16:08:58 -0500 (Wed, 13 Aug 2008) | 8 lines
- Correctly end locally ended calls. (closes issue #12170) Reported
- by: pj Patches: 20080702__issue12170_clear_pendinginvite.diff
- uploaded by bbryant (license 36) Tested by: bbryant, pabelanger
- ........
-
- * /, apps/app_fax.c: Merged revisions 137496 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r137496 |
- qwell | 2008-08-13 15:05:50 -0500 (Wed, 13 Aug 2008) | 6 lines
- Add FAXMODE variable with what fax transport was used. (closes
- issue #13252) Patches: v1-13252.patch uploaded by dimas (license
- 88) ........
-
-2008-08-13 14:47 +0000 [r137350-137407] Sean Bright <sean.bright@gmail.com>
-
- * /, doc/tex/cdrdriver.tex: Merged revisions 137406 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r137406 | seanbright | 2008-08-13 10:41:49 -0400
- (Wed, 13 Aug 2008) | 9 lines Merged revisions 137405 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r137405 | seanbright | 2008-08-13 10:33:49 -0400 (Wed,
- 13 Aug 2008) | 1 line Update docs to reflect the change to
- cdr_tds ........ ................
-
- * cdr/cdr_tds.c, /: Merged revisions 137403 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r137403 |
- seanbright | 2008-08-13 10:22:47 -0400 (Wed, 13 Aug 2008) | 1
- line Use the ast_vasprintf macro instead of vasprintf directly.
- ........
-
-2008-08-12 19:48 +0000 [r137300-137302] Russell Bryant <russell@digium.com>
-
- * doc/tex/asterisk.tex, /: Merged revisions 137301 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r137301 | russell | 2008-08-12 14:48:38 -0500 (Tue, 12 Aug 2008)
- | 2 lines Grammar hax from Qwell ........
-
- * doc/tex/asterisk.tex, /: Merged revisions 137299 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r137299 | russell | 2008-08-12 14:40:35 -0500 (Tue, 12 Aug 2008)
- | 3 lines Note that developer documentation belongs in doxygen,
- and not integrated with the user manual stuff in doc/tex/.
- ........
-
-2008-08-11 16:15 +0000 [r137240] Russell Bryant <russell@digium.com>
-
- * Makefile, /: Merged revisions 137239 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r137239 |
- russell | 2008-08-11 11:14:29 -0500 (Mon, 11 Aug 2008) | 2 lines
- Make PRINT_DIR work as advertised. ........
-
-2008-08-11 14:31 +0000 [r137217] Sean Bright <sean.bright@gmail.com>
-
- * cdr/cdr_tds.c, /, UPGRADE.txt: Merged revisions 137203 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r137203 | seanbright | 2008-08-11 10:25:15 -0400 (Mon,
- 11 Aug 2008) | 7 lines Log the userfield CDR variable like the
- other CDR backends, assuming the column is actually there. If
- it's not, we still log everything else as before. (closes issue
- #13281) Reported by: falves11 ........
-
-2008-08-11 00:27 +0000 [r137160] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_odbc.c, /: Merged revisions 137150 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r137150 | tilghman | 2008-08-10 19:25:28 -0500 (Sun, 10 Aug 2008)
- | 13 lines Merged revisions 137138 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r137138 | tilghman | 2008-08-10 19:20:38 -0500 (Sun, 10 Aug 2008)
- | 5 lines Deallocate database connection handle on disconnect, as
- we allocate another one on connect. (closes issue #13271)
- Reported by: dveiga ........ ................
-
-2008-08-09 15:27 +0000 [r136948] Tilghman Lesher <tlesher@digium.com>
-
- * /, include/asterisk/compat.h, include/asterisk/astobj2.h: Merged
- revisions 136947 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r136947 | tilghman | 2008-08-09 10:26:27 -0500 (Sat, 09 Aug 2008)
- | 18 lines Merged revisions 136946 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r136946 | tilghman | 2008-08-09 10:25:36 -0500
- (Sat, 09 Aug 2008) | 10 lines 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 ........ ................
- ................
-
-2008-08-09 01:16 +0000 [r136860] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_agi.c: Merged revisions 136859 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r136859 |
- tilghman | 2008-08-08 20:15:38 -0500 (Fri, 08 Aug 2008) | 4 lines
- Update documentation as to the behavior of AGI in 1.6.0 and
- higher. Also, add an OOB message that answers the question of, if
- AGI no longer shuts down the connection on hangup, how will
- FastAGI know when to stop processing the call? ........
-
-2008-08-08 15:33 +0000 [r136785] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 136784 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r136784 | mmichelson | 2008-08-08 10:31:31 -0500 (Fri, 08 Aug
- 2008) | 3 lines Fix compilation for ODBC voicemail ........
-
-2008-08-08 06:45 +0000 [r136778] 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-test19,
- pbx/ael/ael-test/ref.ael-vtest13, res/ael/pval.c, /,
- pbx/ael/ael-test/ref.ael-ntest10, include/asterisk/ael_structs.h,
- utils/ael_main.c: Merged revisions 136746 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r136746 | murf | 2008-08-07 18:48:35 -0600 (Thu, 07 Aug 2008) |
- 40 lines Merged revisions 136726 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r136726 | murf | 2008-08-07 18:15:34 -0600 (Thu, 07 Aug 2008) |
- 32 lines (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. ........ ................
-
-2008-08-08 02:36 +0000 [r136753] Tilghman Lesher <tlesher@digium.com>
-
- * /: Merged revisions 136751 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r136751 |
- tilghman | 2008-08-07 21:34:17 -0500 (Thu, 07 Aug 2008) | 2 lines
- Removing bad properties ........
-
-2008-08-07 23:42 +0000 [r136719-136724] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c: This is weird. Either SVN or vim tabbed a
- bunch of functions over one level during a merge.
-
- * apps/app_voicemail.c, /: Merged revisions 136722 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r136722 | mmichelson | 2008-08-07 18:39:50 -0500 (Thu, 07 Aug
- 2008) | 3 lines Remove one last batch of debug messages ........
-
- * apps/app_voicemail.c, /: Merged revisions 136715 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r136715 | mmichelson | 2008-08-07 17:25:50 -0500 (Thu, 07 Aug
- 2008) | 18 lines Merging the imap_consistency_trunk branch to
- trunk. For an explanation of what "imap_consistency" is, please
- see svn revision 134223 to the 1.4 branch. Coincidentally, this
- also fixes a recent bug report regarding the inability to save
- messages to the new folder when using IMAP storage since they
- will would be flagged as "seen" and not be recognized as new
- messages. (closes issue #13234) Reported by: jaroth ........
-
-2008-08-07 20:41 +0000 [r136672-136674] Shaun Ruffell <sruffell@digium.com>
-
- * codecs/codec_dahdi.c: Removing code that was commented out.
-
- * codecs/codec_dahdi.c: Updated codec_dahdi to use the transcoder
- interface in the DAHDI. (Issue: DAHDI-42)
-
-2008-08-07 20:26 +0000 [r136632-136663] Mark Michelson <mmichelson@digium.com>
-
- * /, main/features.c: Merged revisions 136660 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r136660 |
- mmichelson | 2008-08-07 15:25:43 -0500 (Thu, 07 Aug 2008) | 4
- lines Bump a LOG_NOTICE message to LOG_DEBUG since it appears
- once for every bridged call ........
-
- * main/pbx.c, /: Merged revisions 136635 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r136635 |
- mmichelson | 2008-08-07 14:58:32 -0500 (Thu, 07 Aug 2008) | 5
- lines Don't allow Answer() to accept a negative argument.
- Negative argument means an infinite delay and we don't want that.
- ........
-
- * main/channel.c, /: Merged revisions 136633 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r136633 |
- mmichelson | 2008-08-07 14:54:27 -0500 (Thu, 07 Aug 2008) | 7
- lines Fix a calculation error I had made in the poll. The poll
- would reset to 500 ms every time a non-voice frame was received.
- The total time we poll should be 500 ms, so now we save the
- amount of time left after the poll returned and use that as our
- argument for the next call to poll ........
-
- * main/channel.c, /: Merged revisions 136631 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r136631 |
- mmichelson | 2008-08-07 14:36:46 -0500 (Thu, 07 Aug 2008) | 13
- lines Scrap the 500 ms delay when Asterisk auto-answers a
- channel. Instead, poll the channel until receiving a voice frame.
- The cap on this poll is 500 ms. The optional delay is still
- allowable in the Answer() application, but the delay has been
- moved back to its original position, after the call to the
- channel's answer callback. The poll for the voice frame will not
- happen if a delay is specified when calling Answer(). (closes
- issue #12708) Reported by: kactus ........
-
-2008-08-07 19:19 +0000 [r136598] Richard Mudgett <rmudgett@digium.com>
-
- * channels/misdn_config.c, channels/chan_misdn.c, /,
- configs/misdn.conf.sample: Merged revisions 136594 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r136594 | rmudgett | 2008-08-07 14:01:03 -0500
- (Thu, 07 Aug 2008) | 13 lines Merged revisions 136241 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r136241 | rmudgett | 2008-08-06 16:18:53 -0500 (Wed, 06 Aug 2008)
- | 5 lines * The allowed_bearers setting in misdn.conf misspelled
- one of its options: digital_restricted. * Fixed some other
- spelling errors and typos. ........ ................
-
-2008-08-07 17:44 +0000 [r136506-136543] Kevin P. Fleming <kpfleming@digium.com>
-
- * include/asterisk/doxyref.h, /: Merged revisions 136542 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r136542 | kpfleming | 2008-08-07 12:44:20 -0500
- (Thu, 07 Aug 2008) | 6 lines Merged revisions 136541 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- ........ ................
-
-2008-08-07 16:57 +0000 [r136490] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 136489 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r136489 | tilghman | 2008-08-07 11:55:57 -0500 (Thu, 07 Aug 2008)
- | 15 lines Merged revisions 136488 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r136488 | tilghman | 2008-08-07 11:50:47 -0500 (Thu, 07 Aug 2008)
- | 7 lines 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
- ........ ................
-
-2008-08-06 20:16 +0000 [r136113-136192] Tilghman Lesher <tlesher@digium.com>
-
- * /, contrib/init.d/rc.redhat.asterisk: Merged revisions 136191 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r136191 | tilghman | 2008-08-06 15:15:34 -0500
- (Wed, 06 Aug 2008) | 12 lines Merged revisions 136190 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r136190 | tilghman | 2008-08-06 15:14:54 -0500 (Wed, 06 Aug 2008)
- | 4 lines -C option takes a filename, not a directory path.
- (closes issue #13007) Reported by: klaus3000 ........
- ................
-
- * /, funcs/func_dialgroup.c: Merged revisions 136112 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r136112 | tilghman | 2008-08-06 11:58:42 -0500 (Wed, 06 Aug 2008)
- | 7 lines Persist DIALGROUP() values in astdb (closes issue
- #13138) Reported by: Corydon76 Patches:
- 20080725__bug13138.diff.txt uploaded by Corydon76 (license 14)
- Tested by: pj ........
-
-2008-08-06 16:00 +0000 [r136064] Mark Michelson <mmichelson@digium.com>
-
- * main/rtp.c, /, channels/chan_skinny.c: Merged revisions 136063
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r136063 | mmichelson | 2008-08-06 10:59:29 -0500
- (Wed, 06 Aug 2008) | 24 lines Merged revisions 136062 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r136062 | mmichelson | 2008-08-06 10:58:40 -0500 (Wed, 06 Aug
- 2008) | 16 lines 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 13:59 +0000 [r136006] Olle Johansson <oej@edvina.net>
-
- * /, res/res_jabber.c: Merged revisions 136005 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r136005 |
- oej | 2008-08-06 15:34:08 +0200 (Ons, 06 Aug 2008) | 6 lines -
- Formatting - Changing debug messages from VERBOSE to DEBUG
- channel - Adding a few todo's - Adding a few more "XMPP"'s to
- compliment Jabber... ........
-
-2008-08-06 03:56 +0000 [r135901-135951] Tilghman Lesher <tlesher@digium.com>
-
- * main/channel.c, /: Merged revisions 135950 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r135950 | tilghman | 2008-08-05 22:55:49 -0500 (Tue, 05 Aug 2008)
- | 12 lines Merged revisions 135949 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r135949 | tilghman | 2008-08-05 22:53:36 -0500 (Tue, 05 Aug 2008)
- | 4 lines Fix a longstanding bug in channel walking logic, and
- fix the explanation to make sense. (Closes issue #13124) ........
- ................
-
- * /, main/translate.c: Merged revisions 135938 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r135938 | tilghman | 2008-08-05 22:29:42 -0500 (Tue, 05 Aug 2008)
- | 12 lines Merged revisions 135915 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r135915 | tilghman | 2008-08-05 22:24:56 -0500 (Tue, 05 Aug 2008)
- | 4 lines 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, /:
- Merged revisions 135900 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r135900 | tilghman | 2008-08-05 22:04:01 -0500 (Tue, 05 Aug 2008)
- | 12 lines Merged revisions 135899 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r135899 | tilghman | 2008-08-05 22:02:59 -0500 (Tue, 05 Aug 2008)
- | 4 lines 1) Bugfix for debugging code 2) Reduce compiler
- warnings for another section of debugging code (Closes issue
- #13237) ........ ................
-
-2008-08-06 00:31 +0000 [r135852] Mark Michelson <mmichelson@digium.com>
-
- * include/asterisk/abstract_jb.h, main/channel.c, /,
- main/abstract_jb.c, main/fixedjitterbuf.h: Merged revisions
- 135851 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r135851 | mmichelson | 2008-08-05 19:30:53 -0500 (Tue, 05 Aug
- 2008) | 48 lines Merged revisions 135841,135847,135850 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r135841 | mmichelson | 2008-08-05 19:25:10 -0500 (Tue, 05 Aug
- 2008) | 27 lines 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
- ........ r135847 | mmichelson | 2008-08-05 19:27:54 -0500 (Tue,
- 05 Aug 2008) | 4 lines Revert inadvertent changes to app_skel
- that occurred when I was testing for a memory leak ........
- r135850 | mmichelson | 2008-08-05 19:29:54 -0500 (Tue, 05 Aug
- 2008) | 3 lines Remove properties that should not be here
- ........ ................
-
-2008-08-05 23:52 +0000 [r135822] Steve Murphy <murf@digium.com>
-
- * apps/app_dial.c, main/cdr.c, main/channel.c, /, main/features.c,
- include/asterisk/cdr.h: Merged revisions 135821 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r135821 | murf | 2008-08-05 17:45:32 -0600 (Tue, 05 Aug 2008) |
- 42 lines Merged revisions 135799 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r135799 | murf | 2008-08-05 17:13:20 -0600 (Tue, 05 Aug 2008) |
- 34 lines (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 21:38 +0000 [r135749] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 135748 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r135748 | tilghman | 2008-08-05 16:37:35 -0500
- (Tue, 05 Aug 2008) | 17 lines Merged revisions 135747 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r135747 | tilghman | 2008-08-05 16:34:46 -0500 (Tue, 05 Aug 2008)
- | 9 lines 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 13:27 +0000 [r135599] Sean Bright <sean.bright@gmail.com>
-
- * main/cli.c, /: Merged revisions 135598 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r135598 | seanbright | 2008-08-05 09:26:34 -0400 (Tue, 05 Aug
- 2008) | 9 lines Merged revisions 135597 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r135597 | seanbright | 2008-08-05 09:25:00 -0400 (Tue, 05 Aug
- 2008) | 1 line Use PATH_MAX for filenames ........
- ................
-
-2008-08-04 20:15 +0000 [r135538] Russell Bryant <russell@digium.com>
-
- * configs/chan_dahdi.conf.sample, /: Merged revisions 135537 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r135537 | russell | 2008-08-04 15:15:27 -0500
- (Mon, 04 Aug 2008) | 10 lines Merged revisions 135536 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r135536 | russell | 2008-08-04 15:15:03 -0500 (Mon, 04 Aug 2008)
- | 2 lines fix a config sample typo ........ ................
-
-2008-08-04 17:12 +0000 [r135478-135486] Tilghman Lesher <tlesher@digium.com>
-
- * contrib/init.d/rc.mandriva.asterisk (added), Makefile,
- contrib/init.d/rc.mandrake.asterisk (removed), /,
- contrib/init.d/rc.mandriva.zaptel (added),
- contrib/init.d/rc.mandrake.zaptel (removed): Merged revisions
- 135485 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r135485 |
- tilghman | 2008-08-04 12:12:15 -0500 (Mon, 04 Aug 2008) | 3 lines
- Rename Mandrake scripts to Mandriva (Closes issue #13221)
- ........
-
- * contrib/init.d/rc.mandrake.asterisk, /: Merged revisions 135483
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r135483 | tilghman | 2008-08-04 12:08:42 -0500
- (Mon, 04 Aug 2008) | 11 lines Merged revisions 135482 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r135482 | tilghman | 2008-08-04 12:07:52 -0500 (Mon, 04 Aug 2008)
- | 2 lines Define ASTSBINDIR for script (Closes issue #13221)
- ........ ................
-
- * apps/app_voicemail.c, /: Merged revisions 135480 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r135480 | tilghman | 2008-08-04 11:58:29 -0500
- (Mon, 04 Aug 2008) | 14 lines Merged revisions 135479 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r135479 | tilghman | 2008-08-04 11:56:19 -0500 (Mon, 04 Aug 2008)
- | 6 lines Memory leak on unload (closes issue #13231) Reported
- by: eliel Patches: app_voicemail.leak.patch uploaded by eliel
- (license 64) ........ ................
-
-2008-08-04 16:28 +0000 [r135440-135475] Russell Bryant <russell@digium.com>
-
- * configs/chan_dahdi.conf.sample, /: Merged revisions 135474 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r135474 | russell | 2008-08-04 11:28:07 -0500
- (Mon, 04 Aug 2008) | 10 lines Merged revisions 135473 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r135473 | russell | 2008-08-04 11:26:17 -0500 (Mon, 04 Aug 2008)
- | 2 lines Add a minor clarification to the documentation of
- mohinterpret and mohsuggest ........ ................
-
- * /, channels/chan_console.c: Merged revisions 135439 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r135439 | russell | 2008-08-04 10:02:12 -0500 (Mon, 04 Aug 2008)
- | 4 lines Be explicit that we don't want a result from this
- callback. The callback would never indicate a match, so nothing
- would have been returned anyway, but it was still a poor example
- of proper usage. ........
-
-2008-08-02 05:15 +0000 [r135266] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, /: Merged revisions 135265 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r135265 |
- murf | 2008-08-01 22:51:29 -0600 (Fri, 01 Aug 2008) | 31 lines
- (closes issue #13202) Reported by: falves11 Tested by: murf
- falves11 == The changes I introduce here seem to clear up the
- problem for me. However, if they do not for you, please reopen
- this bug, and we'll keep digging. The root of this problem seems
- to be a subtle memory corruption introduced when creating an
- extension with an empty extension name. While valgrind cannot
- detect it outside of DEBUG_MALLOC mode, when compiled with
- DEBUG_MALLOC, this is certain death. The code in main/features.c
- is a puzzle to me. On the initial module load, the code is
- attempting to add the parking extension before the features.conf
- file has even been opened! I just wrapped the offending call with
- an if() that will not try to add the extension if the extension
- name is empty. THis seems to solve the corruption, and let the
- "memory show allocations" work as one would expect. But, really,
- adding an extension with an empty name is a seriously bad thing
- to allow, as it will mess up all the pattern matching algorithms,
- etc. So, I added a statement to the add_extension2 code to return
- a -1 if this is attempted. in 1.6.0, the changes to only
- main/pbx.c were applicable, as apparently the code added to
- main/features by jpeeler were not included in 1.6.0. ........
-
-2008-08-01 19:30 +0000 [r135198] Sean Bright <sean.bright@gmail.com>
-
- * channels/chan_mgcp.c, /: Merged revisions 135197 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r135197 | seanbright | 2008-08-01 15:29:26 -0400 (Fri, 01 Aug
- 2008) | 6 lines Remove some code that used to do something but
- does not anymore, mainly to get rid of a shadow warning (but this
- seemed legitimate enough to fix here instead of in my branch).
- Thanks to putnopvut for taking a look as well. ........
-
-2008-08-01 17:10 +0000 [r135127-135129] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 135128 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r135128 |
- tilghman | 2008-08-01 12:09:50 -0500 (Fri, 01 Aug 2008) | 2 lines
- Picky, picky, buildbot ........
-
- * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
- 135126 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r135126 |
- tilghman | 2008-08-01 11:39:51 -0500 (Fri, 01 Aug 2008) | 9 lines
- SIP should use the transport type set in the Moved Temporarily
- for the next invite. (closes issue #11843) Reported by:
- pestermann Patches:
- 20080723__issue11843_302_ignores_transport_16branch.diff uploaded
- by bbryant (license 36)
- 20080723__issue11843_302_ignores_transport_trunk.diff uploaded by
- bbryant (license 36) Tested by: pabelanger ........
-
-2008-08-01 14:43 +0000 [r135070] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged
- revisions 135067-135068 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r135067 |
- mmichelson | 2008-08-01 09:29:48 -0500 (Fri, 01 Aug 2008) | 13
- lines IMAP storage functioned under the assumption that folders
- such as "Work" and "Family" would be subfolders of the INBOX.
- This is an invalid assumption to make, but it could be desirable
- to set up folders in this manner, so a new option for
- voicemail.conf, "imapparentfolder" has been added to allow for
- this. (closes issue #13142) Reported by: jaroth Patches:
- parentfolder.patch uploaded by jaroth (license 50) ........
- r135068 | mmichelson | 2008-08-01 09:42:24 -0500 (Fri, 01 Aug
- 2008) | 3 lines IMAP-specific items must go in IMAP_STORAGE
- defines... ........
-
-2008-08-01 12:18 +0000 [r135057-135062] Michiel van Baak <michiel@vanbaak.info>
-
- * /, apps/app_ices.c: Merged revisions 135059 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r135059 | mvanbaak | 2008-08-01 13:47:34 +0200 (Fri, 01 Aug 2008)
- | 10 lines Merged revisions 135058 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r135058 | mvanbaak | 2008-08-01 13:43:46 +0200 (Fri, 01 Aug 2008)
- | 2 lines make app_ices compile on OpenBSD. ........
- ................
-
- * /, channels/chan_skinny.c: Merged revisions 135056 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r135056 | mvanbaak | 2008-08-01 13:00:13 +0200
- (Fri, 01 Aug 2008) | 16 lines Merged revisions 135055 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r135055 | mvanbaak | 2008-08-01 12:55:27 +0200 (Fri, 01 Aug 2008)
- | 8 lines 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 22:34 +0000 [r135034] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, main/http.c: Merged revisions 135016 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r135016 | kpfleming | 2008-07-31 17:28:42 -0500 (Thu, 31 Jul
- 2008) | 11 lines Merged revisions 134983 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r134983 | kpfleming | 2008-07-31 17:18:11 -0500 (Thu, 31 Jul
- 2008) | 3 lines accomodate users who seem to lack a sense of
- humor :-) ........ ................
-
-2008-07-31 21:58 +0000 [r134926-134981] Tilghman Lesher <tlesher@digium.com>
-
- * sample.call, main/manager.c, pbx/pbx_spool.c, /: Merged revisions
- 134980 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r134980 | tilghman | 2008-07-31 16:55:42 -0500 (Thu, 31 Jul 2008)
- | 16 lines Blocked revisions 134976 via svnmerge ........ r134976
- | tilghman | 2008-07-31 16:53:19 -0500 (Thu, 31 Jul 2008) | 9
- lines 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 ........ ................
-
- * res/res_config_sqlite.c, /: Merged revisions 134977 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r134977 | tilghman | 2008-07-31 16:53:59 -0500 (Thu, 31 Jul 2008)
- | 2 lines Switch command order, to meet with current specs
- ........
-
-2008-07-31 19:54 +0000 [r134923] Steve Murphy <murf@digium.com>
-
- * /, main/features.c: Merged revisions 134922 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r134922 | murf | 2008-07-31 13:48:08 -0600 (Thu, 31 Jul 2008) |
- 63 lines Merged revisions 134883 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r134883 | murf | 2008-07-31 13:23:42 -0600 (Thu, 31 Jul 2008) |
- 51 lines (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 19:41 +0000 [r134918] Russell Bryant <russell@digium.com>
-
- * /, apps/app_ices.c: Merged revisions 134917 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r134917 | russell | 2008-07-31 14:39:50 -0500 (Thu, 31 Jul 2008)
- | 17 lines Merged revisions 134915 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r134915 | russell | 2008-07-31 14:37:26 -0500 (Thu, 31 Jul 2008)
- | 9 lines 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 16:53 +0000 [r134816] Russell Bryant <russell@digium.com>
-
- * channels/iax2-parser.c: Merged revisions 134815 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r134815 | russell | 2008-07-31 11:50:10 -0500 (Thu, 31 Jul 2008)
- | 15 lines Merged revisions 134814 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r134814 | russell | 2008-07-31 11:45:31 -0500 (Thu, 31 Jul 2008)
- | 7 lines 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 16:07 +0000 [r134760] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 134759 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r134759 | mmichelson | 2008-07-31 11:05:12 -0500 (Thu, 31 Jul
- 2008) | 24 lines Merged revisions 134758 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r134758 | mmichelson | 2008-07-31 10:56:18 -0500 (Thu, 31 Jul
- 2008) | 16 lines 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 22:41 +0000 [r134651-134707] Tilghman Lesher <tlesher@digium.com>
-
- * main/sched.c, /, include/asterisk/sched.h: Merged revisions
- 134703 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r134703 |
- tilghman | 2008-07-30 17:38:58 -0500 (Wed, 30 Jul 2008) | 2 lines
- Oops, wrong define ........
-
- * /, configure, configure.ac: Merged revisions 134650 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r134650 | tilghman | 2008-07-30 16:40:08 -0500
- (Wed, 30 Jul 2008) | 12 lines Merged revisions 134649 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r134649 | tilghman | 2008-07-30 16:38:50 -0500 (Wed, 30 Jul 2008)
- | 4 lines 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 21:06 +0000 [r134599] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 134598 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r134598 | mmichelson | 2008-07-30 16:05:37 -0500 (Wed, 30 Jul
- 2008) | 15 lines Merged revisions 134556 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r134556 |
- mmichelson | 2008-07-30 15:24:40 -0500 (Wed, 30 Jul 2008) | 7
- lines Fix the parsing of the "reason" parameter in the Diversion:
- header. (closes issue #13195) Reported by: woodsfsg ........
- ................
-
-2008-07-30 20:39 +0000 [r134597] Russell Bryant <russell@digium.com>
-
- * /, pbx/pbx_dundi.c: Merged revisions 134596 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r134596 | russell | 2008-07-30 15:38:35 -0500 (Wed, 30 Jul 2008)
- | 14 lines Merged revisions 134595 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r134595 | russell | 2008-07-30 15:37:17 -0500 (Wed, 30 Jul 2008)
- | 6 lines 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 ........ ................
-
-2008-07-30 20:25 +0000 [r134561] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 134556 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r134556 |
- mmichelson | 2008-07-30 15:24:40 -0500 (Wed, 30 Jul 2008) | 7
- lines Fix the parsing of the "reason" parameter in the Diversion:
- header. (closes issue #13195) Reported by: woodsfsg ........
-
-2008-07-30 19:56 +0000 [r134542] Russell Bryant <russell@digium.com>
-
- * funcs/func_curl.c, /: Merged revisions 134541 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r134541 | russell | 2008-07-30 14:55:31 -0500 (Wed, 30 Jul 2008)
- | 12 lines Merged revisions 134540 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r134540 | russell | 2008-07-30 14:52:53 -0500 (Wed, 30 Jul 2008)
- | 4 lines 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 19:49 +0000 [r134482-134539] Tilghman Lesher <tlesher@digium.com>
-
- * /, configure, configure.ac: Merged revisions 134538 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r134538 | tilghman | 2008-07-30 14:48:37 -0500
- (Wed, 30 Jul 2008) | 12 lines Merged revisions 134536 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r134536 | tilghman | 2008-07-30 14:47:16 -0500 (Wed, 30 Jul 2008)
- | 4 lines Only override sysconfdir and mandir when prefix=/usr
- (closes issue #13093) Reported by: pabelanger ........
- ................
-
- * /, apps/app_queue.c: Merged revisions 134483 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r134483 |
- tilghman | 2008-07-30 14:17:38 -0500 (Wed, 30 Jul 2008) | 4 lines
- Let "roundrobin" also reference rrmemory, for the 1.6 release (as
- described in UPGRADE-1.4.txt) (Closes issue #13181) ........
-
- * /, res/res_agi.c: Merged revisions 134481 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r134481 | tilghman | 2008-07-30 14:05:35 -0500 (Wed, 30 Jul 2008)
- | 13 lines Merged revisions 134480 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r134480 | tilghman | 2008-07-30 14:03:44 -0500 (Wed, 30 Jul 2008)
- | 5 lines 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 18:33 +0000 [r134477] Mark Michelson <mmichelson@digium.com>
-
- * /, main/app.c: Merged revisions 134476 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r134476 | mmichelson | 2008-07-30 13:33:12 -0500 (Wed, 30 Jul
- 2008) | 12 lines Merged revisions 134475 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r134475 | mmichelson | 2008-07-30 13:31:47 -0500 (Wed, 30 Jul
- 2008) | 4 lines Fix a spot where a function could return without
- bringing a channel out of autoservice. ........ ................
-
-2008-07-30 15:34 +0000 [r134356] Kevin P. Fleming <kpfleming@digium.com>
-
- * Makefile, /: Merged revisions 134355 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r134355 | kpfleming | 2008-07-30 10:32:14 -0500 (Wed, 30 Jul
- 2008) | 10 lines Merged revisions 134352 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r134352 | kpfleming | 2008-07-30 10:29:17 -0500 (Wed, 30 Jul
- 2008) | 2 lines use the proper method for building version.h
- ........ ................
-
-2008-07-29 22:29 +0000 [r134283] Kevin P. Fleming <kpfleming@digium.com>
-
- * apps/app_rpt.c, apps/app_dahdibarge.c, channels/chan_dahdi.c, /,
- apps/app_meetme.c, apps/app_dahdiscan.c, apps/app_dahdiras.c:
- Merged revisions 134260 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r134260 |
- kpfleming | 2008-07-29 17:22:13 -0500 (Tue, 29 Jul 2008) | 2
- lines build against the now-typedef-free dahdi/user.h, and remove
- some #ifdefs for features that will always be present in DAHDI
- ........
-
-2008-07-28 22:16 +0000 [r134164] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 134163 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r134163 | tilghman | 2008-07-28 17:07:12 -0500
- (Mon, 28 Jul 2008) | 15 lines Merged revisions 134161 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r134161 | tilghman | 2008-07-28 16:50:50 -0500 (Mon, 28 Jul 2008)
- | 7 lines 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 ........
- ................
-
-2008-07-28 19:55 +0000 [r134126] Mark Michelson <mmichelson@digium.com>
-
- * /, configure, main/Makefile, configure.ac, CHANGES: Merged
- revisions 134125 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r134125 |
- mmichelson | 2008-07-28 14:53:56 -0500 (Mon, 28 Jul 2008) | 27
- lines This commit compensates for buggy poll(2) implementations.
- Asterisk has, for a long time, had its own implementation of
- poll(2) which just used the input arguments to call select(2). In
- 1.4, this internal implementation was used for Darwin systems.
- This was removed in Asterisk trunk at some point, but it seems as
- though this was not the right move to make. On Mac OS X, it
- appears as though the poll used to gather CLI input does not
- respond properly when connecting via a remote Asterisk console.
- Reverting to the use of Asterisk's poll fixed the issue. Also,
- there is now an option for the configure script,
- --enable-internal-poll, which will allow for anyone to use
- Asterisk's internal poll implementation in case they suspect that
- their system's poll implementation is buggy. closes issue #11928)
- Reported by: adriavidal Patches: 1.6.0-configurev2.patch uploaded
- by putnopvut (license 60) ........
-
-2008-07-28 16:49 +0000 [r134087] Kevin P. Fleming <kpfleming@digium.com>
-
- * apps/app_parkandannounce.c, main/loader.c, sample.call,
- contrib/scripts/autosupport, build_tools/cflags.xml,
- main/channel.c, apps/app_dahdibarge.c, channels/chan_dahdi.c,
- configs/chan_dahdi.conf.sample, doc/ss7.txt, /, main/features.c,
- doc/osp.txt, main/file.c, pbx/pbx_config.c: Merged revisions
- 134086 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r134086 |
- kpfleming | 2008-07-28 11:42:00 -0500 (Mon, 28 Jul 2008) | 3
- lines remove remaining Zaptel references in various places
- ........
-
-2008-07-28 16:13 +0000 [r134052] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_chanspy.c, include/asterisk/options.h, main/asterisk.c,
- /, apps/app_meetme.c, apps/app_dahdiscan.c: Merged revisions
- 134050 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r134050 |
- mmichelson | 2008-07-28 11:00:19 -0500 (Mon, 28 Jul 2008) | 3
- lines merging the zap_and_dahdi_trunk branch up to trunk ........
-
-2008-07-26 15:34 +0000 [r133942-133982] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c, include/asterisk/doxyref.h, /: Include the
- licensing page in 1.6.0 as well. Now, this page exists in 1.4,
- trunk, and 1.6.0.
-
- * /: unblock 133575
-
- * /, main/devicestate.c: Merged revisions 133945-133946 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r133945 | russell | 2008-07-26 10:15:14 -0500 (Sat, 26
- Jul 2008) | 6 lines ast_device_state() gets called in two
- different ways. The first way is when called from elsewhere in
- Asterisk to find the current state of a device. In that case, we
- want to use the cached value if it exists. The other way is when
- processing a device state change. In that case, we do not want to
- check the cache because returning the last known state is counter
- productive. ........ r133946 | russell | 2008-07-26 10:16:20
- -0500 (Sat, 26 Jul 2008) | 1 line actually use the cache_cache
- argument ........
-
-2008-07-25 22:09 +0000 [r133863-133905] Tilghman Lesher <tlesher@digium.com>
-
- * contrib/scripts/asterisk.ldif, /: Merged revisions 133902 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r133902 | tilghman | 2008-07-25 16:59:39 -0500 (Fri, 25
- Jul 2008) | 6 lines Update version (closes issue #13163) Reported
- by: suretec Patches: asterisk.ldif uploaded by suretec (license
- 70) ........
-
-2008-07-25 19:37 +0000 [r133804-133806] Brandon Kruse <bkruse@digium.com>
-
- * /: Blocking revert of code changes in r133770
-
- * main/http.c: Include the http_decode function from trunk to
- replace the + with a space.
-
-2008-07-25 17:33 +0000 [r133694] Brandon Kruse <bkruse@digium.com>
-
- * /: Blocking a fix from trunk for the function http_decode. 1.6.0
- does not have this function.
-
-2008-07-25 17:26 +0000 [r133671] Tilghman Lesher <tlesher@digium.com>
-
- * main/channel.c, /, channels/chan_agent.c, main/devicestate.c:
- Merged revisions 133665 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r133665 | tilghman | 2008-07-25 12:24:43 -0500 (Fri, 25 Jul 2008)
- | 16 lines Merged revisions 133649 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r133649 | tilghman | 2008-07-25 12:19:39 -0500 (Fri, 25 Jul 2008)
- | 8 lines 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 15:01 +0000 [r133576-133580] Russell Bryant <russell@digium.com>
-
- * /, LICENSE: Merged revisions 133579 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r133579 | russell | 2008-07-25 10:00:49 -0500 (Fri, 25 Jul 2008)
- | 18 lines Merged revisions 133578 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r133578 | russell | 2008-07-25 10:00:31 -0500
- (Fri, 25 Jul 2008) | 10 lines 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 14:41 +0000 [r133571-133574] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 133573 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r133573 | mmichelson | 2008-07-25 09:40:52 -0500 (Fri, 25 Jul
- 2008) | 15 lines Merged revisions 133572 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r133572 | mmichelson | 2008-07-25 09:40:10 -0500 (Fri, 25 Jul
- 2008) | 7 lines 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. ........
- ................
-
-2008-07-25 13:25 +0000 [r133567-133569] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 133568 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r133568 |
- russell | 2008-07-25 08:01:59 -0500 (Fri, 25 Jul 2008) | 4 lines
- Minor coding guidelines tweaks ... - Use ast_strlen_zero in one
- place - check for successful string comparison the way most of
- Asterisk code does it ........
-
-2008-07-24 21:31 +0000 [r133524] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 133509 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r133509 | tilghman | 2008-07-24 16:27:06 -0500 (Thu, 24 Jul 2008)
- | 11 lines Merged revisions 133488 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r133488 | tilghman | 2008-07-24 16:17:55 -0500 (Thu, 24 Jul 2008)
- | 3 lines Fix rtautoclear and rtcachefriends (Closes issue
- #12707) ........ ................
-
-2008-07-24 20:41 +0000 [r133487] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 133486 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r133486 | russell | 2008-07-24 15:40:15 -0500 (Thu, 24 Jul 2008)
- | 3 lines I made this change from DEVICE_STATE to
- DEVICE_STATE_CHANGE, but I had it backwards, this is the right
- event to subscribe to ... ........
-
-2008-07-24 19:55 +0000 [r133449] Mark Michelson <mmichelson@digium.com>
-
- * /, main/logger.c: Merged revisions 133448 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r133448 |
- mmichelson | 2008-07-24 14:53:37 -0500 (Thu, 24 Jul 2008) | 12
- lines Print the correct PID in log messages. Prior to this
- commit, only the logger thread's PID would be printed. (closes
- issue #13150) Reported by: atis Patches: log_pid.diff uploaded by
- putnopvut (license 60) Tested by: eliel ........
-
-2008-07-24 05:21 +0000 [r133392-133405] Tilghman Lesher <tlesher@digium.com>
-
- * contrib/scripts/asterisk.logrotate, Makefile, /: Merged revisions
- 133400 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r133400 |
- tilghman | 2008-07-24 00:21:00 -0500 (Thu, 24 Jul 2008) | 3 lines
- Build the logrotate script according to paths (Closes issue
- #13147) ........
-
- * Makefile, /: Merged revisions 133391 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r133391 |
- tilghman | 2008-07-23 23:51:42 -0500 (Wed, 23 Jul 2008) | 3 lines
- Optionally install logrotate file (Closes issue #13148) ........
-
-2008-07-23 22:07 +0000 [r133300] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, /: Merged revisions 133299 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r133299 |
- murf | 2008-07-23 16:03:48 -0600 (Wed, 23 Jul 2008) | 27 lines
- (closes issue #13144) Reported by: murf Tested by: murf For: J.
- Geis The 'data' field in the ast_exten struct was being 'moved'
- from the current dialplan to the replacement dialplan. This was
- not good, as the current dialplan could have problems in the time
- between the change and when the new dialplan is swapped in. So, I
- modified the merge_and_delete code to strdup the 'data' field
- (the args to the app call), and then it's freed as normal. I
- improved a few messages; I added code to limit the number of
- calls to the context_merge_incls_swits_igps_other_registrars() to
- one per context. I don't think having it called multiple times
- per context was doing anything bad, but it was inefficient. I
- hope this fixes the problems Mr. Geiss was noting in
- asterisk-users, see
- http://lists.digium.com/pipermail/asterisk-users/2008-July/215634.html
- ........
-
-2008-07-23 21:50 +0000 [r133297] Jason Parker <jparker@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 133296 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r133296 | qwell | 2008-07-23 16:50:20 -0500
- (Wed, 23 Jul 2008) | 9 lines Merged revisions 133295 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r133295 | qwell | 2008-07-23 16:49:03 -0500 (Wed, 23 Jul
- 2008) | 1 line inbandrelease is gone - it's now inbanddisconnect
- ........ ................
-
-2008-07-23 20:39 +0000 [r133218] Brett Bryant <bbryant@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 133197 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r133197 |
- bbryant | 2008-07-23 15:33:22 -0500 (Wed, 23 Jul 2008) | 2 lines
- Fix issue where tcp in sip is enabled by default, despite what it
- says in the config sample file. Also fix "sip show settings" for
- tcp connections. ........
-
-2008-07-23 19:50 +0000 [r133042-133172] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_chanspy.c, include/asterisk/options.h, main/asterisk.c,
- /: Merged revisions 133171 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r133171 | mmichelson | 2008-07-23 14:48:03 -0500 (Wed, 23 Jul
- 2008) | 20 lines Merged revisions 133169 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r133169 | mmichelson | 2008-07-23 14:39:47 -0500 (Wed, 23 Jul
- 2008) | 12 lines 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, /: Merged revisions 133106 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r133106 | mmichelson | 2008-07-23 14:07:56 -0500 (Wed, 23 Jul
- 2008) | 13 lines Merged revisions 133104 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r133104 | mmichelson | 2008-07-23 14:06:16 -0500 (Wed, 23 Jul
- 2008) | 5 lines Zap/pseudo is ten characters, but DAHDI/pseudo is
- twelve. The strncmp call in next_channel should account for this.
- ........ ................
-
- * apps/app_chanspy.c, /: Merged revisions 133102 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r133102 | mmichelson | 2008-07-23 13:58:37 -0500 (Wed, 23 Jul
- 2008) | 14 lines Merged revisions 133101 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r133101 | mmichelson | 2008-07-23 13:57:17 -0500 (Wed, 23 Jul
- 2008) | 6 lines 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, /: Merged revisions 133041 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r133041 | mmichelson | 2008-07-23 12:54:03 -0500
- (Wed, 23 Jul 2008) | 15 lines Merged revisions 133038 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r133038 | mmichelson | 2008-07-23 12:50:01 -0500 (Wed, 23 Jul
- 2008) | 7 lines 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 ........
- ................
-
-2008-07-23 17:21 +0000 [r132978-132983] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 132981 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r132981 | tilghman | 2008-07-23 12:20:43 -0500 (Wed, 23 Jul 2008)
- | 6 lines Yet another conversion of '|' to ',' (closes issue
- #13137) Reported by: eliel Patches: chan_iax2trunk-IAXPEER.patch
- uploaded by eliel (license 64) ........
-
- * contrib/scripts/asterisk.logrotate (added), /: Merged revisions
- 132977 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r132977 |
- tilghman | 2008-07-23 12:14:56 -0500 (Wed, 23 Jul 2008) | 6 lines
- Add logrotate script for Asterisk (closes issue #13085) Reported
- by: pabelanger Patches: logrotate uploaded by pabelanger (license
- 224) ........
-
-2008-07-23 16:42 +0000 [r132965-132967] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/misdn/isdn_lib.c, /: Merged revisions 132883,132966 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r132883 | crichter | 2008-07-23 07:07:15 -0500
- (Wed, 23 Jul 2008) | 9 lines Merged revisions 132826 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r132826 | crichter | 2008-07-23 13:37:50 +0200 (Mi, 23
- Jul 2008) | 1 line 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. ........ ................
- r132966 | kpfleming | 2008-07-23 11:38:28 -0500 (Wed, 23 Jul
- 2008) | 2 lines use correct function name... please compile with
- --enable-dev-mode ................
-
- * include/asterisk/stringfields.h, /, main/utils.c: Merged
- revisions 132964 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r132964 | kpfleming | 2008-07-23 11:30:18 -0500 (Wed, 23 Jul
- 2008) | 10 lines Merged revisions 132872 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r132872 | kpfleming | 2008-07-23 06:52:18 -0500 (Wed, 23 Jul
- 2008) | 2 lines 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 08:18 +0000 [r132824] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 132823 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r132823 |
- oej | 2008-07-23 10:13:07 +0200 (Ons, 23 Jul 2008) | 8 lines
- Well, the content of a channel variable may be longer than the
- size of a pointer... Thanks, eliel! Reported by: eliel Patches:
- chan_siptrunk.SIPPEER.patch uploaded by eliel (license 64)
- (closes issue #13135) ........
-
-2008-07-22 22:20 +0000 [r132797] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 132795 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r132795 | mmichelson | 2008-07-22 17:17:09 -0500 (Tue, 22 Jul
- 2008) | 11 lines Merged revisions 132777 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ 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 Modified:
- branches/1.4/channels/chan_sip.c ........ ................
-
-2008-07-22 22:15 +0000 [r132793] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 132791 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r132791 | kpfleming | 2008-07-22 17:14:37 -0500 (Tue, 22 Jul
- 2008) | 2 lines correct fix made in r132777... the code *did*
- compile in dev-mode, as long as libpri was installed and enabled
- ........
-
-2008-07-22 21:59 +0000 [r132782] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c, doc/sip-retransmit.txt (added): Merged
- revisions 132703 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r132703 | oej | 2008-07-22 22:46:11 +0200 (Tis, 22 Jul 2008) | 17
- lines Merged revisions 132645 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r132645 | oej | 2008-07-22 22:10:26 +0200 (Tis, 22 Jul 2008) | 9
- lines 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 21:55 +0000 [r132780] Tilghman Lesher <tlesher@digium.com>
-
- * configs/iax.conf.sample, /, channels/chan_iax2.c: Merged
- revisions 132778 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r132778 | tilghman | 2008-07-22 16:53:40 -0500 (Tue, 22 Jul 2008)
- | 18 lines Merged revisions 132713 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r132713 | tilghman | 2008-07-22 16:19:39 -0500
- (Tue, 22 Jul 2008) | 10 lines 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 21:54 +0000 [r132779] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 132777 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r132777 | mmichelson | 2008-07-22 16:52:24 -0500 (Tue, 22 Jul
- 2008) | 3 lines Get chan_dahdi to compile in devmode ........
-
-2008-07-22 21:23 +0000 [r132574-132729] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 132721 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r132721 | kpfleming | 2008-07-22 16:21:56 -0500
- (Tue, 22 Jul 2008) | 14 lines Merged revisions 132712 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r132712 | kpfleming | 2008-07-22 16:17:23 -0500 (Tue, 22 Jul
- 2008) | 6 lines 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: Merged revisions 132705
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r132705 | kpfleming | 2008-07-22 15:54:07 -0500
- (Tue, 22 Jul 2008) | 10 lines Merged revisions 132704 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r132704 | kpfleming | 2008-07-22 15:49:41 -0500 (Tue, 22 Jul
- 2008) | 2 lines make AST_C_COMPILE_CHECK able to print a 'pretty'
- description of what it is doing ........ ................
-
- * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /,
- configure, include/asterisk/autoconfig.h.in, configure.ac: Merged
- revisions 132643 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r132643 | kpfleming | 2008-07-22 14:59:10 -0500 (Tue, 22 Jul
- 2008) | 10 lines Merged revisions 132641 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r132641 | kpfleming | 2008-07-22 14:49:11 -0500 (Tue, 22 Jul
- 2008) | 2 lines use renamed libpri API call for controlling this
- feature (was improperly named before) ........ ................
-
- * channels/chan_dahdi.c, /: Merged revisions 132573 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r132573 | kpfleming | 2008-07-21 17:51:16 -0500
- (Mon, 21 Jul 2008) | 10 lines Merged revisions 132571 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r132571 | kpfleming | 2008-07-21 17:45:16 -0500 (Mon, 21 Jul
- 2008) | 2 lines 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 21:13 +0000 [r132515] Brett Bryant <bbryant@digium.com>
-
- * configs/features.conf.sample, configs/gtalk.conf.sample, /,
- configs/jingle.conf.sample, configs/manager.conf.sample: Merged
- revisions 132514 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r132514 |
- bbryant | 2008-07-21 16:12:51 -0500 (Mon, 21 Jul 2008) | 8 lines
- Update configuration files to add missing options for jingle,
- gtalk, manager.conf, and features.conf. (closes issue #13128)
- Reported by: caio1982 Patches: missing_options1.diff uploaded by
- caio1982 (license 22) ........
-
-2008-07-21 21:02 +0000 [r132512-132513] Tilghman Lesher <tlesher@digium.com>
-
- * main/fskmodem.c (added), /, include/asterisk/fskmodem.h (added):
- Merged revisions 132511 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r132511 |
- tilghman | 2008-07-21 16:00:47 -0500 (Mon, 21 Jul 2008) | 2 lines
- (Step 2 of 2) ........
-
- * main/fskmodem.c (removed), include/asterisk/fskmodem_int.h
- (added), build_tools/cflags.xml, main/fskmodem_float.c (added),
- /, main/tdd.c, include/asterisk/fskmodem.h (removed),
- main/fskmodem_int.c (added), main/callerid.c,
- include/asterisk/fskmodem_float.h (added): Merged revisions
- 132510 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r132510 |
- tilghman | 2008-07-21 15:59:03 -0500 (Mon, 21 Jul 2008) | 5 lines
- Optionally build integer-based routines for FSK tone decoding
- (but default to the more accurate float-based routines). (Closes
- issue #11679) (Step 1 of 2) ........
-
-2008-07-21 20:55 +0000 [r132467-132509] Brett Bryant <bbryant@digium.com>
-
- * /, apps/app_sendtext.c: Merged revisions 132508 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r132508 |
- bbryant | 2008-07-21 15:54:09 -0500 (Mon, 21 Jul 2008) | 9 lines
- Fix a bug where SENDTEXTSTATUS isn't set properly when it isn't
- supported on a channel (yet _another_ useful patch by eliel).
- (closes issue #13081) Reported by: eliel Patches:
- app_sendtext.c.patch uploaded by eliel (license 64) Tested by:
- eliel ........
-
- * /, channels/chan_sip.c: Merged revisions 132468 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r132468 |
- bbryant | 2008-07-21 12:42:45 -0500 (Mon, 21 Jul 2008) | 5 lines
- Fix bug where ast_parse_arg would inadvertantly enable sip tcp
- when parsing a tcpbindaddr if it was disabled. (closes issue
- #13117) Reported by: pj ........
-
- * /, channels/chan_iax2.c: Merged revisions 132466 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r132466 | bbryant | 2008-07-21 12:22:02 -0500 (Mon, 21 Jul 2008)
- | 3 lines Fix an issue in iax2 where a call that's been rejected
- still kept an open channel on the side that attempted to make the
- call (not the side of the call that rejected the call). Changes
- were load tested and also approved by Russell. ........
-
-2008-07-21 15:34 +0000 [r132426] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 132425 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r132425 | jpeeler | 2008-07-21 10:33:13 -0500 (Mon, 21 Jul 2008)
- | 2 lines make buffers config option (chan_dahdi.conf) parsing
- safer and added logging in case of failure ........
-
-2008-07-21 14:48 +0000 [r132389-132391] Russell Bryant <russell@digium.com>
-
- * apps/app_jack.c, include/asterisk/libresample.h (removed), /,
- build_tools/menuselect-deps.in, configure,
- include/asterisk/autoconfig.h.in, main/Makefile, main/libresample
- (removed), configure.ac, codecs/codec_resample.c, makeopts.in:
- Merged revisions 132390 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r132390 |
- russell | 2008-07-21 09:47:41 -0500 (Mon, 21 Jul 2008) | 16 lines
- Remove libresample from the Asterisk source tree. It is now
- available in its own repository, and must be installed like any
- other library for Asterisk to use. The two modules that require
- it are codec_resample and app_jack. To install libresample: $ svn
- co http://svn.digium.com/svn/libresample/trunk libresample $ cd
- libresample $ ./configure $ make $ sudo make install This code is
- currently in our own repository because the build system did not
- include the appropriate targets for building a dynamic library or
- for installing the library. ........
-
- * apps/app_jack.c, /, codecs/codec_resample.c: Merged revisions
- 132388 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r132388 |
- russell | 2008-07-21 08:51:05 -0500 (Mon, 21 Jul 2008) | 3 lines
- Enable higher quality resampling, as it doesn't have a noticeable
- performance impact on my machine .. ........
-
-2008-07-19 16:47 +0000 [r132313] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, LICENSE: Merged revisions 132312 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r132312 | kpfleming | 2008-07-19 11:46:33 -0500 (Sat, 19 Jul
- 2008) | 10 lines Merged revisions 132311 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r132311 | kpfleming | 2008-07-19 11:45:52 -0500 (Sat, 19 Jul
- 2008) | 2 lines 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) ........
- ................
-
-2008-07-19 10:47 +0000 [r132278] Michiel van Baak <michiel@vanbaak.info>
-
- * res/res_config_sqlite.c, /: Merged revisions 132277 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r132277 | mvanbaak | 2008-07-19 12:46:12 +0200 (Sat, 19 Jul 2008)
- | 7 lines fix a couple of comments in sqlite resource driver.
- (closes issue #13110) Reported by: gknispel_proformatique
- Patches: res_config_sqlite_comments.patch uploaded by gknispel
- (license 261) ........
-
-2008-07-18 22:20 +0000 [r132245] Brett Bryant <bbryant@digium.com>
-
- * main/manager.c, /: Merged revisions 132242 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r132242 |
- bbryant | 2008-07-18 17:19:56 -0500 (Fri, 18 Jul 2008) | 4 lines
- Fixes problem where manager users loaded from users.conf would be
- removed early (before the routine to load the configuration was
- finished) because a variable wasn't initialized. ........
-
-2008-07-18 20:58 +0000 [r132114-132207] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/say.c: Merged revisions 132113 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r132113 | tilghman | 2008-07-18 14:09:39 -0500 (Fri, 18 Jul 2008)
- | 14 lines Merged revisions 132112 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r132112 | tilghman | 2008-07-18 14:06:10 -0500 (Fri, 18 Jul 2008)
- | 6 lines Fix for Taiwanese number syntax (closes issue #12319)
- Reported by: CharlesWang Patches: saynumber-tw-1.4.18.1.patch
- uploaded by CharlesWang (license 444) ........ ................
-
-2008-07-18 18:53 +0000 [r132111] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 132108 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r132108 | mattf | 2008-07-18 13:50:00 -0500 (Fri, 18 Jul 2008) |
- 1 line Make sure we break the poll so that messages queued will
- be sent on the SS7 when using CLI commands for blocking and
- blocking of CICs and linksets. ........
-
-2008-07-18 18:51 +0000 [r132110] Tilghman Lesher <tlesher@digium.com>
-
- * main/config.c, /: Merged revisions 132109 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r132109 | tilghman | 2008-07-18 13:50:37 -0500 (Fri, 18 Jul 2008)
- | 14 lines Merged revisions 132107 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r132107 | tilghman | 2008-07-18 13:47:50 -0500 (Fri, 18 Jul 2008)
- | 6 lines Textual clarification (closes issue #13106) Reported
- by: flefoll Patches: config.c.br14.120173.patch-unknown-directive
- uploaded by flefoll (license 244) ........ ................
-
-2008-07-18 17:56 +0000 [r132051] Brett Bryant <bbryant@digium.com>
-
- * main/hashtab.c, /, cdr/cdr_radius.c: Merged revisions 132050 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r132050 | bbryant | 2008-07-18 12:55:41 -0500 (Fri, 18
- Jul 2008) | 8 lines Fix magic Revision keywords in hashtab.c and
- change cdr_radius.c to use the same keyword as the other files
- (patch by eliel). (closes issue #13104) Reported by: eliel
- Patches: revision.patch uploaded by eliel (license 64) ........
-
-2008-07-18 17:40 +0000 [r131984-132047] Tilghman Lesher <tlesher@digium.com>
-
- * main/sched.c, /: Merged revisions 131989 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r131989 | tilghman | 2008-07-18 12:10:34 -0500 (Fri, 18 Jul 2008)
- | 10 lines Merged revisions 131988 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r131988 | tilghman | 2008-07-18 12:10:01 -0500 (Fri, 18 Jul 2008)
- | 2 lines Oops ........ ................
-
- * main/sched.c, /, include/asterisk/sched.h: Merged revisions
- 131986 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r131986 | tilghman | 2008-07-18 11:48:18 -0500 (Fri, 18 Jul 2008)
- | 10 lines Merged revisions 131985 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r131985 | tilghman | 2008-07-18 11:46:23 -0500 (Fri, 18 Jul 2008)
- | 2 lines Preserve ABI compatibility with last change ........
- ................
-
- * main/sched.c, /, include/asterisk/sched.h, channels/chan_iax2.c:
- Merged revisions 131982 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r131982 | tilghman | 2008-07-18 11:33:56 -0500 (Fri, 18 Jul 2008)
- | 10 lines Merged revisions 131970 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r131970 | tilghman | 2008-07-18 11:30:31 -0500 (Fri, 18 Jul 2008)
- | 2 lines Make the ast_assert call within ast_sched_del report
- something useful. ........ ................
-
-2008-07-18 16:16 +0000 [r131924] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/dlfcn.c (removed), main/loader.c, /, main/Makefile,
- include/asterisk/dlfcn-compat.h (removed): Merged revisions
- 131923 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r131923 | kpfleming | 2008-07-18 11:16:12 -0500 (Fri, 18 Jul
- 2008) | 10 lines Merged revisions 131921 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r131921 | kpfleming | 2008-07-18 11:15:41 -0500 (Fri, 18 Jul
- 2008) | 2 lines 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 15:39 +0000 [r131917] Brett Bryant <bbryant@digium.com>
-
- * /, main/features.c: Merged revisions 131916 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r131916 | bbryant | 2008-07-18 10:38:22 -0500 (Fri, 18 Jul 2008)
- | 12 lines Merged revisions 131915 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r131915 | bbryant | 2008-07-18 10:34:42 -0500 (Fri, 18 Jul 2008)
- | 4 lines 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 22:45 +0000 [r131869] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 131868 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r131868 | jpeeler | 2008-07-17 17:40:00 -0500 (Thu, 17 Jul 2008)
- | 6 lines 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". ........
-
-2008-07-17 21:27 +0000 [r131830] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_senddtmf.c: Merged revisions 131824 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r131824 |
- mmichelson | 2008-07-17 16:26:41 -0500 (Thu, 17 Jul 2008) | 10
- lines Document that the duration of dtmf may be passed to the
- SendDTMF application. Also correct the default pause between
- digits. (closes issue #13102) Reported by: eliel Patches:
- app_senddtmf.c.patch uploaded by eliel (license 64) ........
-
-2008-07-17 20:38 +0000 [r131754-131792] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 131791 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r131791 | tilghman | 2008-07-17 15:37:14 -0500
- (Thu, 17 Jul 2008) | 15 lines Merged revisions 131790 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r131790 | tilghman | 2008-07-17 15:35:44 -0500 (Thu, 17 Jul 2008)
- | 7 lines 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) ........ ................
-
- * res/res_config_sqlite.c, /: Merged revisions 131753 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r131753 | tilghman | 2008-07-17 13:36:34 -0500 (Thu, 17 Jul 2008)
- | 6 lines Fix memory leaks (closes issue #13099) Reported by:
- gknispel_proformatique Patches:
- res_config_sqlite_leak_on_error.patch uploaded by gknispel
- (license 261) ........
-
-2008-07-17 18:15 +0000 [r131718] Brett Bryant <bbryant@digium.com>
-
- * /, main/features.c: Merged revisions 131717 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r131717 |
- bbryant | 2008-07-17 13:14:42 -0500 (Thu, 17 Jul 2008) | 8 lines
- Fix a memory leak in register_group_feature when attempting to
- register a feature without specifying a group or feature to
- register. (closes issue #13101) Reported by: eliel Patches:
- features.c.patch uploaded by eliel (license 64) ........
-
-2008-07-17 15:46 +0000 [r131682] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_config_sqlite.c, /: Merged revisions 131681 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r131681 | tilghman | 2008-07-17 10:45:25 -0500 (Thu, 17 Jul 2008)
- | 4 lines Fix memory leak. (Closes issue #13096) Reported by
- gknispel_proformatique ........
-
-2008-07-16 23:56 +0000 [r131571] Steve Murphy <murf@digium.com>
-
- * /: The commit from 131570 should not be applied to 1.6.0, as it
- is not as necessary, because log_show_lock in trunk is not
- available in 1.6.0, and is estimated to be the only function that
- might care about the lock_addr values.
-
-2008-07-16 22:18 +0000 [r131493] Brett Bryant <bbryant@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 131492 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r131492 | bbryant | 2008-07-16 17:17:36 -0500
- (Wed, 16 Jul 2008) | 14 lines Merged revisions 131491 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r131491 | bbryant | 2008-07-16 17:17:07 -0500 (Wed, 16 Jul 2008)
- | 6 lines 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 21:54 +0000 [r131455-131486] Brett Bryant <bbryant@digium.com>
-
- * /, funcs/func_sysinfo.c: Merged revisions 131484 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r131484 | bbryant | 2008-07-16 16:54:08 -0500 (Wed, 16 Jul 2008)
- | 4 lines Fixes sysinfo operator issue also fixed elsewhere in
- r131445. (issue #13057) ........
-
- * main/asterisk.c, /: Merged revisions 131445 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r131445 |
- bbryant | 2008-07-16 16:24:18 -0500 (Wed, 16 Jul 2008) | 9 lines
- Fixes an issue with "core show sysinfo" that used the wrong
- operator to calculate the number of bytes from a sysinfo
- structure. unsigned long. (closes issue #13057) Reported by:
- eliel Patches: asterisk.c.patch uploaded by eliel (license 64)
- ........
-
-2008-07-16 20:48 +0000 [r131423] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 131422 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r131422 | russell | 2008-07-16 15:48:27 -0500
- (Wed, 16 Jul 2008) | 15 lines Merged revisions 131421 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r131421 | russell | 2008-07-16 15:47:53 -0500 (Wed, 16 Jul 2008)
- | 7 lines 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 20:24 +0000 [r131301-131378] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c: Merged revisions 131375 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r131375 | mmichelson | 2008-07-16 15:24:12 -0500 (Wed, 16 Jul
- 2008) | 22 lines Merged revisions 131369 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r131369 | mmichelson | 2008-07-16 15:23:02 -0500 (Wed, 16 Jul
- 2008) | 14 lines 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 ........ ................
-
- * res/res_config_sqlite.c, /: Merged revisions 131361 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r131361 | mmichelson | 2008-07-16 14:57:02 -0500 (Wed, 16 Jul
- 2008) | 9 lines Don't try to dereference the dbfile pointer if we
- know that it's NULL. (closes issue #13092) Reported by:
- gknispel_proformatique Patches:
- trunk_sqlite_check_vars_null.patch uploaded by gknispel (license
- 261) ........
-
- * /, apps/app_queue.c: Merged revisions 131358 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r131358 | mmichelson | 2008-07-16 14:37:42 -0500 (Wed, 16 Jul
- 2008) | 14 lines Merged revisions 131357 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r131357 | mmichelson | 2008-07-16 14:37:08 -0500 (Wed, 16 Jul
- 2008) | 6 lines Apparently, "thread safety" is important,
- whatever that means. :P (Thanks Russell!) ........
- ................
-
- * /, apps/app_queue.c: Merged revisions 131300 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r131300 | mmichelson | 2008-07-16 13:59:27 -0500 (Wed, 16 Jul
- 2008) | 21 lines Merged revisions 131299 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r131299 | mmichelson | 2008-07-16 13:57:34 -0500 (Wed, 16 Jul
- 2008) | 13 lines 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 18:20 +0000 [r131248] Steve Murphy <murf@digium.com>
-
- * res/ael/pval.c, /: Merged revisions 131243 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r131243 | murf | 2008-07-16 11:59:33 -0600 (Wed, 16 Jul 2008) |
- 27 lines Merged revisions 131242 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r131242 | murf | 2008-07-16 11:53:43 -0600 (Wed, 16 Jul 2008) |
- 19 lines (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.
- ........ ................
-
-2008-07-16 15:04 +0000 [r131206] Luigi Rizzo <rizzo@icir.org>
-
- * channels/chan_agent.c: add missing terminator argument for
- ast_event_subscribe(). Without it the function will randomly walk
- on the stack possibly causing a panic
-
-2008-07-16 00:54 +0000 [r131168] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/logger.c: Merged revisions 131166 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r131166 |
- tilghman | 2008-07-15 19:52:48 -0500 (Tue, 15 Jul 2008) | 3 lines
- Fix rotate strategy (Closes issue #13086) ........
-
-2008-07-15 23:41 +0000 [r131131] Steve Murphy <murf@digium.com>
-
- * main/pbx.c, /: Merged revisions 131129 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r131129 |
- murf | 2008-07-15 17:36:19 -0600 (Tue, 15 Jul 2008) | 21 lines
- (closes issue #12960) Reported by: mnicholson Spent most of the
- day on this bug, and the solution was so simple. Just had to find
- and understand the problem. The problem was, that the routine to
- copy the existing switches, includes, and ignorepats from the old
- context to the new one, wasn't getting called when the context is
- already existent. (In other words, if AEL is adding a new context
- to the mix, they get copied, but if pbx_config already defined a
- context, then the copy wasn't happening. This made no sense, so I
- moved the call to copy the includes & etc, no matter the case.
- ........
-
-2008-07-15 18:47 +0000 [r131073] Russell Bryant <russell@digium.com>
-
- * /, res/res_agi.c: Merged revisions 131072 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r131072 |
- russell | 2008-07-15 13:46:40 -0500 (Tue, 15 Jul 2008) | 5 lines
- Fix a couple of places in res_agi where the agi_commands lock
- would not be released, causing a deadlock. (Reported by mvanbaak
- in #asterisk-dev, discovered by bbryant's change to the lock
- tracking code to yell at you if a thread exits with a lock still
- held) ........
-
-2008-07-15 18:29 +0000 [r131060] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c, main/manager.c, /, channels/chan_sip.c: Merged
- revisions 131044 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r131044 | tilghman | 2008-07-15 13:25:34 -0500 (Tue, 15 Jul 2008)
- | 16 lines Merged revisions 130959 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r130959 | tilghman | 2008-07-15 12:19:13 -0500 (Tue, 15 Jul 2008)
- | 8 lines 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) ........ ................
-
-2008-07-15 18:00 +0000 [r131014] Michiel van Baak <michiel@vanbaak.info>
-
- * main/cdr.c, /: Merged revisions 131013 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r131013 | mvanbaak | 2008-07-15 19:49:48 +0200 (Tue, 15 Jul 2008)
- | 15 lines Merged revisions 131012 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r131012 | mvanbaak | 2008-07-15 19:47:15 +0200 (Tue, 15 Jul 2008)
- | 7 lines 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 13:14 +0000 [r130946] Steve Murphy <murf@digium.com>
-
- * utils/conf2ael.c, utils/Makefile, res/ael/pval.c,
- channels/chan_skinny.c, res/ael/ael.tab.c, main/features.c,
- pbx/pbx_ael.c, res/ael/ael_lex.c, res/ael/ael.tab.h,
- utils/ael_main.c, include/asterisk/pbx.h, utils/extconf.c,
- res/ael/ael.flex, pbx/pbx_config.c, apps/app_stack.c,
- apps/app_dial.c, main/pbx.c, include/asterisk/pval.h, /,
- channels/chan_sip.c, apps/app_meetme.c, res/ael/ael.y,
- channels/chan_iax2.c, apps/app_queue.c: Merged revisions 130145
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- Merging this rev from trunk to 1.6.0 was not simple. Why? Because
- we've enhanced trunk to do a [fast] merge-and-delete operation
- which also solved problems with contexts having entries from
- different registrars. Fast as in the amount of time the contexts
- are locked down. That *is* fast, but traversing the entire
- dialplan looking for priorities to delete takes more time
- overall. This particular fix involved pulling in those
- enhancements from trunk, along with all the various fixes and
- refinements made along the way. Merging all this from trunk into
- 1.6 involved: a. mergetrunk6 in the stuff from 130145; b. revert
- all but the prop changes c. catalog all revisions to pbx.c since
- 1.6.0 was forked (at rev 105596). d. catalog all revisions to
- pbx.c in trunk since 1.6.0 was forked, making special note of all
- revs that were not merged into 1.6.0. e. study each rev in trunk
- not applied to 1.6.0, and determine if it was involved in the
- merge_and_delete enhancements in trunk. 25 commits were done in
- 1.6.0, all but one (106306) was a merge from trunk. Trunk had 22
- additional changes, of which 7 were involved in the
- merge_and_delete enhancements: 106757 108894 109169 116461 123358
- 130145 130297 f. Go to trunk and collect patches, one by one, of
- the changes made by each rev across the entire source tree, using
- svn diff -c <num> > pfile g. Apply each patch in order to 1.6.0,
- and resolve all failures and compilation problems before
- proceding to the next patch. h. test the stuff. i. profit!
- ........ r130145 | murf | 2008-07-11 12:24:31 -0600 (Fri, 11 Jul
- 2008) | 40 lines (closes issue #13041) Reported by: eliel Tested
- by: murf (closes issue #12960) Reported by: mnicholson In this
- 'omnibus' fix, I **think** I solved both the problem in 13041,
- where unloading pbx_ael.so caused crashes, or incomplete removal
- of previous registrar'ed entries. And I added code to completely
- remove all includes, switches, and ignorepats that had a matching
- registrar entry, which should appease 12960. I also added a lot
- of seemingly useless brackets around single statement if's, which
- helped debug so much that I'm leaving them there. I added a
- routine to check the correlation between the extension tree lists
- and the hashtab tables. It can be amazingly helpful when you have
- lots of dialplan stuff, and need to narrow down where a problem
- is occurring. It's ifdef'd out by default. I cleaned up the code
- around the new CIDmatch code. It was leaving hanging extens with
- bad ptrs, getting confused over which objects to remove, etc. I
- tightened up the code and changed the call to remove_exten in the
- merge_and_delete code. I added more conditions to check for empty
- context worthy of deletion. It's not empty if there are any
- includes, switches, or ignorepats present. If I've missed
- anything, please re-open this bug, and be prepared to supply
- example dialplan code. ........
-
-2008-07-15 00:00 +0000 [r130891] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 130890 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r130890 | tilghman | 2008-07-14 18:59:54 -0500
- (Mon, 14 Jul 2008) | 16 lines Merged revisions 130889 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r130889 | tilghman | 2008-07-14 18:59:13 -0500 (Mon, 14 Jul 2008)
- | 8 lines 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 22:24 +0000 [r130795-130855] Mark Michelson <mmichelson@digium.com>
-
- * main/asterisk.c, /: Merged revisions 130854 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r130854 |
- mmichelson | 2008-07-14 17:22:57 -0500 (Mon, 14 Jul 2008) | 9
- lines Fix a memory leak in the case that /dev/null cannot be
- opened when running startup commands from cli.conf (closes issue
- #13066) Reported by: eliel Patches: asterisk.c.patch uploaded by
- eliel (license 64) ........
-
- * apps/app_dial.c, /: Merged revisions 130794 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r130794 | mmichelson | 2008-07-14 12:54:11 -0500 (Mon, 14 Jul
- 2008) | 16 lines Merged revisions 130792 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r130792 | mmichelson | 2008-07-14 12:50:21 -0500 (Mon, 14 Jul
- 2008) | 8 lines 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 17:22 +0000 [r130752] Michiel van Baak <michiel@vanbaak.info>
-
- * main/dnsmgr.c, /: Merged revisions 130744 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r130744 | mvanbaak | 2008-07-14 19:21:18 +0200 (Mon, 14 Jul 2008)
- | 18 lines Merged revisions 130735 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r130735 | mvanbaak | 2008-07-14 19:10:21 +0200 (Mon, 14 Jul 2008)
- | 10 lines 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 10:40 +0000 [r130636-130637] Russell Bryant <russell@digium.com>
-
- * /, include/asterisk/astobj.h: Merged revisions 129987 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r129987 | russell | 2008-07-11 09:22:44 -0500
- (Fri, 11 Jul 2008) | 10 lines Merged revisions 129970 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r129970 | russell | 2008-07-11 09:18:43 -0500 (Fri, 11 Jul 2008)
- | 2 lines add a simple ASTOBJ_TRYWRLOCK macro ... ........
- ................
-
- * /, main/audiohook.c: Merged revisions 130635 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r130635 | russell | 2008-07-14 05:39:23 -0500 (Mon, 14 Jul 2008)
- | 10 lines Merged revisions 130634 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r130634 | russell | 2008-07-14 05:38:14 -0500 (Mon, 14 Jul 2008)
- | 2 lines Bump up the debug level for a message. ........
- ................
-
-2008-07-13 23:20 +0000 [r130575-130582] Michiel van Baak <michiel@vanbaak.info>
-
- * /, doc/tex/Makefile, build_tools/prep_tarball, res/Makefile:
- Merged revisions 130578 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r130578 |
- mvanbaak | 2008-07-14 01:14:03 +0200 (Mon, 14 Jul 2008) | 15
- lines Make all sed calls Posix sed compatible. To make sure
- nobody commits script-modified files we first make a backup of
- asterisk.tex, run the script, generate the pdf and / or html, and
- put the original asterisk.tex back. This will guard us for the
- stuff that happened before that someone committed a locally
- modified asterisk.tex, with changes done by this script. (closes
- issue #13062) Reported by: mvanbaak Patches:
- sed_without-i-v3.diff uploaded by mvanbaak (license 7) Tested by:
- mvanbaak Feedback from Corydon. Thanks for taking the time to go
- through this. ........
-
- * main/manager.c, /: Merged revisions 130574 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r130574 | mvanbaak | 2008-07-14 00:50:31 +0200 (Mon, 14 Jul 2008)
- | 16 lines Merged revisions 130573 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r130573 | mvanbaak | 2008-07-14 00:48:51 +0200 (Mon, 14 Jul 2008)
- | 8 lines 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 17:59 +0000 [r130516] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 130515 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r130515 | tilghman | 2008-07-13 12:58:47 -0500
- (Sun, 13 Jul 2008) | 12 lines Merged revisions 130514 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r130514 | tilghman | 2008-07-13 12:56:10 -0500 (Sun, 13 Jul 2008)
- | 4 lines Reverting 2 changesets, as it breaks incoming IAX2
- calls (Related to issue #12963) Reported by: mvanbaak ........
- ................
-
-2008-07-13 15:00 +0000 [r130480] Michiel van Baak <michiel@vanbaak.info>
-
- * doc/tex/asterisk.tex, /: Merged revisions 130479 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r130479 | mvanbaak | 2008-07-13 16:58:40 +0200 (Sun, 13 Jul 2008)
- | 3 lines restore ASTERISKVERSION marker to asterisk.tex. This
- got lost in commit 97634 ........
-
-2008-07-13 02:35 +0000 [r130445] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 130444 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r130444 | tilghman | 2008-07-12 21:34:32 -0500 (Sat, 12 Jul 2008)
- | 2 lines Unlock list before returning ........
-
-2008-07-11 21:39 +0000 [r130294] Matthew Fredrickson <creslin@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 130293 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r130293 | mattf | 2008-07-11 16:36:26 -0500 (Fri, 11 Jul 2008) |
- 1 line Support new TRANSPORT definitions in libss7 ........
-
-2008-07-11 20:04 +0000 [r130238] Mark Michelson <mmichelson@digium.com>
-
- * /, main/audiohook.c: Merged revisions 130237 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r130237 | mmichelson | 2008-07-11 15:03:55 -0500 (Fri, 11 Jul
- 2008) | 11 lines Merged revisions 130236 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r130236 | mmichelson | 2008-07-11 15:03:23 -0500 (Fri, 11 Jul
- 2008) | 3 lines Remove redundant logic ........ ................
-
-2008-07-11 19:57 +0000 [r130231-130235] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_dahdi.c, /, channels/chan_agent.c, utils/astman.c:
- Merged revisions 130230 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r130230 |
- tilghman | 2008-07-11 14:40:55 -0500 (Fri, 11 Jul 2008) | 2 lines
- Fix trunk breakage ........
-
-2008-07-11 19:14 +0000 [r130175] Mark Michelson <mmichelson@digium.com>
-
- * /, main/audiohook.c: Merged revisions 130174 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r130174 | mmichelson | 2008-07-11 14:14:15 -0500 (Fri, 11 Jul
- 2008) | 15 lines Merged revisions 130173 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r130173 | mmichelson | 2008-07-11 14:13:29 -0500 (Fri, 11 Jul
- 2008) | 7 lines 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 18:53 +0000 [r130171] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 130170 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r130170 | tilghman | 2008-07-11 13:52:42 -0500
- (Fri, 11 Jul 2008) | 15 lines Merged revisions 130169 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r130169 | tilghman | 2008-07-11 13:51:56 -0500 (Fri, 11 Jul 2008)
- | 7 lines 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)
- ........ ................
-
-2008-07-11 18:33 +0000 [r130168] Sean Bright <sean.bright@gmail.com>
-
- * /, channels/chan_sip.c: Merged revisions 130167 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r130167 |
- seanbright | 2008-07-11 14:32:26 -0400 (Fri, 11 Jul 2008) | 1
- line Missed one. Formatting only. ........
-
-2008-07-11 18:14 +0000 [r130130] Brett Bryant <bbryant@digium.com>
-
- * main/cli.c, channels/chan_jingle.c, channels/chan_dahdi.c,
- channels/chan_skinny.c, main/abstract_jb.c, apps/app_minivm.c,
- codecs/codec_resample.c, codecs/codec_dahdi.c,
- apps/app_chanspy.c, main/asterisk.c, apps/app_milliwatt.c,
- main/dsp.c, codecs/codec_g722.c, /, channels/chan_sip.c,
- main/threadstorage.c, utils/astman.c, main/utils.c,
- channels/chan_gtalk.c, pbx/dundi-parser.c: Merged revisions
- 130129 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r130129 |
- bbryant | 2008-07-11 13:09:35 -0500 (Fri, 11 Jul 2008) | 8 lines
- Janitor patch to change uses of sizeof to ARRAY_LEN (closes issue
- #13054) Reported by: pabelanger Patches: ARRAY_LEN.patch2
- uploaded by pabelanger (license 224) Tested by: seanbright
- ........
-
-2008-07-11 17:30 +0000 [r130127] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_agent.c: Merged revisions 130126 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r130126 | tilghman | 2008-07-11 12:29:24 -0500
- (Fri, 11 Jul 2008) | 17 lines Merged revisions 130102 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r130102 | tilghman | 2008-07-11 11:50:42 -0500 (Fri, 11 Jul 2008)
- | 9 lines 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 16:18 +0000 [r129936-130045] Kevin P. Fleming <kpfleming@digium.com>
-
- * doc/ss7.txt, /, contrib/utils/zones2indications.c, CHANGES:
- Merged revisions 130044 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r130044 |
- kpfleming | 2008-07-11 11:18:01 -0500 (Fri, 11 Jul 2008) | 2
- lines clean up a bunch more Zaptel-related references ........
-
- * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /,
- configure, include/asterisk/autoconfig.h.in, configure.ac: Merged
- revisions 130040 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r130040 | kpfleming | 2008-07-11 10:57:17 -0500 (Fri, 11 Jul
- 2008) | 12 lines Merged revisions 130039 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r130039 | kpfleming | 2008-07-11 10:41:56 -0500 (Fri, 11 Jul
- 2008) | 4 lines 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) ........ ................
-
- * /, main/astmm.c: Merged revisions 129968 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r129968 | kpfleming | 2008-07-11 09:16:15 -0500 (Fri, 11 Jul
- 2008) | 18 lines Merged revisions 129966 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r129966 | kpfleming | 2008-07-11 09:03:52 -0500 (Fri, 11 Jul
- 2008) | 5 lines 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 :-) ........ r129967 | kpfleming |
- 2008-07-11 09:03:52 -0500 (Fri, 11 Jul 2008) | 5 lines simplify
- calculation ........ ................
-
- * /, sounds/Makefile: Merged revisions 129916 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r129916 | kpfleming | 2008-07-11 07:21:29 -0500 (Fri, 11 Jul
- 2008) | 10 lines Merged revisions 129907 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r129907 | kpfleming | 2008-07-11 07:15:42 -0500 (Fri, 11 Jul
- 2008) | 2 lines don't attempt to set user/group ownership of
- extracted sound files (reported on asterisk-users) ........
- ................
-
-2008-07-11 01:01 +0000 [r129865] Sean Bright <sean.bright@gmail.com>
-
- * res/res_config_pgsql.c, /, res/res_config_ldap.c: Merged
- revisions 129864 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r129864 |
- seanbright | 2008-07-10 20:55:06 -0400 (Thu, 10 Jul 2008) | 1
- line Fix some usages of snprintf, and clarify a couple variable
- names. ........
-
-2008-07-10 22:07 +0000 [r129764-129805] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_iax2.c: Merged revisions 129804 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r129804 | tilghman | 2008-07-10 17:06:07 -0500
- (Thu, 10 Jul 2008) | 16 lines Merged revisions 129803 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r129803 | tilghman | 2008-07-10 16:57:05 -0500 (Thu, 10 Jul 2008)
- | 8 lines 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: