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authorlmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b>2010-11-15 19:27:55 +0000
committerlmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b>2010-11-15 19:27:55 +0000
commit59710d58611dcc4144d272f5ea9fcbbbd2e3c24a (patch)
treedfaa086a0a1af2cb46d0cdc983446d168831fb15
parent2aeb111bc4e335072f356df04c678708f29b9520 (diff)
parent758e6ad7a9346642dc7c97fe5693a9de3e780805 (diff)
Creating tag for the release of asterisk-1.6.2.15-rc1
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.2.15-rc1@295120 f38db490-d61c-443f-a65b-d21fe96a405b
-rw-r--r--.lastclean1
-rw-r--r--.version1
-rw-r--r--ChangeLog27611
-rw-r--r--asterisk-1.6.2.15-rc1-summary.html176
-rw-r--r--asterisk-1.6.2.15-rc1-summary.txt226
-rw-r--r--tests/test_expr.c191
6 files changed, 191 insertions, 28015 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 ed838106b..000000000
--- a/.version
+++ /dev/null
@@ -1 +0,0 @@
-1.6.2.15-rc1
diff --git a/ChangeLog b/ChangeLog
deleted file mode 100644
index f34c3b39f..000000000
--- a/ChangeLog
+++ /dev/null
@@ -1,27611 +0,0 @@
-2010-11-15 Leif Madsen <lmadsen@digium.com>
-
- * Asterisk 1.6.2.15-rc1 Released.
-
-2010-11-15 07:42 +0000 [r294988] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_curl.c: It is possible to crash Asterisk by feeding
- the curl engine invalid data. (closes issue #18161) Reported by:
- wdoekes Patches: 20101029__issue18161.diff.txt uploaded by
- tilghman (license 14) Tested by: tilghman
-
-2010-11-12 21:14 +0000 [r294904-294910] Jeff Peeler <jpeeler@digium.com>
-
- * apps/app_voicemail.c: Return correct error code if lock path
- fails. The recent changes to open_mailbox actually caused it to
- be fixed, but let's be consistent. Reported by alecdavis in
- asterisk-dev.
-
- * apps/app_voicemail.c, /: Merged revisions 294903 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r294903 | jpeeler | 2010-11-12 14:49:09 -0600 (Fri, 12
- Nov 2010) | 16 lines Fix regression causing abort in voicemail
- after opening a mailbox with no mesgs. In order to be more safe,
- some error handling code was changed to respect more error
- conditions including the potential memory allocation failure for
- deleted and heard message tracking introduced in 293004. However,
- last_message_index returns -1 for zero messages (perhaps as
- expected) and was triggering the stricter error checking. Because
- last_message_index is only called directly in one place, just
- return 0 from open_mailbox (for file based storage) when no
- messages are detected unless a real error has occurred. (closes
- issue #18240) Reported by: leobrown Patches:
- bug18240.1-6-2.diff.txt uploaded by alecdavis (license 585)
- Tested by: pabelanger ........
-
-2010-11-12 02:44 +0000 [r294822] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 294821 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r294821 | rmudgett | 2010-11-11 20:41:13 -0600 (Thu, 11
- Nov 2010) | 11 lines Asterisk is getting a "No D-channels
- available!" warning message every 4 seconds. Asterisk is just
- whining too much with this message: "No D-channels available!
- Using Primary channel XXX as D-channel anyway!". Filtered the
- message so it only comes out once if there is no D channel
- available without an intervening D channel available period.
- (closes issue #17270) Reported by: jmls ........
-
-2010-11-11 21:57 +0000 [r294639-294733] Jeff Peeler <jpeeler@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 294688 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r294688 | jpeeler | 2010-11-11 15:12:27 -0600 (Thu, 11 Nov 2010)
- | 18 lines Fix problem with qualify option packets for realtime
- peers never stopping. The option packets not only never stopped,
- but if a realtime peer was not in the peer list multiple options
- dialogs could accumulate over time. This scenario has the
- potential to progress to the point of saturating a link just from
- options packets. The fix was to ensure that the poke scheduler
- checks to see if a peer is in the peer list before continuing to
- poke. The reason a peer must be in the peer list to be able to
- properly manage an options dialog is because otherwise the call
- pointer is lost when the peer is regenerated from the database,
- which is how existing qualify dialogs are detected. (closes issue
- #16382) (closes issue #17779) Reported by: lftsy Patches:
- bug16382-3.patch uploaded by jpeeler (license 325) Tested by:
- zerohalo ........
-
- * main/asterisk.c, include/asterisk.h, main/pbx.c, /: Merged
- revisions 294384 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r294384 | jpeeler | 2010-11-09 11:37:59 -0600 (Tue, 09 Nov 2010)
- | 47 lines Fix a deadlock in device state change processing.
- Copied from some notes from the original author (Russell):
- Deadlock scenario: Thread 1: device state change thread Holds -
- rdlock on contexts Holds - hints lock Waiting on channels
- container lock Thread 2: SIP monitor thread Holds the "iflock"
- Holds a sip_pvt lock Holds channel container lock Waiting for a
- channel lock Thread 3: A channel thread (chan_local in this case)
- Holds 2 channel locks acquired within app_dial Holds a 3rd
- channel lock it got inside of chan_local Holds a local_pvt lock
- Waiting on a rdlock of the contexts lock A bunch of other threads
- waiting on a wrlock of the contexts lock To address this
- deadlock, some locking order rules must be put in place and
- enforced. Existing relevant rules: 1) channel lock before a pvt
- lock 2) contexts lock before hints lock 3) channels container
- before a channel What's missing is some enforcement of the order
- when you involve more than any two. To fix this problem, I put in
- some code that ensures that (at least in the code paths involved
- in this bug) the locks in (3) come before the locks in (2). To
- change the operation of thread 1 to comply, I converted the
- storage of hints to an astobj2 container. This allows processing
- of hints without holding the hints container lock. So, in the
- code path that led to thread 1's state, it no longer holds either
- the contexts or hints lock while it attempts to lock the channels
- container. (closes issue #18165) Reported by: antonio ABE-2583
- ........
-
-2010-11-10 23:16 +0000 [r294571] Tilghman Lesher <tlesher@digium.com>
-
- * main/features.c: Actually pay attention to documented settings in
- features.conf. (closes issue #16757) Reported by: voxter Patches:
- 20101012__issue16757.diff.txt uploaded by tilghman (license 14)
- Review: https://reviewboard.asterisk.org/r/994/
-
-2010-11-10 12:41 +0000 [r294500] Russell Bryant <russell@digium.com>
-
- * main/devicestate.c: Improve a debug message to be more readable
- and consistent. (closes issue #18282) Reported by: klaus3000
- Patches: ast_devstate2str-patch.txt uploaded by klaus3000
- (license 65)
-
-2010-11-09 20:27 +0000 [r294429] Tilghman Lesher <tlesher@digium.com>
-
- * configure, configure.ac: Detect GMime properly on systems where
- gmime flags and libs are configured with pkg-config. (closes
- issue #16155) Reported by: jcollie Patches:
- 20100917__issue16155.diff.txt uploaded by tilghman (license 14)
- Tested by: tilghman
-
-2010-11-08 22:30 +0000 [r294277-294312] Jeff Peeler <jpeeler@digium.com>
-
- * res/res_timing_timerfd.c: add missing unlock not present in
- 294277
-
- * main/timing.c, main/channel.c, res/res_timing_timerfd.c,
- include/asterisk/timing.h: Fix playback failure when using IAX
- with the timerfd module. To fix this issue the alert pipe will
- now be used when the timerfd module is in use. There appeared to
- be a race that was not solved by adding locking in the timerfd
- module, but needed to be there anyway. The race was between the
- timer being put in non-continuous mode in ast_read on the channel
- thread and the IAX frame scheduler queuing a frame which would
- enable continuous mode before the non-continuous mode event was
- read. This race for now is simply avoided. (closes issue #18110)
- Reported by: tpanton Tested by: tpanton I put tested by tpanton
- because it was tested on his hardware. Thanks for the remote
- access to debug this issue!
-
-2010-11-08 20:50 +0000 [r294242] Matthew Nicholson <mnicholson@digium.com>
-
- * channels/chan_sip.c: Go off hold when we get an empty reinvite
- telling us to. (closes issue 0014448) Reported by: frawd (closes
- issue #17878) Reported by: frawd
-
-2010-11-05 00:06 +0000 [r293969] Shaun Ruffell <sruffell@digium.com>
-
- * codecs/codec_dahdi.c, /: Merged revisions 293968 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r293968 | sruffell | 2010-11-04 19:02:53 -0500 (Thu, 04
- Nov 2010) | 17 lines codecs/codec_dahdi: Prevent "choppy" audio
- when receiving unexpected frame sizes. dahdi-linux 2.4.0
- (specifically commit 9034) added the capability for the wctc4xxp
- to return more than a single packet of data in response to a
- read. However, when decoding packets, codec_dahdi was still
- assuming that the default number of samples was in each read. In
- other words, each packet your provider sent you, regardless of
- size, would result in 20 ms of decoded data (30 ms if decoding
- G723). If your provider was sending 60 ms packets then
- codec_dahdi would end up stripping 40 ms of data from each
- transcoded frame resulting in "choppy" audio. This would only
- affect systems where G729 packets are arriving in sizes greater
- than 20ms or G723 packets arriving in sizes greater than 30ms.
- DAHDI-744. ........
-
-2010-11-03 18:31 +0000 [r293806] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 293805 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r293805 | rmudgett | 2010-11-03 13:23:04 -0500 (Wed, 03
- Nov 2010) | 20 lines Party A in an analog 3-way call would
- continue to hear ringback after party C answers. All parties are
- analog FXS ports. 1) A calls B. 2) A flash hooks to call C. 3) A
- flash hooks to bring C into 3-way call before C answers. (A and B
- hear ringback) 4) C answers 5) A continues to hear ringback
- during the 3-way call. (All parties can hear each other.) * Fixed
- use of wrong variable in dahdi_bridge() that stopped ringback on
- the wrong subchannel. * Made several debug messages have more
- information. A similar issue happens if B and C are SIP channels.
- B continues to hear ringback. For some reason this only affects
- v1.8 and trunk. * Don't start ringback on the real and 3-way
- subchannels when creating the 3-way conference. Removing this
- code is benign on v1.6.2 and earlier. ........
-
-2010-11-02 23:07 +0000 [r293723] Jeff Peeler <jpeeler@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 293722 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r293722 | jpeeler | 2010-11-02 18:02:51 -0500 (Tue, 02 Nov 2010)
- | 8 lines Add enabled/disabled information for rtautoclear sip
- show settings output. When setting to zero/"no", the numeric
- default was shown making it not obvious the disabled setting was
- respected. (closes issue #18123) Reported by: zerohalo ........
-
-2010-11-02 21:26 +0000 [r293647] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 293639 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r293639 | rmudgett | 2010-11-02 16:24:13 -0500 (Tue, 02
- Nov 2010) | 6 lines Make warning message have more useful
- information in it. Change "Unable to get index, and nullok is not
- asserted" to "Unable to get index for '<channel-name>' on channel
- <number> (<function>(), line <number>)". ........
-
-2010-11-02 Leif Madsen <lmadsen@digium.com>
-
- * Asterisk 1.6.2.14 Released.
-
-2010-09-20 Leif Madsen <lmadsen@digium.com>
-
- * Asterisk 1.6.2.14-rc1 Released.
-
-2010-09-20 15:56 +0000 [r287556-287558] Matthew Nicholson <mnicholson@digium.com>
-
- * main/pbx.c, /: Use ast_str when processing hint state changes
- Merged revisions 287555 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r287555 | mnicholson | 2010-09-20 10:48:14 -0500 (Mon, 20 Sep
- 2010) | 5 lines Use ast_dynamic_str when processing hint state
- changes (related to issue #17928) Reported by: mdu113 ........
-
- * /: Revert r287556.
-
- * /: Use ast_str when processing hint state changes Merged
- revisions 287555 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r287555 | mnicholson | 2010-09-20 10:48:14 -0500 (Mon, 20 Sep
- 2010) | 5 lines Use ast_dynamic_str when processing hint state
- changes (related to issue #17928) Reported by: mdu113 ........
-
-2010-09-19 16:06 +0000 [r287470] Olle Johansson <oej@edvina.net>
-
- * main/manager.c, /: Merged revisions 287469 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r287469 | oej | 2010-09-19 17:56:50 +0200 (Sön, 19 Sep 2010) | 7
- lines Make sure we always free variables properly in manager
- originate. (closes issue #17891) reported, solved and tested by
- oej Review: https://reviewboard.asterisk.org/r/869/ ........
-
-2010-09-17 21:08 +0000 [r287387] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_queue.c, /: Merged revisions 287386 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r287386 | tilghman | 2010-09-17 16:06:03 -0500 (Fri, 17 Sep 2010)
- | 7 lines Blank columns should get set on reload, not ignored.
- (closes issue #16893) Reported by: haakon Patches:
- 20100818__issue16893.diff.txt uploaded by tilghman (license 14)
- ........
-
-2010-09-17 13:36 +0000 [r287308] Matthew Nicholson <mnicholson@digium.com>
-
- * main/pbx.c, /: Merged revisions 287307 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r287307 | mnicholson | 2010-09-17 08:34:34 -0500 (Fri, 17 Sep
- 2010) | 5 lines Use ast_strdup() instead of ast_strdupa() while
- processing in ast_hint_state_changed(). (related to issue #17928)
- Reported by: mdu113 ........
-
-2010-09-16 22:12 +0000 [r287198] Jason Parker <jparker@digium.com>
-
- * contrib/init.d/rc.debian.asterisk, /: Merged revisions 287197 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r287197 | qwell | 2010-09-16 17:12:30 -0500 (Thu, 16 Sep 2010) |
- 7 lines Add LSB headers for Debian init script, since Debian will
- complain if it isn't there. Headers were taken from trunk.
- (closes issue #17958) Reported by: javyer ........
-
-2010-09-16 20:06 +0000 [r287115-287119] Matthew Nicholson <mnicholson@digium.com>
-
- * main/pbx.c, /: Merged revisions 287118 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r287118 | mnicholson | 2010-09-16 15:04:46 -0500 (Thu, 16 Sep
- 2010) | 8 lines Don't limit hint processing in
- ast_hint_state_changed() to AST_MAX_EXTENSION length strings.
- (closes issue #17928) Reported by: mdu113 Patches:
- 20100831__issue17928.diff.txt uploaded by tilghman (license 14)
- Tested by: mdu113 ........
-
- * main/cdr.c, /: Merged revisions 287114 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r287114 | mnicholson | 2010-09-16 14:52:39 -0500 (Thu, 16 Sep
- 2010) | 8 lines Don't stop printing cdr variables if we encounter
- one with a blank name or value. (closes issue #17900) Reported
- by: under Patches: core-show-channel-cdr-fix1.diff uploaded by
- mnicholson (license 96) Tested by: mnicholson ........
-
-2010-09-15 20:28 +0000 [r286998] Jeff Peeler <jpeeler@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 286941 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r286941 | jpeeler | 2010-09-15 15:08:52 -0500 (Wed, 15
- Sep 2010) | 7 lines Ensure mailbox is not filled to capacity
- before doing message forwarding. Specifically, before prompting
- to record a prepended message the capacity is checked first. If
- the mailbox is full the extension will be reprompted. ABE-2517
- ........
-
-2010-09-14 19:27 +0000 [r286681-286757] Matthew Nicholson <mnicholson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 286756 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r286756 | mnicholson | 2010-09-14 14:26:18 -0500 (Tue, 14 Sep
- 2010) | 13 lines Don't clear the username from a realtime
- database when a registration expires. Non-realtime chan_sip does
- not clear the username from memory when a registration expiries
- so realtime probably shouldn't either. (closes issue #17551)
- Reported by: ricardolandim Patches:
- reg-expiry-username-1.4-fix1.diff uploaded by mnicholson (license
- 96) reg-expiry-username-1.6.2-fix1.diff uploaded by mnicholson
- (license 96) reg-expiry-username-1.8-fix1.diff uploaded by
- mnicholson (license 96) reg-expiry-username-trunk-fix1.diff
- uploaded by mnicholson (license 96) Tested by: ricardolandim,
- mnicholson ........
-
- * main/channel.c, /: Merged revisions 286679 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r286679 | mnicholson | 2010-09-14 13:00:01 -0500 (Tue, 14 Sep
- 2010) | 7 lines Only drop duplicate answer frames if the channel
- is bridged. Back in r3710 ast_read() was modified to drop answer
- frames on channels that were in the UP state. This modification
- prevented bridges that were up before the answer from being
- broken and reestablished by an ANSWER control frame. That change
- also prevents pickup of channels called from the ast_dial
- framework from working properly. The ast_dial framework expects
- to see an ANSWER frame after dialing and the pickup code queues
- one but ast_read() drops it. This new change only drops ANSWER
- frames when the channel is bridged, allowing the answer queued by
- the pickup code to properly pass through ast_read() on to the
- ast_dial framework. ABE-2473 (related to issue #2342) ........
-
-2010-09-14 05:06 +0000 [r286527-286587] Tilghman Lesher <tlesher@digium.com>
-
- * contrib/realtime/mysql/voicemail_messages.sql (added),
- contrib/realtime/mysql/voicemail_data.sql (added): Add
- documentation on missing backend tables for Voicemail
-
- * main/features.c: C precedence got me
-
- * main/features.c: Refactor conversion to ast_poll() to fix
- callparking regression.
-
-2010-09-13 19:38 +0000 [r286456] Jason Parker <jparker@digium.com>
-
- * channels/chan_sip.c: Remove "Internal IP" from sip show settings,
- as it's not at all useful to display. (closes issue #17840)
- Reported by: oej
-
-2010-09-11 17:05 +0000 [r286268] Olle Johansson <oej@edvina.net>
-
- * /, main/file.c: Merged revisions 286267 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r286267 | oej | 2010-09-11 18:59:20 +0200 (Lör, 11 Sep 2010) | 4
- lines Handle error response when we can't make file compatible
- Review: https://reviewboard.asterisk.org/r/911/ ........
-
-2010-09-10 22:56 +0000 [r286223] Terry Wilson <twilson@digium.com>
-
- * channels/chan_local.c, /: Merged revisions 286222 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r286222 | twilson | 2010-09-10 17:54:23 -0500 (Fri, 10
- Sep 2010) | 1 line Return -1 if chan_local doesn't support an
- option ........
-
-2010-09-10 20:55 +0000 [r286117] Paul Belanger <paul.belanger@polybeacon.com>
-
- * channels/chan_iax2.c, /: Merged revisions 286114 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r286114 | pabelanger | 2010-09-10 16:35:08 -0400 (Fri,
- 10 Sep 2010) | 4 lines Load iax.conf before registering any
- functions/applications/actions. Review:
- https://reviewboard.asterisk.org/r/914/ ........
-
-2010-09-10 20:42 +0000 [r286116] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 286113 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r286113 | rmudgett | 2010-09-10 15:33:16 -0500 (Fri, 10
- Sep 2010) | 11 lines An outgoing call may not get hung up if a
- pre-connect incoming ISDN call is disconnected. If the ISDN link
- a pre-connect incoming call is using fails or is reset, the
- outgoing leg may not hang up or be delayed in hanging up.
- (Causes: PRI_CAUSE_NETWORK_OUT_OF_ORDER,
- PRI_CAUSE_DESTINATION_OUT_OF_ORDER, and
- PRI_CAUSE_NORMAL_TEMPORARY_FAILURE.) Just hang up the call if the
- incoming call leg hangs up before connecting for any reason. It
- makes no sense to send a BUSY or CONGESTION control frame to the
- outgoing call leg under these circumstances. ........
-
-2010-09-10 20:35 +0000 [r286115] Terry Wilson <twilson@digium.com>
-
- * include/asterisk/pbx.h, include/asterisk/frame.h,
- channels/chan_local.c, /, funcs/func_channel.c,
- include/asterisk/channel.h: Merged revisions 286059 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r286059 | twilson | 2010-09-10 14:25:08 -0500 (Fri, 10
- Sep 2010) | 16 lines Inherit CHANNEL() writes to both sides of a
- Local channel Having Local (/n) channels as queue members and
- setting the language in the extension with
- Set(CHANNEL(language)=fr) sets the language on the Local/...,2
- channel. Hold time report playbacks happen on the Local/...,1
- channel and therefor do not play in the specified language. This
- patch modifies func_channel_write to call the setoption callback
- and pass the CHANNEL() write info to the callback. chan_local
- uses this information to look up the other side of the channel
- and apply the same changes to it. (closes issue #17673) Reported
- by: Guggemand Review: https://reviewboard.asterisk.org/r/903/
- ........
-
-2010-09-10 18:30 +0000 [r285930-286024] Tilghman Lesher <tlesher@digium.com>
-
- * tests/test_heap.c, /, main/test.c: Merged revisions 286023 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r286023 | tilghman | 2010-09-10 13:22:04 -0500 (Fri, 10 Sep 2010)
- | 2 lines Missing newline ........
-
- * include/asterisk/select.h: Another fix for Mac OS X. While trying
- to fix this the "right" way, I wandered into dependency hell. Two
- hours later, I backed out, and just removed the offending code.
- ast_inline_api only goes one level deep and then it breaks. Ouch.
-
- * tests/test_poll.c, include/asterisk/select.h, /, configure,
- include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
- 285889 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r285889 | tilghman | 2010-09-09 19:13:45 -0500 (Thu, 09 Sep 2010)
- | 7 lines Fix Mac OS X build. This also fixes a rather grievous
- calculation error for the offset of ast_fdset, which was masked
- on Linux and FreeBSD, because these platforms check the first 256
- FDs regardless of the bitmask setting (due to backwards
- compatibility). ........
-
-2010-09-09 22:49 +0000 [r285818] Paul Belanger <paul.belanger@polybeacon.com>
-
- * /, codecs/gsm/Makefile: Merged revisions 285817 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r285817 | pabelanger | 2010-09-09 18:34:35 -0400 (Thu, 09 Sep
- 2010) | 8 lines GCC 4.2.x optimizations result in improper
- behavior of GSM codec (closes issue #17688) Reported by:
- pprindeville Patches: asterisk-trunk-bugid11243.patch uploaded by
- pprindeville (license 347) Tested by: mkeuter, pprindeville
- ........
-
-2010-09-09 20:09 +0000 [r285744] Jason Parker <jparker@digium.com>
-
- * main/channel.c, /: Merged revisions 285742 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r285742 | qwell | 2010-09-09 15:06:31 -0500 (Thu, 09 Sep 2010) |
- 9 lines Transmit silence when reading DTMF in ast_readstring.
- Otherwise, you could get issues with DTMF timeouts causing
- hangups. (closes issue #17370) Reported by: makoto Patches:
- channel-readstring-silence-generator.patch uploaded by makoto
- (license 38) ........
-
-2010-09-09 18:50 +0000 [r285639-285710] Brett Bryant <bbryant@digium.com>
-
- * main/pbx.c: Fixes an issue with dialplan pattern matching where
- the specificity for pattern ranges and pattern special characters
- was inconsistent. (closes issue #16903) Reported by: Nick_Lewis
- Patches: pbx.c-specificity.patch uploaded by Nick Lewis (license
- 657) Tested by: Nick_Lewis
-
- * res/res_musiconhold.c, /: Merged revisions 285638 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r285638 | bbryant | 2010-09-09 13:20:17 -0400 (Thu, 09
- Sep 2010) | 7 lines Fixes an issue with MOH where it doesn't
- recover cleanly when it can't play a file and would just stop,
- instead of continuing to find the next playable file in the MOH
- class. (closes issue #17807) Reported by: kshumard Review:
- https://reviewboard.asterisk.org/r/910/ ........
-
-2010-09-08 22:11 +0000 [r285563-285567] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 285566 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r285566 | dvossel | 2010-09-08 17:07:31 -0500 (Wed, 08 Sep 2010)
- | 2 lines In retrans_pkt, do not unlock pvt until the end of the
- function on a transmit failure. ........
-
- * channels/chan_sip.c: Fixes interoperability problems with session
- timer behavior in Asterisk. CHANGES: 1. Never put "timer" in
- "Require" header. This is not to our benefit and RFC 4028 section
- 7.1 even warns against it. It is possible for one endpoint to
- perform session-timer refreshes while the other endpoint does not
- support them. If in this case the end point performing the
- refreshing puts "timer" in the Require field during a refresh,
- the dialog will likely get terminated by the other end. 2. Change
- the behavior of 'session-timer=accept' in sip.conf (which is the
- default behavior of Asterisk with no session timer configuration
- specified) to only run session-timers as result of an incoming
- INVITE request if the INVITE contains an "Session-Expires"
- header... Asterisk is currently treating having the "timer"
- option in the "Supported" header as a request for session timers
- by the UAC. I do not agree with this. Session timers should only
- be negotiated in "accept" mode when the incoming INVITE supplies
- a "Session-Expires" header, otherwise RFC 4028 says we should
- treat a request containing no "Session-Expires" header as a
- session with no expiration. Below I have outlined some situations
- and what Asterisk's behavior is. The table reflects the behavior
- changes implemented by this patch. SITUATIONS: -Asterisk as UAS
- 1. Incoming INVITE: NO "Session-Expires" 2. Incoming INVITE: HAS
- "Session-Expires" -Asterisk as UAC 3. Outgoing INVITE: NO
- "Session-Expires". 200 Ok Response HAS "Session-Expires" header
- 4. Outgoing INVITE: NO "Session-Expires". 200 Ok Response NO
- "Session-Expires" header 5. Outgoing INVITE: HAS
- "Session-Expires". Active - Asterisk will have an active refresh
- timer regardless if the other endpoint does. Inactive - Asterisk
- does not have an active refresh timer regardless if the other
- endpoint does. XXXXXXX - Not possible for mode.
- ______________________________________ |SITUATIONS |
- 'session-timer' MODES | |___________|________________________| |
- | originate | accept | |-----------|------------|-----------| |1.
- | Active | Inactive | |2. | Active | Active | |3. | XXXXXXXX |
- Active | |4. | XXXXXXXX | Inactive | |5. | Active | XXXXXXXX |
- -------------------------------------- (closes issue #17005)
- Reported by: alexrecarey
-
-2010-09-08 20:56 +0000 [r285532] Brett Bryant <bbryant@digium.com>
-
- * apps/app_meetme.c: Fixes a bug with MeetMe where after announcing
- the amount of time left in a conference, if music on hold was
- playing, it doesn't restart. (closes issue #17408) Reported by:
- sysreq Patches: asterisk-issue-17408_fixed.patch uploaded by
- sysreq (license 1009) Tested by: sysreq
-
-2010-09-08 20:42 +0000 [r285526-285529] Jason Parker <jparker@digium.com>
-
- * res/res_musiconhold.c, include/asterisk/astobj2.h: Follow coding
- guidelines in moh rescan fix. Also fix the documentation that got
- me in trouble.
-
- * res/res_musiconhold.c: Fixes issue where moh files were no longer
- rescanned during a reload. (closes issue #16744) Reported by: pj
- Patches: 16744-reload.diff uploaded by qwell (license 4) Tested
- by: qwell
-
-2010-09-07 20:31 +0000 [r285267-285366] Tilghman Lesher <tlesher@digium.com>
-
- * pbx/pbx_config.c, /: Merged revisions 285365 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r285365 | tilghman | 2010-09-07 15:30:22 -0500 (Tue, 07 Sep 2010)
- | 9 lines Catch invalid extensions at the parser, instead of
- making the core deal with them. (closes issue #17794) Reported
- by: PavelL Patches: 20100820__issue17794__1.6.2.diff.txt uploaded
- by tilghman (license 14) 20100820__issue17794__1.4.diff.txt
- uploaded by tilghman (license 14) Tested by: PavelL ........
-
- * main/poll.c, /: Merged revisions 285266 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r285266 | tilghman | 2010-09-07 14:04:50 -0500 (Tue, 07 Sep 2010)
- | 4 lines Use poll, if indicated to do so, in the ast_poll2
- implementation. This fixes the unit tests on FreeBSD 8.0.
- ........
-
-2010-09-07 17:49 +0000 [r285196] Brett Bryant <bbryant@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 285194 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r285194 | bbryant | 2010-09-07 13:45:41 -0400 (Tue, 07
- Sep 2010) | 10 lines Fixes voicemail.conf issues where mailboxes
- with passwords that don't precede a comma would throw unnecessary
- error messages. (closes issue #15726) Reported by: 298 Patches:
- M15726.diff uploaded by junky (license 177) Tested by: junky
- Review: [full review board URL with trailing slash] ........
-
-2010-09-06 06:55 +0000 [r285089] Tilghman Lesher <tlesher@digium.com>
-
- * makeopts.in, /, BSDmakefile (added): Merged revisions 285088 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r285088 | tilghman | 2010-09-06 01:54:18 -0500 (Mon, 06 Sep 2010)
- | 2 lines Silly convenience script for BSD platforms. ........
-
-2010-09-03 18:15 +0000 [r284958] Brett Bryant <bbryant@digium.com>
-
- * channels/chan_iax2.c: This is a patch provided for issue #17935
- to add the ActionID to the IAXregistry AMI response. (closes
- issue #17935) Reported by: alexkuklin Patches: iaxshowreg
- uploaded by alexkuklin (license 1115) Tested by: alexkuklin
-
-2010-09-03 16:20 +0000 [r284897] Terry Wilson <twilson@digium.com>
-
- * apps/app_chanspy.c, /: Merged revisions 284881 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r284881 | twilson | 2010-09-03 11:10:23 -0500 (Fri, 03 Sep 2010)
- | 5 lines Properly detect when a sound file doesn't exist
- ast_fileexists returns -1 for error and 0 for a non-existant
- file. The existing code treated missing files as though they
- existed. ........
-
-2010-09-02 20:54 +0000 [r284778] Brett Bryant <bbryant@digium.com>
-
- * main/manager.c, /: Merged revisions 284777 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r284777 | bbryant | 2010-09-02 16:25:03 -0400 (Thu, 02 Sep 2010)
- | 7 lines Fixes a bug in manager.c where the default
- configuration values weren't reset when the manager configuration
- was reloaded. (closes issue #17917) Reported by: lmadsen Review:
- https://reviewboard.asterisk.org/r/883/ ........
-
-2010-09-02 16:48 +0000 [r284704] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 284703 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r284703 | dvossel | 2010-09-02 11:47:15 -0500 (Thu, 02 Sep 2010)
- | 7 lines Removed relatedpeer code from sip_autodestruct Handling
- of the relatedpeer structure associated with a sip_pvt should be
- done during the final sip_destruction function, not in
- sip_autodestruct. ........
-
-2010-09-02 16:07 +0000 [r284399-284665] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_usbradio.c: Fixing build.
-
- * apps/app_queue.c: Don't reset queue stats on a module reload.
- (closes issue #17535) Reported by: raarts Patches:
- 20100819__issue17535.diff.txt uploaded by tilghman (license 14)
-
- * configure, include/asterisk/autoconfig.h.in: Failed to rerun
- bootstrap.sh after last commit
-
- * res/res_jabber.c, main/rtp.c, main/poll.c,
- include/asterisk/select.h (added), channels/chan_usbradio.c,
- channels/chan_phone.c, channels/chan_misdn.c, main/features.c,
- include/asterisk/poll-compat.h, tests/test_poll.c (added),
- main/asterisk.c, utils/clicompat.c, res/res_ais.c, /,
- configure.ac, channels/console_video.c,
- include/asterisk/channel.h: Merged revisions 284478 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r284478 | tilghman | 2010-09-01 13:49:11 -0500 (Wed, 01
- Sep 2010) | 11 lines Ensure that all areas that previously used
- select(2) now use poll(2), with implementations that need poll(2)
- implemented with select(2) safe against 1024-bit overflows. This
- is a followup to the fix for the pthread timer in 1.6.2 and
- beyond, fixing a potential crash bug in all supported releases.
- (closes issue #17678) Reported by: russell Branch:
- https://origsvn.digium.com/svn/asterisk/team/tilghman/ast_select
- Review: https://reviewboard.asterisk.org/r/824/ ........
-
- * res/res_config_pgsql.c: Don't warn on floats and timestamps
- (closes issue #17082) Reported by: coolmig
-
- * /, channels/chan_sip.c: Merged revisions 284393 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r284393 | tilghman | 2010-08-31 15:13:21 -0500 (Tue, 31 Aug 2010)
- | 7 lines Don't send a devstate change on poke_noanswer if the
- state did not change. (closes issue #17741) Reported by: schmidts
- Patches: chan_sip.c.patch uploaded by schmidts (license 1077)
- ........
-
-2010-08-31 18:59 +0000 [r284317] Leif Madsen <lmadsen@digium.com>
-
- * configs/say.conf.sample, /: Merged revisions 284316 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r284316 | lmadsen | 2010-08-31 13:57:59 -0500 (Tue, 31
- Aug 2010) | 7 lines Update say.conf.sample to match the rules in
- say.c (closes issue #17835) Reported by: RoadKill Patches:
- say.conf.sample.patch.rules uploaded by RoadKill (license 933)
- Tested by: RoadKill ........
-
-2010-08-30 22:27 +0000 [r284280] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_festival.c: Fix 3 coding errors: 1) After we close FD,
- we should not be trying to write to it. 2) Call _exit(0), not
- exit(0), to avoid running shutdown routines in a child. 3) Use
- endian, not processor, detection to ensure bytes are written in
- the correct order. (closes issue #15706) Reported by: modelnine
- Patches: asterisk-1.6.1.1-festival-debug.patch uploaded by
- modelnine (license 865) Tested by: gmartinez
-
-2010-08-27 22:27 +0000 [r284002] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 283960 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r283960 | dvossel | 2010-08-27 17:17:26 -0500 (Fri, 27 Aug 2010)
- | 8 lines Parse all "Accept" headers for SIP SUBSCRIBE requests.
- (closes issue #17758) Reported by: ibc Patches:
- multiple_accept_headers_1.4.diff uploaded by dvossel (license
- 671) ........
-
-2010-08-27 20:30 +0000 [r283881] Jason Parker <jparker@digium.com>
-
- * res/res_config_pgsql.c, res/res_config_odbc.c, /: Merged
- revisions 283880 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r283880 | qwell | 2010-08-27 15:29:11 -0500 (Fri, 27 Aug 2010) |
- 8 lines Fix issue with decoding ^-escaped characters in realtime.
- (closes issue #17790) Reported by: denzs Patches:
- 17790-chunky.diff uploaded by qwell (license 4) Tested by: qwell,
- denzs ........
-
-2010-08-26 15:24 +0000 [r283381-283691] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 283690 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r283690 | dvossel | 2010-08-26 10:22:28 -0500 (Thu, 26 Aug 2010)
- | 19 lines Fixed how Asterisk destroys a dialog on channel hangup
- before invite receives a response. If an ast_channel with a SIP
- tech pvt hangs up before the sip dialog gets a response to its
- outgoing INVITE, Asterisk used to pretend_ack the INVITE. This is
- not rfc compliant and results in confusion at the other endpoint.
- sip_pretend_ack will ack and remove all the packets in the
- retransmit queue. This means that the INVITE will stop
- retransmitting, and that any response to that INVITE that comes
- after the pretend_ack occurs will be ignored. Instead of faking
- any sort of acknowledgement for an outgoing INVITE during an
- internal hangup, we should let the protocol stack process the
- INVITE transaction and terminate the dialog properly. This is
- achieved by setting the PENDING_BYE flag. When this flag is used,
- once the dialog proceeds to an escapable state the transaction
- will either be canceled with a SIP_CANCEL or completed followed
- immediately by a BYE. Attempting to do this any other way is
- incorrect. If the endpoint is not responding to the INVITE
- request, the INVITE must continue to be retransmitted until it
- times out which will result in the dialog being destroyed.
- ........
-
- * channels/chan_sip.c: Add to and from tags to NOTIFY dialog-info
- xml body so pickup can occur. When pedantic mode is used, the
- dialog-info xml generated during a ringing event must contain the
- to and from tag values. Otherwise if a pickup occurs using INVITE
- with replaces, Astrisk will not be able to locate the
- subscription.
-
- * channels/chan_sip.c: Asterisk will not advertise session timers
- are supported when 'session-timers=refuse' is used. Asterisk now
- dynamically builds the "Supported" header depending on what is
- enabled/disabled in sip.conf. Session timers used to always be
- advertised as being supported even when they were disabled in the
- configuration. This caused problems with some end points. (issue
- #17005)
-
- * /, channels/chan_sip.c: Merged revisions 283380 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r283380 | dvossel | 2010-08-24 11:01:51 -0500 (Tue, 24 Aug 2010)
- | 11 lines This fix makes sure the ast_channel hangs up correctly
- when the dialog's PENDING_BYE flag is set. When the pending bye
- flag is used, it is possible that the dialog will terminate and
- leave the sip_pvt->owner channel up. This is because we never
- hangup the ast_channel after sending the SIP_BYE request. When we
- receive the response for the SIP_BYE we set need_destroy which we
- would expect to destroy the dialog on the next do_monitor loop,
- but this is not the case. The dialog will only be destroyed once
- the owner is hungup even with the need_destroy flag set. This
- patch sets the softhangup flag on the ast_channel when a SIP_BYE
- request is sent as a result of the pending bye flag. ........
-
-2010-08-23 21:32 +0000 [r283318] Tilghman Lesher <tlesher@digium.com>
-
- * cdr/cdr_odbc.c, cdr/cdr_adaptive_odbc.c: CDR drivers depend upon
- res_odbc, not directly on the ODBC libraries
-
-2010-09-15 Leif Madsen <lmadsen@digium.com>
-
- * Asterisk 1.6.2.13 released.
-
- * Incorrect .version and ChangeLog files updated. Re-release
- of Asterisk 1.6.2.12 with corrections and version
- number bump.
-
-2010-09-15 Leif Madsen <lmadsen@digium.com>
-
- * Asterisk 1.6.2.12 released.
-
-2010-08-23 Leif Madsen <lmadsen@digium.com>
-
- * Asterisk 1.6.2.12-rc1 Released.
-
-2010-08-20 16:48 +0000 [r283049-283124] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 283123 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ................ r283123 | rmudgett | 2010-08-20 11:46:22 -0500
- (Fri, 20 Aug 2010) | 9 lines Merged revision 278274 from
- https://origsvn.digium.com/svn/asterisk/trunk .......... r278274
- | rmudgett | 2010-07-20 17:38:13 -0500 (Tue, 20 Jul 2010) | 1
- line Reference correct struct member for unlikely event
- PRI_EVENT_CONFIG_ERR. .......... ................
-
- * channels/chan_dahdi.c, /: Merged revisions 283048 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r283048 | rmudgett | 2010-08-20 10:24:36 -0500 (Fri, 20
- Aug 2010) | 22 lines Q931 - Sending PROGRESS after sending
- ALERTING is a protocol error The PRI layer in chan_dadhi will
- check if a PROGRESS message has already been sent, and not allow
- sending another (although that is technically allowed by the Q931
- spec), however it does not protect against sending an ALERTING
- and then sending a PROGRESS message, which is a violation of the
- specification. Most switches don't seem to care too deeply about
- this, but some do, and will disconnect the call when receiving
- this invalid sequence. Protocol specification reference:
- T-REC-Q.931-199805-I page 223, "Figure A.5/Q.931 -- Overview
- protocol control (network side) point-point (sheet 3 of 8)"
- (closes issue #17874) Reported by: nic_bellamy Patches:
- asterisk-1.4-r282537_no-progress-after-alerting.patch uploaded by
- nic bellamy (license 299)
- asterisk-1.6.2-r282537_no-progress-after-alerting.patch uploaded
- by nic bellamy (license 299)
- asterisk-trunk-r282537_no-progress-after-alerting.patch uploaded
- by nic bellamy (license 299) ........
-
-2010-08-19 21:05 +0000 [r282890-282894] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 282893 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r282893 | dvossel | 2010-08-19 16:03:24 -0500 (Thu, 19 Aug 2010)
- | 11 lines tos_sip option was not being set correctly When
- tos_sip is used, the tos of the sip socket is only set correctly
- if the socket binding changes on a reload. If the binding stays
- the same but the TOS changes, the new tos value would not take
- into effect. This patch fixes that. (closes issue #17712)
- Reported by: nickb ........
-
- * channels/chan_sip.c: fixes sip peer memory leaks in the
- peer_by_ip table (issue #17798)
-
-2010-08-19 19:44 +0000 [r282859] Matthew Nicholson <mnicholson@digium.com>
-
- * channels/chan_sip.c: Merged revisions 277944 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r277944 | pabelanger | 2010-07-19 15:56:07 -0500 (Mon, 19 Jul
- 2010) | 16 lines Regression with T.38 negotiation Prior to
- 1.4.26.3 T.38 negotiation worked properly, in the case of the
- reporter. (issue #16852) Reported by: cfc (closes issue #16705)
- Reported by: mpiazzatnetbug Patches: issue16705_2.diff uploaded
- by ebroad (license 878) Tested by: vrban, ebroad, c0rnoTa,
- samdell3 Review: https://reviewboard.asterisk.org/r/754/ ........
-
-2010-08-19 02:14 +0000 [r282730] Terry Wilson <twilson@digium.com>
-
- * configs/sip.conf.sample, /: Merged revisions 282729 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r282729 | twilson | 2010-08-18 21:12:55 -0500 (Wed, 18
- Aug 2010) | 2 lines Add some documentation about codec
- negotiation to sip.conf ........
-
-2010-08-18 14:28 +0000 [r282668] David Vossel <dvossel@digium.com>
-
- * channels/chan_sip.c: fixes crash with notifycid (closes issue
- #17868) Reported by: francesco_r Patches: issue_17868.diff
- uploaded by dvossel (license 671) Tested by: francesco_r
-
-2010-08-18 07:43 +0000 [r282607] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_dahdi.c: Don't warn on callerid when completely
- text, instead of numeric with localdialplan prefixes. (closes
- issue #16770) Reported by: jamicque Patches:
- 20100413__issue16770.diff.txt uploaded by tilghman (license 14)
- 20100811__issue16770.diff.txt uploaded by tilghman (license 14)
- Tested by: jamicque
-
-2010-08-17 21:35 +0000 [r282576] David Vossel <dvossel@digium.com>
-
- * channels/chan_sip.c: fixes no default transport for temp peer
- creation in chan_sip (closes issue #17829) Reported by: falves11
- Patches: issue_17829.rev1.txt uploaded by russell (license 2)
- issue_17829.diff uploaded by dvossel (license 671) Tested by:
- falves11
-
-2010-08-16 18:00 +0000 [r282469] Leif Madsen <lmadsen@digium.com>
-
- * doc/tex/asterisk.tex, doc/tex/sounds.tex (added): Add information
- about creating sounds files using the sounds tools publically
- available so that others can create their own sounds prompts
- using the same tools we use to generate sounds releases. This
- allows people creating their own prompts to sound consistent with
- the prompts available from the open source project. SWP-595
-
-2010-08-16 17:32 +0000 [r282467] Terry Wilson <twilson@digium.com>
-
- * main/channel.c, /: Merged revisions 282430 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r282430 | twilson | 2010-08-16 12:06:37 -0500 (Mon, 16 Aug 2010)
- | 16 lines Send a SRCCHANGE indication when we masquerade
- Masquerading a channel means that the src of the audio is
- potentially changing, so send a SRCCHANGE so that RTP-based media
- streams can get a new SSRC generated to reflect the change.
- Original patch by addix (along with lots of testing--thanks!).
- (closes issue #17007) Reported by: addix Patches:
- 1001-reset-SSRC-original-channel.diff uploaded by addix (license
- 1006) srcchange.diff uploaded by twilson (license 396) Tested by:
- addix, twilson Review: https://reviewboard.asterisk.org/r/862/
- ........
-
-2010-08-13 18:54 +0000 [r282235] David Vossel <dvossel@digium.com>
-
- * channels/chan_sip.c: only do magic pickup when notifycid is
- enabled A new way of doing BLF pickup was introduced into 1.6.2.
- This feature adds a call-id value into the XML of a SIP_NOTIFY
- message sent to alert a subscriber that a device is ringing. This
- option should only be enabled when the new 'notifycid' option is
- set... but this was not the case. Instead the call-id value was
- included for every RINGING Notify message, which caused a
- regression for people who used other methods for call pickup.
- (closes issue #17633) Reported by: urosh Patches: chan_sip.txt
- uploaded by urosh (license ) blf_cid_issue.diff uploaded by
- dvossel (license 671) Tested by: dvossel, urosh, okrief,
- alecdavis
-
-2010-08-12 22:50 +0000 [r282130] Jason Parker <jparker@digium.com>
-
- * pbx/pbx_config.c, /: Merged revisions 282129 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r282129 | qwell | 2010-08-12 17:49:28 -0500 (Thu, 12 Aug 2010) |
- 1 line Register CLI commands before parsing config, in case there
- is a config error. ........
-
-2010-08-12 03:01 +0000 [r281912] Jeff Peeler <jpeeler@digium.com>
-
- * main/channel.c, /: Merged revisions 281911 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r281911 | jpeeler | 2010-08-11 22:00:14 -0500 (Wed, 11 Aug 2010)
- | 20 lines Ensure SSRC is changed when media source is changed to
- resolve audio delay. This change causes the SSRC to change right
- before the channels are bridged, which is what used to happen. It
- seems that fixes were made to attempt limiting SSRC changes,
- targeted mainly at sending DTMF. DTMF is not affecting the SSRC
- with this change. There are two other control frames sent in
- ast_channel_bridge that probably should also be changed to
- AST_CONTROL_SRCCHANGE as well, but I'm going to leave this change
- up to the discretion of resolving issue #17007. For reference -
- old review implementing new control frame SRCCHANGE:
- https://reviewboard.asterisk.org/r/540 (closes issue #17404)
- Reported by: sdolloff Patches: bug17404.patch uploaded by jpeeler
- (license 325) Tested by: sdolloff ........
-
-2010-08-11 21:09 +0000 [r281763-281873] Leif Madsen <lmadsen@digium.com>
-
- * configs/say.conf.sample, /: Merged revisions 281819 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r281819 | lmadsen | 2010-08-11 13:28:10 -0500 (Wed, 11
- Aug 2010) | 6 lines Add Danish support to say.conf.sample (closes
- issue #17836) Reported by: RoadKill Patches:
- say.conf.sample.patch.dk uploaded by RoadKill (license 933)
- ........
-
- * configs/say.conf.sample, /: Merged revisions 281762 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r281762 | lmadsen | 2010-08-11 12:51:40 -0500 (Wed, 11
- Aug 2010) | 6 lines Allow say.conf to handle large numbers ending
- with multiple zeros. (closes issue #17833) Reported by: RoadKill
- Patches: say.conf.sample.patch.largenumbers uploaded by RoadKill
- (license 933) ........
-
-2010-08-11 15:17 +0000 [r281722] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_readexten.c: Only set status TIMEOUT, if we have no
- digits. (closes issue #15188) Reported by: jcovert Patches:
- app_readexten.c.patch-1.6.2.8-rc1 uploaded by jcovert (license
- 551)
-
-2010-08-10 18:04 +0000 [r281567-281574] Russell Bryant <russell@digium.com>
-
- * main/sched.c: Don't move the time threshold for running scheduled
- events on every iteration. Instead, only calculate the time
- threshold each time ast_sched_runq() is called. (closes issue
- #17742) Reported by: schmidts Patches: sched.c.patch uploaded by
- schmidts (license 1077)
-
- * apps/app_dial.c, /: Merged revisions 281566 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r281566 | russell | 2010-08-10 12:45:45 -0500 (Tue, 10 Aug 2010)
- | 8 lines Reset visible indication after answer. (closes issue
- #17641) Reported by: klaus3000 Patches:
- ast1.6.2.9-app_dial-visible_indication.patch.txt uploaded by
- klaus3000 (license 65) Tested by: schmidts ........
-
-2010-08-09 20:46 +0000 [r281430] David Vossel <dvossel@digium.com>
-
- * channels/chan_sip.c: fixes SIP peers memory leak We zeroed out
- the peer's addr before it was removed from the peers_by_ip
- container. This made it impossible to be removed from the
- container as the addr is the key used by the container to find
- the peer. (closes issue #17774) Reported by: kkm Patches:
- 017774-sip-peer-leak-1.6.2.10.diff uploaded by kkm (license 888)
- 017774-sip-peer-leak-1.8.diff uploaded by kkm (license 888)
-
-2010-08-09 20:07 +0000 [r281391] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_local.c, /: Merged revisions 281390 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r281390 | jpeeler | 2010-08-09 15:04:30 -0500 (Mon, 09
- Aug 2010) | 13 lines Prevent loss of Caller ID information set on
- local channel after masquerade. Caller ID set on the channel
- before a masquerade occurs when using a local channel would cause
- the information to be lost. The problem was that the information
- was set on a channel destined to be hung up. The somewhat
- confusing fix is to detect if any Caller ID has been set on the
- channel and if so preswap the Caller ID data so that basically
- the masquerade puts the data back. (closes issue #17138) Reported
- by: kobaz Review: https://reviewboard.asterisk.org/r/847/
- ........
-
-2010-08-05 13:11 +0000 [r281051] Russell Bryant <russell@digium.com>
-
- * main/cdr.c: Cleanup default option value handling for cdr.conf
- [general]. The default values would differ depending on whether
- or not cdr.conf exists. That is no longer the case. Apply a
- default value to the unanswered option. Define all default values
- as named constants.
-
-2010-08-05 07:40 +0000 [r280983] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/pbx.h, main/pbx.c, /: Merged revisions 280982
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r280982 | tilghman | 2010-08-05 02:28:33 -0500 (Thu, 05 Aug 2010)
- | 8 lines Change context lock back to a mutex, because
- functionality depends upon the lock being recursive. (closes
- issue #17643) Reported by: zerohalo Patches:
- 20100726__issue17643.diff.txt uploaded by tilghman (license 14)
- Tested by: zerohalo ........
-
-2010-08-03 20:52 +0000 [r280671-280812] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_callerid.c, channels/chan_dahdi.c, /: Merged revisions
- 280811 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r280811 | tilghman | 2010-08-03 15:49:10 -0500 (Tue, 03 Aug 2010)
- | 9 lines Prevent DAHDI channels from overriding the callerid,
- once it's been set by the user. (closes issue #16661) Reported
- by: jstapleton Patches: 20100414__issue16661.diff.txt uploaded by
- tilghman (license 14) 20100415__issue16661__1.6.2.diff.txt
- uploaded by tilghman (license 14) Tested by: jstapleton ........
-
- * doc/asterisk.sgml, doc/asterisk.8, doc/Makefile (added): Document
- -B and -W flags and regenerate manpage from sgml
-
- * apps/app_voicemail.c: Allow the pipe, but also allow the comma
-
-2010-08-02 21:14 +0000 [r280669] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_sip.c: Change SIP NOTIFY requests to expect a
- response so authentication will work. This changes the request to
- be sent with the transmit type XMIT_RELIABLE so that sip_ack
- doesn't return false and cause the 401 to be ignored in cases
- where authentication is required. (closes issue #14255) Reported
- by: zktech
-
-2010-07-29 21:07 +0000 [r280556] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_config_curl.c: Off-by-one error (closes issue #17590)
- Reported by: atis Patches: 20100729__issue17590.diff.txt uploaded
- by tilghman (license 14)
-
-2010-07-29 20:42 +0000 [r280449-280551] David Vossel <dvossel@digium.com>
-
- * channels/chan_sip.c: fixes wrong SRV query for TLS connection
- (closes issue #17612) Reported by: marcelloceschia Patches:
- chan-sip_srvQuery.patch uploaded by marcelloceschia (license
- 1079) chan-sip_Trunk_srvQuery.patch uploaded by st (license 907)
- chan-sip_asterisk18b1_srvQuery.patch uploaded by marcelloceschia
- (license 1079) Tested by: marcelloceschia, st, pabelanger
-
- * main/channel.c, /: Merged revisions 280448 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r280448 | dvossel | 2010-07-29 14:04:23 -0500 (Thu, 29 Jul 2010)
- | 12 lines fixes issue with translator frame not getting freed A
- translator frame even if it local storage so the translation path
- can be freed. This issue prevented g729 licenses from being freed
- up. (closes issue #17630) Reported by: manvirr Patches:
- encoder_fix.diff uploaded by dvossel (license 671) Tested by:
- manvirr, dvossel ........
-
-2010-07-29 16:01 +0000 [r280345] Jean Galarneau <jgalarneau@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 280341 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r280341 | jeang | 2010-07-29 10:52:31 -0500 (Thu, 29 Jul 2010) |
- 2 lines Fix a dsp structure leak occuring when a local channel is
- put into a meetme conference, then masquaraded away. ABE-2422
- ........
-
-2010-07-29 13:45 +0000 [r280306] Matthew Nicholson <mnicholson@digium.com>
-
- * channels/chan_local.c: Implement support for
- ast_channel_queryoption on local channels. Currently only
- AST_OPTION_T38_STATE is supported. ABE-2229 Review:
- https://reviewboard.asterisk.org/r/813/
-
-2010-07-28 20:02 +0000 [r280231] Jason Parker <jparker@digium.com>
-
- * sounds/Makefile: Work around some silly behavior on BSD. A
- non-zero exit from a subshell should make the build fail. (closes
- issue #17621)
-
-2010-07-28 19:57 +0000 [r280229] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_dahdi.c: Add missing enum value "unknown" to the
- SS7 called_nai and calling_nai config options.
-
-2010-07-28 19:54 +0000 [r280193-280227] Jason Parker <jparker@digium.com>
-
- * build_tools/sha1sum-sh (added): Add sha1sum-sh in case there is
- no util on the system.
-
- * sounds/Makefile: Remove unnecessary subshells. Attempt to make
- checksumming work. Also improves readability. (issue #17621)
- Reported by: bjm Review: https://reviewboard.asterisk.org/r/808/
-
-2010-07-28 16:51 +0000 [r280160] Sean Bright <sean@malleable.com>
-
- * apps/app_queue.c: Plug a reference leak in app_queue when adding
- members dynamically. (closes issue #17738) Reported by:
- bobwienholt Patches: issue17738.patch uploaded by bobwienholt
- (license 950) Tested by: bobwienholt, seanbright
-
-2010-07-28 13:51 +0000 [r280089] Leif Madsen <lmadsen@digium.com>
-
- * contrib/scripts/live_ast, /: Merged revisions 280088 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r280088 | lmadsen | 2010-07-28 08:50:38 -0500 (Wed, 28
- Jul 2010) | 1 line Update help text to be less confusing.
- ........
-
-2010-07-27 20:54 +0000 [r279946] David Vossel <dvossel@digium.com>
-
- * main/audiohook.c, main/channel.c, /,
- include/asterisk/audiohook.h: Merged revisions 279945 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r279945 | dvossel | 2010-07-27 15:33:40 -0500 (Tue, 27 Jul 2010)
- | 19 lines remove empty audiohook write list on channel If a
- channel has an audiohook write list created on it, that list
- stays on the channel until the channel is destroyed. There is no
- reason to keep that list on the channel if it becomes empty. If
- it is empty that just means we are doing needless translating for
- every ast_read and ast_write. This patch removes the audiohook
- list from the channel once it is detected to be empty on either a
- read or write. If a audiohook is added back to the channel after
- this list is destroyed, the list just gets recreated as if it
- never existed to begin with. (closes issue #17630) Reported by:
- manvirr Review: https://reviewboard.asterisk.org/r/799/ ........
-
-2010-07-27 17:54 +0000 [r279849-279883] Jason Parker <jparker@digium.com>
-
- * makeopts.in, configure, configure.ac: Add SHA1SUM to configure,
- since we require it for sounds/
-
- * sounds/Makefile: Remove aptly-named EMPTY and BS vars, since they
- aren't used anymore.
-
- * sounds/Makefile: Simply sounds/Makefile some more.
-
-2010-07-27 15:13 +0000 [r279784] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: Fix bad behavior of dynamic_exclude_static
- option in sip.conf. We were attempting to create a contactdeny
- rule based on the peer's IP address before the peer's IP address
- had been set. By moving the processing further down in the
- function, we can ensure stuff works as we expect for it to.
- (closes issue #17717) Reported by: mmichelson Patches:
- 17717.patch uploaded by mmichelson (license 60) Tested by:
- DennisD
-
-2010-07-26 22:59 +0000 [r279657] Jason Parker <jparker@digium.com>
-
- * sounds/Makefile (added), sounds/Makefile.380 (removed),
- configure, include/asterisk/autoconfig.h.in, sounds/Makefile.381
- (removed), configure.ac: Really fix sounds Makefile (and make it
- readableish). There was a rather large syntax error that should
- have caused ALL versions of GNU make to fail. I don't know how it
- worked.
-
-2010-07-26 21:18 +0000 [r279609] Tilghman Lesher <tlesher@digium.com>
-
- * configure, configure.ac: Dunno why this worked on my machine, but
- it works better this way.
-
-2010-07-26 20:25 +0000 [r279597] Gavin Henry <ghenry@suretecsystems.com>
-
- * res/res_config_ldap.c: Apply all patches in:
- https://issues.asterisk.org/view.php?id=13573 (closes issue
- #13573) Reported by: navkumar Patches:
- res_config_ldap-category.diff uploaded by navkumar (license 580)
- res_config_ldap.patch uploaded by bencer (license 961)
- res_config_ldap uploaded by bencer (license 961) Tested by:
- suretec
-
-2010-07-26 19:15 +0000 [r279561] Tilghman Lesher <tlesher@digium.com>
-
- * sounds/Makefile (removed), configure, sounds/Makefile.380
- (added), sounds/Makefile.381 (added), configure.ac: Use a special
- Makefile for noobs who still have GNU Make 3.80. (Closes issue
- #17716) Reported by: farisraouf
-
-2010-07-26 15:41 +0000 [r279501] Sean Bright <sean@malleable.com>
-
- * autoconf/ast_ext_lib.m4: Expand the correct value within
- AST_OPTION_ONLY. (closes issue #17703) Reported by: stuarth
-
-2010-07-24 23:58 +0000 [r279347] Bradley Latus <brad.latus@gmail.com>
-
- * doc/asterisk.8: Minor update to man page
-
-2010-07-23 22:11 +0000 [r279207] Richard Mudgett <rmudgett@digium.com>
-
- * apps/app_queue.c, apps/app_dial.c, /: Merged revisions 279206 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r279206 | rmudgett | 2010-07-23 16:56:44 -0500 (Fri, 23 Jul 2010)
- | 7 lines SIP promiscuous redirect could fail to dial the
- redirect. The ast_channel was created with one variable to
- ast_request() but the call to ast_call() that initiates the
- outgoing call was using a different variable. The two variables
- are not equivalent if the call_forward string included a channel
- technology specifier. e.g., SIP/200 ........
-
-2010-07-23 18:29 +0000 [r279112] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: Backport sip_uri_params_cmp() fix from trunk
- to 1.6.2.
-
-2010-07-23 18:22 +0000 [r279072-279088] Russell Bryant <russell@digium.com>
-
- * /: remove old properties
-
- * /: Add branch-1.4-merged and branch-1.4-blocked properties to
- 1.6.2 branch.
-
-2010-07-23 17:06 +0000 [r278983-278986] Tilghman Lesher <tlesher@digium.com>
-
- * autoconf/ast_check_pwlib.m4, /, configure, configure.ac: Merged
- revisions 278985 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r278985 | tilghman | 2010-07-23 12:05:16 -0500 (Fri, 23 Jul 2010)
- | 12 lines Merged revisions 278984 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r278984 | tilghman | 2010-07-23 12:04:15 -0500 (Fri, 23 Jul 2010)
- | 5 lines Establish a maximum version for openh323 (i.e. not
- opal), because chan_h323 will fail to load, even if it links.
- (issue #17679) Reported by: am ........ ................
-
- * main/asterisk.c, /: Merged revisions 278982 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r278982 | tilghman | 2010-07-23 11:43:34 -0500 (Fri, 23 Jul 2010)
- | 15 lines Merged revisions 278981 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r278981 | tilghman | 2010-07-23 11:42:25 -0500 (Fri, 23 Jul 2010)
- | 8 lines Avoid race with consolethread on shutdown (on parallel
- processors). (closes issue #17080) Reported by: sybasesql
- Patches: 20100721__issue17080.diff.txt uploaded by tilghman
- (license 14) Tested by: sybasesql ........ ................
-
-2010-07-23 15:23 +0000 [r278934] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
-
- * channels/chan_dahdi.c: Two more typos to cancell.
-
-2010-07-22 19:52 +0000 [r278709] Jeff Peeler <jpeeler@digium.com>
-
- * main/xmldoc.c, /: Merged revisions 278708 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r278708 |
- jpeeler | 2010-07-22 14:45:30 -0500 (Thu, 22 Jul 2010) | 16 lines
- Add method for finding XML doc files for systems that don't
- support GLOB_BRACE. In particular, Solaris and perhaps others do
- not support the above mentioned GNU extension. In this case the
- paths are simply expanded without the braces and the calls to
- glob are made separately. Note: I could not explain memory
- allocation failures that were being reported from within libxml
- itself when making calls to glob without using GLOB_NOCHECK. This
- is the only reason why that flag is being used. (closes issue
- #15402) Reported by: snuffy Patches: bug_xmlpatt-v3.diff uploaded
- by snuffy (license 35), modified by me ........
-
-2010-07-22 19:32 +0000 [r278703] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_dahdi.c: DNID does not get cleard on a new call
- when using immediate=yes with ISDN signaling. When you are using
- chan_dahdi ISDN signaling with immediate=yes and a call comes in
- without a DNID then you get the DNID of a previous call.
- Chan_dahdi does not touch the DNID field on a new call if it does
- not have a DNID. Made always copy the DNID from the new call. The
- patches backport the relevant changes from trunk -r210387.
- (closes issue #17568) Reported by: wuwu Patches:
- issue17568_v1.4.patch uploaded by rmudgett (license 664)
- issue17568_v1.6.2.patch uploaded by rmudgett (license 664)
-
-2010-08-10 Leif Madsen <lmadsen@digium.com>
-
- * Asterisk 1.6.2.11 Released.
-
-2010-07-26 Leif Madsen <lmadsen@digium.com>
-
- * Asterisk 1.6.2.11-rc2 Released.
-
-2010-07-26 Leif Madsen <lmadsen@digium.com>
-
- * qwell, asterisk, branch-1.6.2, r279657 ***
- Really fix sounds Makefile (and make it readableish).
- There was a rather large syntax error that should have
- caused ALL versions of GNU make to fail.
- I don't know how it worked.
-
- (Closes issue #17716)
-
-2010-07-22 Leif Madsen <lmadsen@digium.com>
-
- * Asterisk 1.6.2.11-rc1 Released.
-
-2010-07-22 15:00 +0000 [r278621] Mark Michelson <mmichelson@digium.com>
-
- * main/channel.c, /: Merged revisions 278620 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r278620 | mmichelson | 2010-07-22 09:58:01 -0500 (Thu, 22 Jul
- 2010) | 19 lines Merged revisions 278618 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r278618 | mmichelson | 2010-07-22 09:55:04 -0500 (Thu, 22 Jul
- 2010) | 13 lines Allow PLC to function properly when channels use
- SLIN for audio. If a channel involved in a bridge was using SLIN
- audio, then translation paths were not guaranteed to be set up
- properly since in all likelihood the number of translation steps
- was only 1. This patch enforces the transcode_via_slin behavior
- if transcode_via_slin or generic_plc is enabled and one of the
- formats to make compatible is SLIN. AST-352 ........
- ................
-
-2010-07-21 18:22 +0000 [r278524] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
-
- * channels/chan_dahdi.c: Fix invalid test for rxisoffhook in FXO
- channels This fixes some cases of no outgoing calls on FXO before
- an incoming call. Remove an unnecessary testing of an "off-hook"
- bit from DAHDI for FXO (KS/GS) channels.In some cases the bit
- would not be initialized properly before the first inbound call
- and thus prevent an outgoing call. If those tests are actually
- required by anybody, they should define DAHDI_CHECK_HOOKSTATE in
- channels/sig_analog.c . (closes issue #14577) Reported by: jkroon
- Patches: asterisk_chan_dahdi_hookstate_fix.diff uploaded by frawd
- (license 610) Tested by: frawd Review:
- https://reviewboard.asterisk.org/r/699/
-
-2010-07-21 16:20 +0000 [r278479] Russell Bryant <russell@digium.com>
-
- * /, res/res_timing_pthread.c: Merged revisions 278465 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r278465 | russell | 2010-07-21 11:15:00 -0500 (Wed, 21 Jul 2010)
- | 41 lines Use poll() instead of select() in res_timing_pthread
- to avoid stack corruption. This code did not properly check
- FD_SETSIZE to ensure that it did not try to select() on fds that
- were too large. Switching to poll() removes the limitation on the
- maximum fd value. (closes issue #15915) Reported by: keiron
- (closes issue #17187) Reported by: Eddie Edwards (closes issue
- #16494) Reported by: Hubguru (closes issue #15731) Reported by:
- flop (closes issue #12917) Reported by: falves11 (closes issue
- #14920) Reported by: vrban (closes issue #17199) Reported by:
- aleksey2000 (closes issue #15406) Reported by: kowalma (closes
- issue #17438) Reported by: dcabot (closes issue #17325) Reported
- by: glwgoes (closes issue #17118) Reported by: erikje possibly
- other issues, too ... ........
-
-2010-07-21 15:58 +0000 [r278025-278464] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 278463 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r278463 |
- tilghman | 2010-07-21 10:56:05 -0500 (Wed, 21 Jul 2010) | 11
- lines Ensure realtime conferences are treated the same as static
- conferences when trying to find an empty one. Also, parse the
- useropts properly, when retrieving from realtime, and add them to
- the existing flags. (closes issue #17502) Reported by: kenji
- Patches: 20100720__issue17502.diff.txt uploaded by tilghman
- (license 14) Tested by: kenji ........
-
- * apps/app_voicemail.c, /: Merged revisions 278275 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r278275 | tilghman | 2010-07-20 17:40:19 -0500
- (Tue, 20 Jul 2010) | 14 lines Merged revisions 278261 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r278261 | tilghman | 2010-07-20 17:23:13 -0500 (Tue, 20 Jul 2010)
- | 7 lines Delete IMAP messages in reverse order, to ensure
- reordering after each expunge does not cause deletion of the
- wrong message. (closes issue #16350) Reported by: noahisaac
- Patches: 20100623__issue16350.diff.txt uploaded by tilghman
- (license 14) ........ ................
-
- * main/autoservice.c, /, main/features.c,
- include/asterisk/channel.h: Merged revisions 278272 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r278272 | tilghman | 2010-07-20 17:26:23 -0500
- (Tue, 20 Jul 2010) | 11 lines Merged revisions 278167 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r278167 | tilghman | 2010-07-20 15:59:06 -0500 (Tue, 20 Jul 2010)
- | 4 lines Do not queue up DTMF frames while a call is on hold.
- (Fixes ABE-2110) ........ ................
-
- * main/manager.c, /: Merged revisions 278024 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r278024 | tilghman | 2010-07-20 11:50:11 -0500 (Tue, 20 Jul 2010)
- | 14 lines Merged revisions 278023 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r278023 | tilghman | 2010-07-20 11:37:18 -0500 (Tue, 20 Jul 2010)
- | 7 lines Off-by-one error (closes issue #16506) Reported by:
- nik600 Patches: 20100629__issue16506.diff.txt uploaded by
- tilghman (license 14) ........ ................
-
-2010-07-19 21:21 +0000 [r277966] Jean Galarneau <jgalarneau@digium.com>
-
- * /, main/features.c: Merged revisions 277945 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r277945 | jeang | 2010-07-19 16:07:08 -0500 (Mon, 19 Jul 2010) |
- 15 lines Merged revisions 277906 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r277906 | jeang | 2010-07-19 15:16:36 -0500 (Mon, 19 Jul 2010) |
- 7 lines Avoid trying to pickup a parked extension before the park
- operation is completed. A crash could occur if the extension is
- picked up while the parking extension is being announced. Testing
- pu->notquiteyet while searching for a parked extension resolves
- this crash. (ABE-2418) ........ ................
-
-2010-07-17 17:52 +0000 [r277774-277777] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_config_pgsql.c: Merge issues...
-
- * /, autoconf/ast_func_fork.m4, configure,
- include/asterisk/autoconfig.h.in: Merged revisions 277775 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r277775 | tilghman | 2010-07-17 12:42:32 -0500
- (Sat, 17 Jul 2010) | 12 lines Merged revisions 277738 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r277738 | tilghman | 2010-07-17 11:59:11 -0500 (Sat, 17 Jul 2010)
- | 5 lines Remove uclibc cross-compile triplet, as uclibc has a
- working fork()... it's only uclinux that does not. (closes issue
- #17616) Reported by: pprindeville ........ ................
-
- * res/res_config_pgsql.c, res/res_config_odbc.c, /: Merged
- revisions 277773 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r277773 | tilghman | 2010-07-17 12:39:28 -0500 (Sat, 17 Jul 2010)
- | 15 lines Merged revisions 277568 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r277568 | tilghman | 2010-07-16 16:54:29 -0500 (Fri, 16 Jul 2010)
- | 8 lines Since we split values at the semicolon, we should store
- values with a semicolon as an encoded value. (closes issue
- #17369) Reported by: gkservice Patches:
- 20100625__issue17369.diff.txt uploaded by tilghman (license 14)
- Tested by: tilghman ........ ................
-
-2010-07-16 23:37 +0000 [r277666] Tim Ringenbach <tim.ringenbach@gmail.com>
-
- * /, main/features.c: Merged revisions 277657 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r277657 | tringenbach | 2010-07-16 18:23:15 -0500 (Fri, 16 Jul
- 2010) | 16 lines Merged revisions 277625 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r277625 | tringenbach | 2010-07-16 17:43:39 -0500 (Fri, 16 Jul
- 2010) | 9 lines Save and restore AST_FLAG_BRIDGE_HANGUP_DONT on
- attended transfer. ast_bridge_call() clears
- AST_FLAG_BRIDGE_HANGUP_DONT. But during an attended transfer,
- ast_bridge_call() is called for a second bridge on the same
- channel, and it clears that flag, which still needs to get set
- for when the original ast_bridge_call() gets control back and
- checks it. Review: https://reviewboard.asterisk.org/r/741
- ........ ................
-
-2010-07-16 21:31 +0000 [r277563] Matthew Nicholson <mnicholson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 277530 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r277530 | mnicholson | 2010-07-16 16:24:45 -0500 (Fri, 16 Jul
- 2010) | 11 lines Merged revisions 277497 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r277497 | mnicholson | 2010-07-16 16:18:38 -0500 (Fri, 16 Jul
- 2010) | 4 lines Default to no udptl error correction so that
- error correction will be disabled in the event that the remote
- end indicates that they do not support the error correction mode
- we requested. FAX-128 ........ ................
-
-2010-07-16 21:16 +0000 [r277489] Jeff Peeler <jpeeler@digium.com>
-
- * apps/app_queue.c, /: Merged revisions 277488 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r277488 |
- jpeeler | 2010-07-16 16:16:08 -0500 (Fri, 16 Jul 2010) | 10 lines
- Fix reporting estimated queue hold time. Just say the number of
- seconds (after minutes) rather than doing some incorrect
- calculation with respect to minutes. (closes issue #17498)
- Reported by: corruptor Patches: holdesecs_bug.diff uploaded by
- corruptor (license 253) ........
-
-2010-07-16 20:35 +0000 [r277485] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 277467 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r277467 | rmudgett | 2010-07-16 15:27:51 -0500
- (Fri, 16 Jul 2010) | 22 lines Merged revisions 277419 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r277419 | rmudgett | 2010-07-16 15:18:54 -0500 (Fri, 16 Jul 2010)
- | 15 lines priexclusive in chan_dahdi.conf ignored when reloading
- dahdi module During a reload, the priexclusive and outsignalling
- parameters are not read in from the config file as intended.
- Unfortunately, they get set to defaults as a result. This patch
- makes sure that they do not get set to defaults during a reload.
- (closes issue #17441) Reported by: mtryfoss Patches:
- issue17441_v1.4.patch uploaded by rmudgett (license 664)
- issue17441_v1.6.2.patch uploaded by rmudgett (license 664)
- issue17441_trunk.patch uploaded by rmudgett (license 664) Tested
- by: rmudgett ........ ................
-
-2010-07-16 20:30 +0000 [r277478] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_musiconhold.c, contrib/realtime/mysql/musiconhold.sql
- (added), /: Merged revisions 277452 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r277452 |
- tilghman | 2010-07-16 15:25:11 -0500 (Fri, 16 Jul 2010) | 2 lines
- Add documentation for MOH realtime fields ........
-
-2010-07-16 19:24 +0000 [r277377] Jeff Peeler <jpeeler@digium.com>
-
- * apps/app_queue.c, /: Merged revisions 277366 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r277366 |
- jpeeler | 2010-07-16 14:22:49 -0500 (Fri, 16 Jul 2010) | 7 lines
- Add missing handling for ringing state for use with queue empty
- options. (closes issue #17471) Reported by: jazzy Patches:
- app_queue.c.diff uploaded by jazzy (license 1056) ........
-
-2010-07-16 18:33 +0000 [r277338] Matthew Nicholson <mnicholson@digium.com>
-
- * main/pbx.c, /: Merged revisions 277331 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r277331 | mnicholson | 2010-07-16 13:31:08 -0500 (Fri, 16 Jul
- 2010) | 15 lines Merged revisions 277327 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r277327 | mnicholson | 2010-07-16 13:30:22 -0500 (Fri, 16 Jul
- 2010) | 8 lines Interpret device state AST_DEVICE_UNKNOWN as
- extension state AST_EXTENSION_NOT_INUSE. (closes issue #16035)
- Reported by: francesco_r Patches: pbx.c.patch uploaded by
- viniciusfontes (license 978) Tested by: francesco_r, agx, lawbar
- ........ ................
-
-2010-07-16 18:14 +0000 [r277264] Tilghman Lesher <tlesher@digium.com>
-
- * main/manager.c, /: Merged revisions 277263 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r277263 | tilghman | 2010-07-16 13:14:05 -0500 (Fri, 16 Jul 2010)
- | 12 lines Merged revisions 277261 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r277261 | tilghman | 2010-07-16 13:04:11 -0500 (Fri, 16 Jul 2010)
- | 5 lines If variable gotten is not set, will segfault on
- Solaris. (closes issue #17636) Reported by: bklang ........
- ................
-
-2010-07-16 17:31 +0000 [r277256] Matthew Nicholson <mnicholson@digium.com>
-
- * main/channel.c, /: Merged revisions 277250 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r277250 | mnicholson | 2010-07-16 12:30:39 -0500 (Fri, 16 Jul
- 2010) | 11 lines Merged revisions 277247 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r277247 | mnicholson | 2010-07-16 12:29:57 -0500 (Fri, 16 Jul
- 2010) | 4 lines For pass through DTMF tones, measure the actual
- duration between the begin and end packets on the wire. If it is
- detected to be less than AST_MIN_DTMF_DURATION, trigger dtmf
- emulation. AST-362 ........ ................
-
-2010-07-16 17:18 +0000 [r277188] Paul Belanger <paul.belanger@polybeacon.com>
-
- * /, apps/app_amd.c: Merged revisions 277183 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r277183 | pabelanger | 2010-07-16 13:13:46 -0400 (Fri, 16 Jul
- 2010) | 15 lines Merged revisions 277182 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r277182 | pabelanger | 2010-07-16 13:10:36 -0400 (Fri, 16 Jul
- 2010) | 8 lines Total analysis time error with SIP and silence
- suppression When using app_amd with SIP providers that have
- silence suppression on, the iTotalTime count increases
- exponentially. (closes issue #17656) Reported by: juls ........
- ................
-
-2010-07-16 15:21 +0000 [r277144] Sean Bright <sean@malleable.com>
-
- * /, main/translate.c: Merged revisions 277143 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r277143 |
- seanbright | 2010-07-16 11:20:40 -0400 (Fri, 16 Jul 2010) | 8
- lines Avoid crashing when installing a duplicate translation path
- with a lower cost. (closes issue #17092) Reported by: moy
- Patches: translate.rev254273.patch uploaded by moy (license 222)
- Tested by: moy ........
-
-2010-07-15 20:42 +0000 [r276572-276809] Jeff Peeler <jpeeler@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 276788 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r276788 |
- jpeeler | 2010-07-15 15:21:03 -0500 (Thu, 15 Jul 2010) | 6 lines
- Correct not setting the bindport before attempting to open the
- socket. Related to changes from 276571, I was accidentally
- testing with a port set in my configuration causing me to miss
- this. Also moved the TCP handling as well to occur before
- build_peer is called. ........
-
- * main/channel.c, /: Merged revisions 276653 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r276653 | jpeeler | 2010-07-15 08:51:11 -0500 (Thu, 15 Jul 2010)
- | 9 lines Merged revisions 276652 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r276652 | jpeeler | 2010-07-15 08:48:58 -0500 (Thu, 15 Jul 2010)
- | 2 lines In a perfect world, the frame source would never be
- NULL. In the meantime, don't crash when it is. ........
- ................
-
- * /, channels/chan_sip.c: Merged revisions 276571 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r276571 |
- jpeeler | 2010-07-14 17:58:24 -0500 (Wed, 14 Jul 2010) | 21 lines
- Fix MWI notification transmission problems over SIP. MWI updates
- were not being sent if no messages were found in the event cache.
- This was corrected since a phone may need to clear its MWI status
- configured previously from another mailbox. Upon module or sip
- reload, MWI updates could not be sent due to the sipsock socket
- not being set early enough in reload_config. The code handling
- the descriptor assignment and such has simply been moved before
- the call to build_peer. Issuing a sip reload cleared the IP
- address of the peer, but skipped checking the database for
- registration information. The database is now checked both for
- sip reload and actually reloading the module. If a transmission
- occurs before the do_monitor thread has started, do not attempt
- to send a signal to it. (closes issue #17398) Reported by: ip-rob
- ........
-
-2010-07-14 20:16 +0000 [r276442] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/loader.c, /: Merged revisions 276441 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r276441 |
- kpfleming | 2010-07-14 15:15:48 -0500 (Wed, 14 Jul 2010) | 4
- lines Don't try to call an embedded module's backup_globals()
- function until after confirming it exists. ........
-
-2010-07-14 11:52 +0000 [r276269] Leif Madsen <lmadsen@digium.com>
-
- * /, configs/voicemail.conf.sample: Merged revisions 276268 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r276268 | lmadsen | 2010-07-14 06:51:48 -0500
- (Wed, 14 Jul 2010) | 9 lines Merged revisions 276267 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r276267 | lmadsen | 2010-07-14 06:49:01 -0500 (Wed, 14
- Jul 2010) | 1 line Update documentation for voicemail.conf
- externpass option. ........ ................
-
-2010-07-13 19:11 +0000 [r276125] Russell Bryant <russell@digium.com>
-
- * /, main/features.c: Merged revisions 276124 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r276124 | russell | 2010-07-13 14:09:42 -0500 (Tue, 13 Jul 2010)
- | 9 lines Merged revisions 276123 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r276123 | russell | 2010-07-13 14:06:53 -0500 (Tue, 13 Jul 2010)
- | 2 lines Use chan->cdr instead of chan_cdr (just like peer->cdr
- instead of peer_cdr in the last commit). ........
- ................
-
-2010-07-13 19:01 +0000 [r276121] Jeff Peeler <jpeeler@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 276074 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r276074 | jpeeler | 2010-07-13 12:37:40 -0500 (Tue, 13 Jul 2010)
- | 19 lines Merged revisions 275773 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r275773 | jpeeler | 2010-07-12 15:34:51 -0500 (Mon, 12 Jul 2010)
- | 12 lines Make user removals and traversals thread safe in
- meetme. Race conditions present in meetme involving the user list
- where a lack of locking has the potential for a user to be
- removed during a traversal or as in the case of the reporter
- after checking if the list is empty could cause a crash. Fixing
- this was done by convering the userlist to an ao2 container.
- (closes issue #17390) Reported by: Vince Review:
- https://reviewboard.asterisk.org/r/746/ ........ ................
-
-2010-07-13 16:55 +0000 [r275996] Russell Bryant <russell@digium.com>
-
- * /, main/features.c: Merged revisions 275995 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r275995 | russell | 2010-07-13 11:53:44 -0500 (Tue, 13 Jul 2010)
- | 21 lines Merged revisions 275994 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r275994 | russell | 2010-07-13 11:51:18 -0500 (Tue, 13 Jul 2010)
- | 14 lines Access peer->cdr directly instead of through a saved
- off reference. At this point in the code, it is possible that
- peer_cdr may be invalid. Specifically, in the blind transfer
- code, CDRs are swapped between channels. So, peer_cdr is no
- longer == peer->cdr. The scenario that exposed a crash in this
- code was a blind transfer that hit the system call limit, causing
- the transferee channel to get destroyed after the transfer
- attempt failed. Even if it succeeds and this code doesn't crash,
- this code was still trying to reset a CDR on a channel that was
- now owned by a different thread, which is a BadThing(tm).
- (ABE-2417) ........ ................
-
-2010-07-13 14:49 +0000 [r275911] Tilghman Lesher <tlesher@digium.com>
-
- * contrib/realtime/mysql, contrib/realtime/oracle,
- contrib/scripts/sip-friends.sql (removed),
- contrib/realtime/mysql/sipfriends.sql,
- contrib/realtime/mysql/voicemail.sql, contrib/scripts/vmdb.sql
- (removed), contrib/realtime/mysql/meetme.sql,
- contrib/realtime/sqlserver, contrib/scripts/realtime_pgsql.sql
- (removed), contrib/scripts/iax-friends.sql (removed), /,
- contrib/realtime/mysql/iaxfriends.sql, contrib/scripts/meetme.sql
- (removed), contrib/realtime (added), contrib/realtime/postgresql,
- contrib/realtime/postgresql/realtime.sql: Merged revisions 275910
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r275910 | tilghman | 2010-07-13 09:48:40 -0500
- (Tue, 13 Jul 2010) | 9 lines Merged revisions 275909 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r275909 | tilghman | 2010-07-13 09:47:30 -0500 (Tue, 13
- Jul 2010) | 2 lines Move SQL scripts into their own
- database-specific directories. ........ ................
-
-2010-07-12 17:26 +0000 [r275706] Jeff Peeler <jpeeler@digium.com>
-
- * main/channel.c, /: Merged revisions 275682 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r275682 | jpeeler | 2010-07-12 12:21:01 -0500 (Mon, 12 Jul 2010)
- | 18 lines Merged revisions 275665 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r275665 | jpeeler | 2010-07-12 11:58:39 -0500 (Mon, 12 Jul 2010)
- | 11 lines Change ast_write to not stop generator when called
- from ast_prod. For SIP channels configured with the
- progressinband option on, the ringback was being immediately
- stopped. This problem was due to ast_prod being moved for a
- deadlock fix in 259858. Prodding the channel after setting up the
- generator triggered the check in ast_write to stop the generator.
- The fix here should write the frame the same as was done before
- the call to ast_prod was moved. (closes issue #17372) Reported
- by: tech_admin ........ ................
-
-2010-07-12 15:38 +0000 [r275627] Leif Madsen <lmadsen@digium.com>
-
- * cdr/cdr_pgsql.c, /: Merged revisions 275626 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r275626 |
- lmadsen | 2010-07-12 10:37:01 -0500 (Mon, 12 Jul 2010) | 11 lines
- cdr_pgsql does not detect when a table is found. This change adds
- an ERROR message to let you know when a failure exists to get the
- columns from the pgsql database, which typically means that the
- table does not exist. (closes issue #17478) Reported by: kobaz
- Patches: cdr_pgsql.patch uploaded by kobaz (license 834) Tested
- by: kobaz, russell, lmadsen ........
-
-2010-07-10 15:11 +0000 [r275311-275469] Russell Bryant <russell@digium.com>
-
- * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions
- 245192 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r245192 |
- mmichelson | 2010-02-06 08:43:03 -0600 (Sat, 06 Feb 2010) | 21
- lines Remove useless sip options related to hash table size.
- First off, these options weren't actually doing anything. By the
- time the options were parsed, the peer and dialog containers had
- already been allocated with their default values. Second, hash
- table size is something that doesn't really make sense to change
- in a config file. If a user is that interested in changing the
- hashtable size, he can modify the source itself. I have removed
- the parsing of the hash_peer, hash_user, and hash_dialog options.
- I have removed the hash_user_size variable altogether since it is
- not used at all. I also changed hash_peer_size and
- hash_dialog_size to be constant, and have changed the symbols to
- be in all caps as constants typically are. I have also removed
- the entire section in sip.conf.sample regarding configurable
- hashtable sizes. ........ (merge to 1.6.2 inspired by issue
- #17553)
-
- * /: unblock a rev
-
- * configs/features.conf.sample, /, main/features.c: Merged
- revisions 275424 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r275424 |
- russell | 2010-07-09 16:57:21 -0500 (Fri, 09 Jul 2010) | 27 lines
- Fix some issues related to dynamic feature groups in
- features.conf. The bridge handling code did not properly consider
- feature groups when setting parameters that would affect whether
- or not a native bridge would be attempted. If DYNAMIC_FEATURES
- only include a feature group, a native bridge would occur that
- may prevent features from working. Fix a bug in verbose output
- that would show the key mapping as empty if it was using the
- default mapping and not a custom mapping in the feature group.
- Add feature groups to the output of "features show". Adjust the
- feature execution logic to match that of the logic when executing
- a feature that was not configured through a feature group. Update
- features.conf.sample to show that an '=' is still required if
- using the default key mapping from [applicationmap]. Finally,
- clean up a little bit of formatting to better coform to coding
- guidelines while in the area. (closes issue #17589) Reported by:
- lmadsen Patches: issue_17589.rev4.txt uploaded by russell
- (license 2) Tested by: russell, lmadsen ........
-
- * /, main/features.c: Merged revisions 275310 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r275310 |
- russell | 2010-07-09 14:58:06 -0500 (Fri, 09 Jul 2010) | 2 lines
- Add missing ao2_iterator_destroy(). ........
-
-2010-07-09 19:23 +0000 [r275260] Paul Belanger <paul.belanger@polybeacon.com>
-
- * /, channels/chan_sip.c: Merged revisions 275249 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r275249 | pabelanger | 2010-07-09 15:21:27 -0400 (Fri, 09 Jul
- 2010) | 15 lines Merged revisions 275241 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r275241 | pabelanger | 2010-07-09 15:20:00 -0400 (Fri, 09 Jul
- 2010) | 8 lines Fix logging message for stale nonce. (closes
- issue #17582) Reported by: kenner Patches: chan_sip.c.diff
- uploaded by kenner (license 1040) Tested by: lmadsen ........
- ................
-
-2010-07-09 18:24 +0000 [r275191] Matthew Nicholson <mnicholson@digium.com>
-
- * main/loader.c, /: Merged revisions 275186 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r275186 | mnicholson | 2010-07-09 13:24:03 -0500 (Fri, 09 Jul
- 2010) | 9 lines Merged revisions 275182 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r275182 | mnicholson | 2010-07-09 13:23:23 -0500 (Fri, 09 Jul
- 2010) | 2 lines give a better error message when attempting to
- unload a module that is not loaded ........ ................
-
-2010-07-09 18:11 +0000 [r275148] Russell Bryant <russell@digium.com>
-
- * configs/features.conf.sample, /: Merged revisions 275147 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r275147 | russell | 2010-07-09 13:11:13 -0500 (Fri, 09
- Jul 2010) | 2 lines Move parking lot sample config out from the
- middle of dynamic features sample config. ........
-
-2010-07-09 17:51 +0000 [r275029-275145] Matthew Nicholson <mnicholson@digium.com>
-
- * main/loader.c, /: Merged revisions 275144 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r275144 | mnicholson | 2010-07-09 12:50:45 -0500 (Fri, 09 Jul
- 2010) | 9 lines Merged revisions 275143 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r275143 | mnicholson | 2010-07-09 12:50:05 -0500 (Fri, 09 Jul
- 2010) | 2 lines don't unload modules that returned
- AST_MODULE_LOAD_DECLINE when they were loaded ........
- ................
-
- * apps/app_dial.c, /: Merged revisions 275028 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r275028 | mnicholson | 2010-07-09 11:05:58 -0500 (Fri, 09 Jul
- 2010) | 15 lines Merged revisions 275027 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r275027 | mnicholson | 2010-07-09 11:04:21 -0500 (Fri, 09 Jul
- 2010) | 8 lines Clear the AST_CDR_FLAG_DIALED flag for channels
- going into the pbx via the G option in app_dial (closes issue
- #17592) Reported by: jamicque Patches: G-flag-cdr-fix1.diff
- uploaded by mnicholson (license 96) Tested by: jamicque,
- mnicholson ........ ................
-
-2010-07-09 15:39 +0000 [r275023] Russell Bryant <russell@digium.com>
-
- * include/asterisk/test.h, /, main/test.c: Merged revisions 275022
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r275022 | russell | 2010-07-09 10:35:53 -0500
- (Fri, 09 Jul 2010) | 11 lines Merged revisions 275021 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r275021 | russell | 2010-07-09 10:33:08 -0500 (Fri, 09 Jul 2010)
- | 4 lines Document that a leading and trailing slash is expected
- for test categories. Also, emit a warning if a test is registered
- without one of these. ........ ................
-
-2010-07-07 18:34 +0000 [r274627-274640] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 274639 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r274639 | rmudgett | 2010-07-07 13:32:35 -0500 (Wed, 07 Jul 2010)
- | 1 line Add missing conditional around chan_dahdi
- mfcr2_skip_category config parameter. ........
-
- * channels/chan_dahdi.c, /: Merged revisions 274595 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r274595 | rmudgett | 2010-07-07 13:20:00 -0500
- (Wed, 07 Jul 2010) | 9 lines Merged revisions 274579 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r274579 | rmudgett | 2010-07-07 13:12:41 -0500 (Wed, 07
- Jul 2010) | 1 line Close the DAHDI FD on error when processing
- chan_dahdi toneduration config parameter. ........
- ................
-
-2010-07-07 06:16 +0000 [r274419] Tilghman Lesher <tlesher@digium.com>
-
- * configs/say.conf.sample, /: Merged revisions 274418 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r274418 | tilghman | 2010-07-07 01:15:43 -0500
- (Wed, 07 Jul 2010) | 15 lines Merged revisions 274417 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r274417 | tilghman | 2010-07-07 01:13:54 -0500 (Wed, 07 Jul 2010)
- | 8 lines Correct how 100, 200, 300, etc. is said. Also add the
- crazy British numbers. (closes issue #16102) Reported by: Delvar
- Patches: say.conf.fix.patch uploaded by Delvar (license 908)
- (plus a few additional fixes and simplifications by me) ........
- ................
-
-2010-07-06 23:06 +0000 [r274360] Terry Wilson <twilson@digium.com>
-
- * configs/sip.conf.sample, channels/chan_sip.c: Merged revisions
- 274284 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r274284 | twilson | 2010-07-06 17:15:27 -0500 (Tue, 06 Jul 2010)
- | 18 lines Merged revisions 274280 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r274280 | twilson | 2010-07-06 17:08:20 -0500 (Tue, 06 Jul 2010)
- | 9 lines Add option to not do a call forward on 482 Loop
- Detected Asterisk has always set up a forwarded call when
- receiving a 482 Loop Detected. This prevents handling the call
- failure by just continuing on in the dialplan. Since this would
- be a change in behavior, the new option to disable this behavior
- is forwardloopdetected which defaults to 'yes'. Review:
- https://reviewboard.asterisk.org/r/764/ ........ ................
-
-2010-07-06 22:30 +0000 [r274347] Jeff Peeler <jpeeler@digium.com>
-
- * configs/sip.conf.sample, /: Merged revisions 274316 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r274316 | jpeeler | 2010-07-06 17:23:35 -0500
- (Tue, 06 Jul 2010) | 14 lines Merged revisions 274283 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r274283 | jpeeler | 2010-07-06 17:15:21 -0500 (Tue, 06 Jul 2010)
- | 7 lines Correct sip.conf.sample comments for prematuremedia
- option. (closes issue #17513) Reported by: festr Patches: patch
- uploaded by festr (license 443) ........ ................
-
-2010-07-06 22:10 +0000 [r274282] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 274281 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r274281 | tilghman | 2010-07-06 17:09:23 -0500 (Tue, 06 Jul 2010)
- | 2 lines Status shows all non-CRC4 lines as "yellow", even if
- "yellow" was not in the bitfield. ........
-
-2010-07-06 14:33 +0000 [r274168] Mark Michelson <mmichelson@digium.com>
-
- * main/rtp.c, /: Merged revisions 274164 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r274164 | mmichelson | 2010-07-06 09:31:13 -0500 (Tue, 06 Jul
- 2010) | 22 lines Merged revisions 274157 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r274157 | mmichelson | 2010-07-06 09:29:23 -0500 (Tue, 06 Jul
- 2010) | 16 lines Fix problem with RFC 2833 DTMF not being
- accepted. A recent check was added to ensure that we did not
- erroneously detect duplicate DTMF when we received packets out of
- order. The problem was that the check did not account for the
- fact that the seqno of an RTP stream will roll over back to 0
- after hitting 65535. Now, we have a secondary check that will
- ensure that the seqno rolling over will not cause us to stop
- accepting DTMF. (closes issue #17571) Reported by: mdeneen
- Patches: rtp_seqno_rollover.patch uploaded by mmichelson (license
- 60) Tested by: richardf, maxochoa, JJCinAZ ........
- ................
-
-2010-07-05 13:55 +0000 [r273888] Paul Belanger <paul.belanger@polybeacon.com>
-
- * main/config.c, /: Merged revisions 273886 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r273886 | pabelanger | 2010-07-05 09:53:44 -0400 (Mon, 05 Jul
- 2010) | 15 lines Merged revisions 273884 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r273884 | pabelanger | 2010-07-05 09:51:29 -0400 (Mon, 05 Jul
- 2010) | 8 lines Remove extra line breaks from 'core show config
- mappings' (closes issue #17583) Reported by: pabelanger Patches:
- issue17583.patch uploaded by pabelanger (license 224) Tested by:
- lmadsen ........ ................
-
-2010-07-03 02:43 +0000 [r273716-273831] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_local.c, /, channels/chan_agent.c,
- channels/chan_h323.c, include/asterisk/lock.h: Merged revisions
- 273830 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r273830 | tilghman | 2010-07-02 21:36:31 -0500 (Fri, 02 Jul 2010)
- | 16 lines Merged revisions 273793 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r273793 | tilghman | 2010-07-02 16:36:39 -0500 (Fri, 02 Jul 2010)
- | 9 lines Have the DEADLOCK_AVOIDANCE macro warn when an unlock
- fails, to help catch potentially large software bugs. (closes
- issue #17407) Reported by: pdf Patches:
- 20100527__issue17407.diff.txt uploaded by tilghman (license 14)
- Review: https://reviewboard.asterisk.org/r/751/ ........
- ................
-
- * main/autoservice.c, /: Merged revisions 273718 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r273718 | tilghman | 2010-07-02 12:10:59 -0500 (Fri, 02 Jul 2010)
- | 15 lines Merged revisions 273717 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r273717 | tilghman | 2010-07-02 12:09:47 -0500 (Fri, 02 Jul 2010)
- | 8 lines Autoservice loop optimization causes a busy loop, when
- channels are serviced while in hangup. (closes issue #17564)
- Reported by: ramonpeek Patches: 20100630__issue17564.diff.txt
- uploaded by tilghman (license 14) Tested by: ramonpeek ........
- ................
-
- * apps/app_queue.c, /: Merged revisions 273714 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r273714 |
- tilghman | 2010-07-02 11:57:28 -0500 (Fri, 02 Jul 2010) | 2 lines
- The switch fallthrough could create some errorneous situations,
- so best to force directly to the default case. ........
-
-2010-07-02 15:59 +0000 [r273642] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
-
- * channels/chan_iax2.c, apps/app_voicemail.c,
- channels/chan_dahdi.c, channels/chan_sip.c, res/res_agi.c: Fix
- typos reported by Lintian
-
-2010-07-01 22:17 +0000 [r273571] Russell Bryant <russell@digium.com>
-
- * main/datastore.c, /: Merged revisions 273566 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r273566 | russell | 2010-07-01 17:16:23 -0500 (Thu, 01 Jul 2010)
- | 14 lines Merged revisions 273565 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r273565 | russell | 2010-07-01 17:09:19 -0500 (Thu, 01 Jul 2010)
- | 7 lines Don't return a partially initialized datastore. If
- memory allocation fails in ast_strdup(), don't return a partially
- initialized datastore. Bad things may happen. (related to
- ABE-2415) ........ ................
-
-2010-07-01 20:29 +0000 [r273356-273529] Jeff Peeler <jpeeler@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 273522 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r273522 | jpeeler | 2010-07-01 15:28:15 -0500 (Thu, 01 Jul 2010)
- | 21 lines Merged revisions 273474 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r273474 | jpeeler | 2010-07-01 15:19:16 -0500 (Thu, 01 Jul 2010)
- | 14 lines Allow admin user to join conference without using
- admin mode and no user pin. Configuring the conference in
- meetme.conf like the following: conf => 2345,,6666 did not prompt
- for pin when used without admin mode. This meant that the
- conference could not be joined as an admin even if the user knew
- the correct pin. The original bug report was submitted claiming
- that the blank user pin should deny entry into the conference. I
- think a better way to handle this would be with a feature
- enhancement that used the following syntax: conf => 2345,X,6666 -
- where X denotes no acceptable pin allowed (closes issue #15704)
- Reported by: modelnine ........ ................
-
- * /, apps/app_meetme.c: Merged revisions 273355 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r273355 | jpeeler | 2010-07-01 10:12:31 -0500 (Thu, 01 Jul 2010)
- | 19 lines Merged revisions 273354 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r273354 | jpeeler | 2010-07-01 10:05:43 -0500 (Thu, 01 Jul 2010)
- | 12 lines Ensure channel placed in meetme in ringing state is
- properly hung up. An outgoing channel placed in meetme while
- still ringing which was then hung up would not exit meetme and
- the channel was not properly destroyed. Specifically checking for
- this scenario by looking at the appropriate control frames
- resolves the issue. (closes issue #15871) Reported by: Ivan
- Patches: meetme_congestion_trunk_v2.patch uploaded by Ivan
- (license 229) ........ ................
-
-2010-07-01 14:39 +0000 [r273271-273353] Matthew Nicholson <mnicholson@digium.com>
-
- * main/manager.c, /: Merged revisions 273352 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r273352 |
- mnicholson | 2010-07-01 09:37:37 -0500 (Thu, 01 Jul 2010) | 2
- lines Fixed whitespace problems ........
-
- * main/manager.c, /: Merged revisions 273350 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r273350 |
- mnicholson | 2010-07-01 09:34:31 -0500 (Thu, 01 Jul 2010) | 2
- lines Altered my comment about TCP_NODELAY ........
-
- * main/manager.c, /: Merged revisions 273270 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r273270 |
- mnicholson | 2010-06-30 13:48:21 -0500 (Wed, 30 Jun 2010) | 2
- lines Set TCP_NODELAY on manager TCP sockets to prevent delays on
- outgoing packets. This regression was introduced in r48338.
- AST-359 ........
-
-2010-06-30 17:32 +0000 [r273193-273234] Paul Belanger <paul.belanger@polybeacon.com>
-
- * main/rtp.c, /: Merged revisions 273233 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r273233 |
- pabelanger | 2010-06-30 13:28:04 -0400 (Wed, 30 Jun 2010) | 11
- lines Fix rt(c)p set debug ip taking wrong argument Also clean up
- some coding errors. (closes issue #17469) Reported by: wdoekes
- Patches: astsvn-rtp-set-debug-ip.patch uploaded by wdoekes
- (license 717) Tested by: wdoekes, pabelanger ........
-
- * /: Revert previous commit; res_rtp_asterisk.c does not exist.
-
- * /: Unblock revisions 218107 ........ r218107 | mvanbaak |
- 2009-09-12 15:08:16 +0200 (Sat, 12 Sep 2009) | 8 lines use the
- actual given ip address for 'rtp set debug ip <foo>' instead of
- the word 'ip' (closes issue 0015711) Reported by: davidw Patches:
- 2009082800-rtpdebug.diff.txt uploaded by mvanbaak (license 7)
- Tested by: davidw ........
-
-2010-06-30 01:07 +0000 [r273056-273145] Tilghman Lesher <tlesher@digium.com>
-
- * main/manager.c, /: Merged revisions 273144 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r273144 |
- tilghman | 2010-06-29 20:07:02 -0500 (Tue, 29 Jun 2010) | 8 lines
- Permission checking for the system application is backwards.
- (closes issue #17550) Reported by: kenner Patches: manager.c.diff
- uploaded by kenner (license 1040) Tested by: kenner ........
-
- * main/config.c, /: Merged revisions 273142 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r273142 |
- tilghman | 2010-06-29 20:01:14 -0500 (Tue, 29 Jun 2010) | 5 lines
- Don't attempt to proceed if our internal parser indicates an
- invalid file. (closes issue #17560) Reported by: Nick_Lewis
- ........
-
- * /, channels/chan_sip.c: Merged revisions 273078 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r273078 | tilghman | 2010-06-29 18:20:40 -0500 (Tue, 29 Jun 2010)
- | 17 lines Merged revisions 273060 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r273060 | tilghman | 2010-06-29 18:15:28 -0500 (Tue, 29 Jun 2010)
- | 10 lines Allow the "useragent" value to be restored into memory
- from the realtime backend. This value is purely informational. It
- does not alter configuration at all. (closes issue #16029)
- Reported by: Guggemand Patches: realtime-useragent.patch uploaded
- by Guggemand (license 897) Tested by: Guggemand ........
- ................
-
- * main/channel.c, /: Merged revisions 273058 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r273058 | tilghman | 2010-06-29 17:59:51 -0500 (Tue, 29 Jun 2010)
- | 11 lines Recorded merge of revisions 273057 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r273057 | tilghman | 2010-06-29 17:58:58 -0500 (Tue, 29 Jun 2010)
- | 4 lines _Really_ skip the channel... don't just retry for
- another 200 cycles. (Closes issue SWP-1652, ABE-2240) ........
- ................
-
- * main/pbx.c, /: Merged revisions 273054 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r273054 |
- tilghman | 2010-06-29 17:39:22 -0500 (Tue, 29 Jun 2010) | 11
- lines Send DialPlanComplete as a response, not as a separate
- event. Otherwise, it goes to all manager sessions and may exclude
- the current session, if the Events mask excludes it. (closes
- issue #17504) Reported by: rrb3942 Patches:
- showdialplan_patch.diff uploaded by rrb3942 (license 1003) Tested
- by: rrb3942 ........
-
-2010-06-29 16:43 +0000 [r272972] Russell Bryant <russell@digium.com>
-
- * main/asterisk.c, /: Merged revisions 253357 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r253357 |
- russell | 2010-03-18 13:18:43 -0500 (Thu, 18 Mar 2010) | 8 lines
- Increase CLI command output timeout for asterisk -rx to 60
- seconds. (closes issue #17049) Reported by: russell Tested by:
- russell Review: https://reviewboard.asterisk.org/r/573/ ........
-
-2010-07-22 Leif Madsen <lmadsen@digium.com>
-
- * Release Asterisk 1.6.2.10
-
- * Included a fix for res_timing_pthread per the description below:
-
- r278465 | russell | 2010-07-21 11:15:00 -0500 (Wed, 21 Jul 2010) | 41 lines
-
- Use poll() instead of select() in res_timing_pthread to avoid stack corruption.
- This code did not properly check FD_SETSIZE to ensure that it did not try to
- select() on fds that were too large. Switching to poll() removes the limitation
- on the maximum fd value.
-
-2010-07-07 Leif Madsen <lmadsen@digium.com>
-
- * Release Asterisk 1.6.2.10-rc2
-
- * Fix problem with RFC 2833 DTMF not being accepted.
-
- A recent check was added to ensure that we did not erroneously
- detect duplicate DTMF when we received packets out of order.
- The problem was that the check did not account for the fact that
- the seqno of an RTP stream will roll over back to 0 after hitting
- 65535. Now, we have a secondary check that will ensure that the
- seqno rolling over will not cause us to stop accepting DTMF.
-
- (closes issue 0017571)
- Reported by: mdeneen
- Patches:
- rtp_seqno_rollover.patch uploaded by mmichelson (license 60)
- Tested by: richardf, maxochoa, JJCinAZ
-
- * Clear the AST_CDR_FLAG_DIALED flag for channels going into the pbx
- via the G option in app_dial
-
- (closes issue 0017592)
- Reported by: jamicque
- Patches:
- G-flag-cdr-fix1.diff uploaded by mnicholson (license 96)
- Tested by: jamicque, mnicholson
-
-2010-06-29 Leif Madsen <lmadsen@digium.com>
-
- * Release Asterisk 1.6.2.10-rc1
-
-2010-06-28 21:51 +0000 [r272924-272927] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c, /: Merged revisions 272926 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r272926 | tilghman | 2010-06-28 16:50:57 -0500 (Mon, 28 Jun 2010)
- | 15 lines Merged revisions 272925 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r272925 | tilghman | 2010-06-28 16:50:02 -0500 (Mon, 28 Jun 2010)
- | 8 lines Don't change ownership/group/permissions on run
- directory, if it already exists. (closes issue #17076) Reported
- by: stuarth Patches: 20100324__issue17076.diff.txt uploaded by
- tilghman (license 14) Tested by: stuarth ........
- ................
-
- * main/config.c, /: Merged revisions 272923 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r272923 | tilghman | 2010-06-28 16:42:52 -0500 (Mon, 28 Jun 2010)
- | 19 lines Merged revisions 272921-272922 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r272921 | tilghman | 2010-06-28 16:29:27 -0500 (Mon, 28 Jun 2010)
- | 8 lines Change the way that we read include files, to
- accommodate for changes in GCC 4.4. (closes issue #17472)
- Reported by: seandarcy Patches: config2.patch uploaded by nivan
- (license 1066) Tested by: nivan ........ r272922 | tilghman |
- 2010-06-28 16:38:49 -0500 (Mon, 28 Jun 2010) | 2 lines Also trim
- trailing blanks on #includes ........ ................
-
-2010-06-28 18:50 +0000 [r272882] Russell Bryant <russell@digium.com>
-
- * tests/test_astobj2.c (added): Backport applicable parts of
- test_astobj2 from trunk.
-
-2010-06-28 17:37 +0000 [r272806] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 272805 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r272805 | mmichelson | 2010-06-28 12:33:12 -0500 (Mon, 28 Jun
- 2010) | 11 lines Merged revisions 272804 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r272804 | mmichelson | 2010-06-28 12:31:40 -0500 (Mon, 28 Jun
- 2010) | 5 lines Decode URI in contact header of 302 response.
- ABE-2352 ........ ................
-
-2010-06-28 15:36 +0000 [r272685-272686] Russell Bryant <russell@digium.com>
-
- * doc/tex/chan-mobile.tex (removed): remove accidentally added
- file.
-
- * doc/tex/cdrdriver.tex, doc/tex/asterisk.tex, /,
- doc/tex/chan-mobile.tex (added): Merged revisions 272684 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r272684 | russell | 2010-06-28 10:33:32 -0500 (Mon, 28
- Jun 2010) | 2 lines Use the underscore package so that
- underscores do not need to be escaped. ........
-
-2010-06-25 20:20 +0000 [r272556-272577] Tilghman Lesher <tlesher@digium.com>
-
- * /, doc/voicemail_odbc_postgresql.txt: Merged revisions 272568 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r272568 | tilghman | 2010-06-25 15:18:47 -0500
- (Fri, 25 Jun 2010) | 12 lines Merged revisions 272562 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r272562 | tilghman | 2010-06-25 15:17:37 -0500 (Fri, 25 Jun 2010)
- | 5 lines Make the structure of the table specified before match
- the queries and results. (closes issue #17557) Reported by: cmaj
- ........ ................
-
- * sounds/Makefile, /: Merged revisions 272533 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r272533 |
- tilghman | 2010-06-25 14:17:16 -0500 (Fri, 25 Jun 2010) | 2 lines
- Symlink sounds files, to save disk space, when multiple
- tarballs/checkouts are on the same system. ........
-
-2010-06-25 18:58 +0000 [r272531] Russell Bryant <russell@digium.com>
-
- * include/asterisk/_private.h, tests/test_sched.c, main/asterisk.c,
- include/asterisk/test.h (added), build_tools/cflags-devmode.xml,
- tests/test_heap.c, tests/test_skel.c, main/Makefile, main/test.c
- (added): Backport unit test API from trunk. Also, update existing
- test modules that were already in this branch but had been
- converted to the unit test API in trunk. Review:
- https://reviewboard.asterisk.org/r/748/
-
-2010-06-24 22:19 +0000 [r272459] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 272447 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r272447 | rmudgett | 2010-06-24 17:11:26 -0500
- (Thu, 24 Jun 2010) | 17 lines Merged revisions 272446 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r272446 | rmudgett | 2010-06-24 16:58:49 -0500 (Thu, 24 Jun 2010)
- | 10 lines ss_thread calls pri_grab without lock during overlap
- dial Recent changes to chan_dahdi with relation to overlap
- dialing call pri_grab without first obtaining a lock. (closes
- issue #17414) Reported by: pdf Patches: bug17414.patch uploaded
- by jpeeler (license 325) ........ ................
-
-2010-06-23 23:40 +0000 [r272440] Terry Wilson <twilson@digium.com>
-
- * autoconf/ast_ext_tool_check.m4, /, configure: Merged revisions
- 272254,272256 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r272254 |
- twilson | 2010-06-23 15:53:48 -0500 (Wed, 23 Jun 2010) | 10 lines
- Honor the --with-${library}=path for AST_EXT_TOOL_CHECK (closes
- issue #16991) Reported by: pprindeville Patches:
- with_netsnmp.patch.txt uploaded by twilson (license 396) Tested
- by: twilson Review: https://reviewboard.asterisk.org/r/739/
- ........ r272256 | twilson | 2010-06-23 15:59:17 -0500 (Wed, 23
- Jun 2010) | 2 lines Update configure when changing autconf m4
- files... ........
-
-2010-06-23 23:14 +0000 [r272371] Russell Bryant <russell@digium.com>
-
- * channels/chan_iax2.c, /: Merged revisions 272370 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r272370 | russell | 2010-06-23 18:09:28 -0500 (Wed, 23 Jun 2010)
- | 23 lines Resolve some errors produced during module unload of
- chan_iax2. The external test suite stops Asterisk using the "core
- stop gracefully" command. The logs from the tests show that there
- are a number of problems with Asterisk trying to cleanly shut
- down. This patch addresses the following type of error that comes
- from chan_iax2: [Jun 22 16:58:11] ERROR[29884]: lock.c:129
- __ast_pthread_mutex_destroy: chan_iax2.c line 11371
- (iax2_process_thread_cleanup): Error destroying mutex
- &thread->lock: Device or resource busy For an example in the
- context of a build, see:
- http://bamboo.asterisk.org/browse/AST-TRUNK-739/log The primary
- purpose of this patch is to change the thread pool shutdown
- procedure to be more explicit to ensure that the thread exits
- from a point where it is not holding a lock. While testing that,
- I encountered various crashes due to the order of operations in
- unload_module() being problematic. I reordered some things there,
- as well. Review: https://reviewboard.asterisk.org/r/736/ ........
-
-2010-06-23 22:37 +0000 [r272369] Matthew Nicholson <mnicholson@digium.com>
-
- * apps/app_queue.c, /: Merged revisions 272368 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r272368 | mnicholson | 2010-06-23 17:36:49 -0500 (Wed, 23 Jun
- 2010) | 16 lines Merged revisions 272367 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 This version
- of the patch only adds AgentComplete for attended transfers. It
- was already present for blind transfers. ........ r272367 |
- mnicholson | 2010-06-23 17:33:51 -0500 (Wed, 23 Jun 2010) | 8
- lines Send AgentComplete manager events in the event of blind and
- attended transfers. (closes issue #16819) Reported by: elbriga
- Patches: app_queue.diff uploaded by elbriga (license 482)
- ........ ................
-
-2010-06-23 21:54 +0000 [r272333] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_musiconhold.c, /: Merged revisions 272332 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r272332 | tilghman | 2010-06-23 16:53:49 -0500 (Wed, 23 Jun 2010)
- | 8 lines If there is realtime configuration, it does not get
- re-read on reload unless the config file also changes. (closes
- issue #16982) Reported by: dmitri Patches: res_musiconhold.patch
- uploaded by dmitri (license 1001) Tested by: atis ........
-
-2010-06-23 21:15 +0000 [r272263] Paul Belanger <paul.belanger@polybeacon.com>
-
- * apps/app_meetme.c: Revert previous commit, ast_test_flag64 does
- not exist in 1.6.2
-
-2010-06-23 21:09 +0000 [r272262] Tilghman Lesher <tlesher@digium.com>
-
- * res/ael/ael.flex, /, res/ael/ael.tab.c, res/ael/ael.y,
- res/ael/ael_lex.c: Merged revisions 272260 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r272260 |
- tilghman | 2010-06-23 16:06:40 -0500 (Wed, 23 Jun 2010) | 8 lines
- Ensure a NULL file while debugging cannot crash AEL. (closes
- issue #17215) Reported by: vazir Patches:
- 20100518__issue17215.diff.txt uploaded by tilghman (license 14)
- Tested by: tilghman ........
-
-2010-06-23 21:07 +0000 [r272253-272261] Paul Belanger <paul.belanger@polybeacon.com>
-
- * /, apps/app_meetme.c: Merged revisions 272259 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r272259 |
- pabelanger | 2010-06-23 17:06:15 -0400 (Wed, 23 Jun 2010) | 2
- lines Fix previous merge. ast_test_flag != ast_test_flag64
- ........
-
- * /, apps/app_meetme.c: Merged revisions 272257 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r272257 | pabelanger | 2010-06-23 17:00:00 -0400 (Wed, 23 Jun
- 2010) | 19 lines Merged revisions 272255 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r272255 | pabelanger | 2010-06-23 16:57:01 -0400 (Wed, 23 Jun
- 2010) | 12 lines First caller into a dynamic conference now enter
- pin once. If MeetMe is configured to use dynamic conference
- numbers, then the first caller (which creates the conference) had
- to enter the PIN number twice. (closes issue #15878) Reported by:
- shawkris Patches: issue15878.patch uploaded by pabelanger
- (license 224) Tested by: pabelanger ........ ................
-
- * main/manager.c, /: Merged revisions 272252 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r272252 |
- pabelanger | 2010-06-23 16:35:45 -0400 (Wed, 23 Jun 2010) | 8
- lines Correct manager variable 'EventList' case. (closes issue
- #17520) Reported by: kobaz Patches: manager.patch uploaded by
- kobaz (license 834) Tested by: lmadsen ........
-
-2010-06-23 18:41 +0000 [r272124-272149] Terry Wilson <twilson@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 272146 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r272146 |
- twilson | 2010-06-23 13:39:20 -0500 (Wed, 23 Jun 2010) | 2 lines
- Don't start the sla thread unless we realy need it ........
-
- * /, apps/app_meetme.c: Merged revisions 272109 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r272109 |
- twilson | 2010-06-23 12:21:40 -0500 (Wed, 23 Jun 2010) | 12 lines
- Make sure reload updates SLA config Even if there are no stations
- or trunks defined, we need to start the sla thread to make sure
- we get the reload event. Also, when doing a reload we need to
- remove the existing trunks and stations or they end up hanging
- around. (closes issue #16818) Reported by: mbonin Patches:
- sla_reload.patch uploaded by twilson (license 396) Tested by:
- twilson ........
-
-2010-06-22 22:14 +0000 [r272015] David Vossel <dvossel@digium.com>
-
- * pbx/pbx_config.c, /: Merged revisions 272014 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r272014 |
- dvossel | 2010-06-22 17:11:50 -0500 (Tue, 22 Jun 2010) | 5 lines
- fixes issue with 'dialplan remove extension blah' segfaulting
- with tab completion (closes issue #17440) Reported by: kobaz
- ........
-
-2010-06-22 17:37 +0000 [r271904] Matthew Nicholson <mnicholson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 271903 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r271903 | mnicholson | 2010-06-22 12:35:17 -0500 (Tue, 22 Jun
- 2010) | 15 lines Merged revisions 271902 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r271902 | mnicholson | 2010-06-22 12:31:57 -0500 (Tue, 22 Jun
- 2010) | 8 lines Decrease the module ref count in sip_hangup when
- SIP_DEFER_BYE_ON_TRANSFER is set. This is necessary to keep the
- ref count correct. (closes issue #16815) Reported by: rain
- Patches: chan_sip-unref-fix.diff uploaded by rain (license 327)
- (modified) Tested by: rain ........ ................
-
-2010-06-22 16:30 +0000 [r271869] Russell Bryant <russell@digium.com>
-
- * /, res/ais/clm.c, res/ais/evt.c: Merged revisions 271867 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r271867 | russell | 2010-06-22 11:28:03 -0500 (Tue, 22
- Jun 2010) | 7 lines Resolve some errors that occur on a graceful
- shutdown. Don't Finalize() if Initialize() did not succeed. This
- resulted in an error about trying to Finalize() an invalid
- handle. Also trim some trailing whitespace while in the area.
- ........
-
-2010-06-22 15:49 +0000 [r271832] David Vossel <dvossel@digium.com>
-
- * /, main/features.c: Merged revisions 271831 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r271831 |
- dvossel | 2010-06-22 10:46:22 -0500 (Tue, 22 Jun 2010) | 10 lines
- fixes attended transfer behavior when both transferee and
- transferer hung up If both the transferer and transferee of a
- attended transfer hangup before the new channel picks up, the new
- channel should be hung up as well as it has no endpoint to talk
- to. This mirrors the expected behavior used in 1.4. (closes issue
- #17444) Reported by: corruptor ........
-
-2010-06-22 15:00 +0000 [r271691-271763] Matthew Nicholson <mnicholson@digium.com>
-
- * configs/dundi.conf.sample, /, pbx/pbx_dundi.c: Merged revisions
- 271762 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r271762 | mnicholson | 2010-06-22 09:54:58 -0500 (Tue, 22 Jun
- 2010) | 15 lines Merged revisions 271761 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r271761 | mnicholson | 2010-06-22 09:49:36 -0500 (Tue, 22 Jun
- 2010) | 9 lines Allow users to specify a port for dundi peers.
- (closes issue #17056) Reported by: klaus3000 Patches:
- dundi-peerport-patch-trunk.txt uploaded by klaus3000 (license 65)
- Tested by: klaus3000 ........ ................
-
- * include/asterisk/strings.h, configs/sip_notify.conf.sample, /,
- channels/chan_sip.c: Merged revisions 271690 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r271690 | mnicholson | 2010-06-22 07:58:28 -0500 (Tue, 22 Jun
- 2010) | 18 lines Merged revisions 271689 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r271689 | mnicholson | 2010-06-22 07:52:27 -0500 (Tue, 22 Jun
- 2010) | 8 lines Modify chan_sip's packet generation api to
- automatically calculate the Content-Length. This is done by
- storing packet content in a buffer until it is actually time to
- send the packet, at which time the size of the packet is
- calculated. This change was made to ensure that the
- Content-Length is always correct. (closes issue #17326) Reported
- by: kenner Tested by: mnicholson, kenner Review:
- https://reviewboard.asterisk.org/r/693/ ........ This change also
- adds an ast_str_copy_string() function (similar to
- ast_copy_string), that copies one ast_str into another, properly
- handling embedded nulls. ................
-
-2010-06-21 20:48 +0000 [r271555] Jeff Peeler <jpeeler@digium.com>
-
- * res/ael/pval.c, /: Merged revisions 271554 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r271554 | jpeeler | 2010-06-21 15:46:53 -0500 (Mon, 21 Jun 2010)
- | 14 lines Merged revisions 271552 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r271552 | jpeeler | 2010-06-21 15:37:47 -0500 (Mon, 21 Jun 2010)
- | 7 lines Do not use sizeof to calculate size of a heap allocated
- character array. Change left out from 271399. (closes issue
- #16053) Reported by: diLLec ........ ................
-
-2010-06-18 21:33 +0000 [r271338-271484] Jeff Peeler <jpeeler@digium.com>
-
- * res/ael/pval.c, /, include/asterisk/pval.h, pbx/pbx_ael.c: Merged
- revisions 271483 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r271483 | jpeeler | 2010-06-18 16:32:09 -0500 (Fri, 18 Jun 2010)
- | 18 lines Merged revisions 271399 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r271399 | jpeeler | 2010-06-18 14:28:24 -0500 (Fri, 18 Jun 2010)
- | 11 lines Fix crash when parsing some heavily nested statements
- in AEL on reload. Due to the recursion used when compiling AEL in
- gen_prios, all the stack space was being consumed when parsing
- some AEL that contained nesting 13 levels deep. Changing a few
- large buffers to be heap allocated fixed the crash, although I
- did not test how many more levels can now be safely used. (closes
- issue #16053) Reported by: diLLec Tested by: jpeeler ........
- ................
-
- * channels/chan_dahdi.c, /: Merged revisions 269307 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r269307 | rmudgett | 2010-06-09 11:54:38 -0500 (Wed, 09 Jun 2010)
- | 12 lines Eliminate deadlock potential in dahdi_fixup(). Calling
- dahdi_indicate() within dahdi_fixup() while the owner pointers
- are in a potentially inconsistent state is a potentially bad
- thing in principle. However, calling dahdi_indicate() when the
- channel private lock is already held can cause a deadlock if the
- PRI lock is needed because dahdi_indicate() will also get the
- channel private lock. The pri_grab() function assumes that the
- channel private lock is held once to avoid deadlock. ........
-
-2010-06-17 Leif Madsen <lmadsen@digium.com>
-
- * Asterisk 1.6.2.9 Released.
-
-2010-06-10 Leif Madsen <lmadsen@digium.com>
-
- * Asterisk 1.6.2.9-rc3 Released.
-
-2010-06-10 Tilghman Lesher <tlesher@digium.com>
-
- * Ensure signals are not blocked inside other signal handlers.
-
- This eliminates the annoying <beep> on the console.
-
- (closes issue 0017477)
- Reported by: jvandal
- Patches:
- 20100610__issue17477.diff.txt uploaded by tilghman (license 14
-
-2010-06-09 Paul Belanger <paul.belanger@polybeacon.com>
-
- * Fix Debian init script to not use -c.
-
- When using the init script as-is currently, it could cause issues on Debian
- such as high CPU usage. This fix has worked for several people so I'm
- implementing the change. We now handle color displays properly.
-
- (closes issue 0016784)
- Reported by: pabelanger
- Patches:
- 20100530__issue16784__2.diff.txt uploaded by tilghman (license 14)
- Tested by: pabelanger, tilghman
-
-2010-06-07 Leif Madsen <lmadsen@digium.com>
-
- * Asterisk 1.6.2.9-rc2 Released.
-
-2010-06-07 Tilghman Lesher <tlesher@digium.com>
-
- * Fix crash in DTMF detection.
-
- What I did not originally see in my previous commit was that even
- though the next digit could be detected before the previous was
- considered ended, the detection of the next digit effectively ends
- the detection of the previous. Therefore, the length moves in
- lockstep with the digit, and no separate counter is needed for the
- length alone.
-
- (closes issue 0017371)
- Reported by: alecdavis
-
- (closes issue 0017474)
- Reported by: kenner
-
-2010-06-01 Leif Madsen <lmadsen@digium.com>
-
- * Asterisk 1.6.2.9-rc1 Released.
-
-2010-06-01 15:20 +0000 [r266598] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c, /: Merged revisions 266592 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r266592 | tilghman | 2010-06-01 10:18:59 -0500 (Tue, 01 Jun 2010)
- | 18 lines Merged revisions 266585 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r266585 | tilghman | 2010-06-01 10:17:46 -0500 (Tue, 01 Jun 2010)
- | 11 lines Prevent CLI prompt from distorting output of lines
- shorter than the prompt. Uses the VT100 method of clearing the
- line from the cursor position to the end of the line: Esc-0K
- (closes issue #17160) Reported by: coolmig Patches:
- 20100531__issue17160.diff.txt uploaded by tilghman (license 14)
- Tested by: coolmig ........ ................
-
-2010-05-31 16:07 +0000 [r266570] Paul Belanger <paul.belanger@polybeacon.com>
-
- * res/res_agi.c: Fix typo in documentation (closes issue #17395)
- Reported by: pabelanger Patches: res_agi.c.patch uploaded by
- pabelanger (license 224)
-
-2010-05-30 04:45 +0000 [r266439] Tilghman Lesher <tlesher@digium.com>
-
- * contrib/init.d/rc.debian.asterisk, /: Merged revisions 266438 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r266438 | tilghman | 2010-05-29 23:44:28 -0500
- (Sat, 29 May 2010) | 9 lines Merged revisions 266437 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r266437 | tilghman | 2010-05-29 23:43:28 -0500 (Sat, 29
- May 2010) | 2 lines Reverting patch and reopening issue #16784,
- as patch breaks color display. ........ ................
-
-2010-05-28 20:55 +0000 [r266338] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c, /: Merged revisions 266337 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r266337 |
- tilghman | 2010-05-28 15:53:04 -0500 (Fri, 28 May 2010) | 1 line
- Only report swap on platforms which can examine those statistics
- ........
-
-2010-05-28 17:57 +0000 [r266293] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 266292 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r266292 |
- dvossel | 2010-05-28 12:55:38 -0500 (Fri, 28 May 2010) | 9 lines
- fixes crash when creation of UDPTL fails (closes issue #17264)
- Reported by: falves11 Patches: issue_17264_reviewboard_fix.diff
- uploaded by dvossel (license 671)
- issue_17264_1.6.2_reviewboard_fix.diff uploaded by dvossel
- (license 671) Tested by: falves11 ........
-
-2010-05-26 21:19 +0000 [r266154] Tilghman Lesher <tlesher@digium.com>
-
- * utils/extconf.c, main/asterisk.c, /, main/logger.c: Merged
- revisions 266146 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r266146 | tilghman | 2010-05-26 16:17:46 -0500 (Wed, 26 May 2010)
- | 21 lines Merged revisions 266142 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r266142 | tilghman | 2010-05-26 16:11:44 -0500 (Wed, 26 May 2010)
- | 14 lines Use sigaction for signals which should persist past
- the initial trigger, not signal. If you call signal() in a
- Solaris signal handler, instead of just resetting the signal
- handler, it causes the signal to refire, because the signal is
- not marked as handled prior to the signal handler being called.
- This effectively causes Solaris to immediately exceed the
- threadstack in recursive signal handlers and crash. (closes issue
- #17000) Reported by: rmcgilvr Patches:
- 20100526__issue17000.diff.txt uploaded by tilghman (license 14)
- Tested by: rmcgilvr ........ ................
-
-2010-05-26 18:37 +0000 [r266007] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 266006 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r266006 |
- dvossel | 2010-05-26 13:32:51 -0500 (Wed, 26 May 2010) | 8 lines
- fixes failed SIP Directed pickup resulting in dead channel
- (closes issue #17339) Reported by: one47 Patches:
- sip_magic_pickup2 uploaded by one47 (license 23) Tested by:
- one47, dvossel ........
-
-2010-05-26 16:31 +0000 [r265895-265959] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_config_pgsql.c, /: Merged revisions 265923 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r265923 | tilghman | 2010-05-26 11:23:28 -0500
- (Wed, 26 May 2010) | 14 lines Merged revisions 265910 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r265910 | tilghman | 2010-05-26 11:21:00 -0500 (Wed, 26 May 2010)
- | 7 lines Not finding rows in the DB does not rise to the level
- of a warning. (closes issue #17062) Reported by: drookie Patches:
- 20100525__issue17062.diff.txt uploaded by tilghman (license 14)
- ........ ................
-
- * configs/res_pgsql.conf.sample, res/res_config_pgsql.c, /: Merged
- revisions 265894 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r265894 |
- tilghman | 2010-05-26 11:14:48 -0500 (Wed, 26 May 2010) | 8 lines
- Construct socket name, according to the Postgres docs, and
- document as such. (closes issue #17392) Reported by: dps Patches:
- 20100525__issue17392.diff.txt uploaded by tilghman (license 14)
- Tested by: dps ........
-
-2010-05-26 15:52 +0000 [r265890] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Recorded merge of revisions 265842 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r265842 | mmichelson | 2010-05-26 09:41:55 -0500 (Wed,
- 26 May 2010) | 9 lines Re-enable "always" option for videosupport
- option in sip.conf. (closes issue #17016) Reported by: twilson
- Patches: 17016.patch uploaded by mmichelson (license 60) Tested
- by: devmod ........
-
-2010-05-26 00:33 +0000 [r265748] Tilghman Lesher <tlesher@digium.com>
-
- * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
- pbx/pbx_lua.c: Merged revisions 265747 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r265747 |
- tilghman | 2010-05-25 19:29:40 -0500 (Tue, 25 May 2010) | 8 lines
- Use configure to determine the prefixes and include directories
- properly. This ensures cross-platform compatibility, even among
- Linux distributions, which don't always put headers in the same
- place. (closes issue #17391) Reported by: loloski ........
-
-2010-05-25 21:05 +0000 [r265699] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 265698 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r265698 |
- mmichelson | 2010-05-25 15:59:04 -0500 (Tue, 25 May 2010) | 12
- lines Properly use peer's outboundproxy for outbound REGISTERs.
- The logic used in transmit_register to get the outboundproxy for
- a peer was flawed since this value would be overridden shortly
- afterwards when create_addr was called. In addition, this also
- fixes some logic used when parsing users.conf so that the peer
- name is placed in the internally-generated register string so
- that an outboundproxy set in the Asterisk GUI will be used for
- outbound REGISTERs. ........
-
-2010-05-25 17:15 +0000 [r265615] David Vossel <dvossel@digium.com>
-
- * channels/chan_dahdi.c: fixes build issue with zaptel (closes
- issue 0017394) Reported by: aragon Patches: half_buffer_fix.diff
- uploaded by dvossel (license 671) Tested by: aragon
-
-2010-05-25 17:06 +0000 [r265612] Matthew Nicholson <mnicholson@digium.com>
-
- * apps/app_queue.c, /: Merged revisions 265611 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r265611 | mnicholson | 2010-05-25 12:00:11 -0500 (Tue, 25 May
- 2010) | 15 lines Merged revisions 265610 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r265610 | mnicholson | 2010-05-25 11:48:19 -0500 (Tue, 25 May
- 2010) | 8 lines Don't mark the cdr records of unanswered queue
- calls with "NOANSWER". This restores the behavior prior to
- r258670. (closes issue #17334) Reported by: jvandal Patches:
- queue-cdr-fixes1.diff uploaded by mnicholson (license 96) Tested
- by: aragon, jvandal ........ ................
-
-2010-05-24 23:52 +0000 [r265521] Terry Wilson <twilson@digium.com>
-
- * include/asterisk/options.h, main/asterisk.c, Makefile,
- doc/manager_1_1.txt, doc/tex/manager.tex, main/manager.c: Merged
- revisions 265320,265467 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r265320 |
- twilson | 2010-05-24 14:06:40 -0500 (Mon, 24 May 2010) | 14 lines
- Add the FullyBooted AMI event It is possible to connect to the
- manager interface before all Asterisk modules are loaded. To
- ensure that an application does not send AMI actions that might
- require a module that has not yet loaded, the application can
- listen for the FullyBooted manager event. It will be sent upon
- connection if all modules have been loaded, or as soon as loading
- is complete. The event: Event: FullyBooted Privilege: system,all
- Status: Fully Booted Review:
- https://reviewboard.asterisk.org/r/639/ ........ r265467 |
- twilson | 2010-05-24 17:21:58 -0500 (Mon, 24 May 2010) | 1 line
- Merge the rest of the FullyBooted patch ........
-
-2010-05-24 22:07 +0000 [r265450-265452] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/h323/ast_h323.cxx: Merged revisions 265451 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r265451 | mmichelson | 2010-05-24 17:05:15 -0500 (Mon,
- 24 May 2010) | 8 lines Print openh323 log to the Asterisk
- console. (closes issue #17109) Reported by: under Patches:
- logstream.diff uploaded by under (license 914) ........
-
- * /, channels/chan_sip.c: Merged revisions 265449 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r265449 |
- mmichelson | 2010-05-24 16:44:30 -0500 (Mon, 24 May 2010) | 11
- lines Allow type=user SIP endpoints to be loaded properly from
- realtime. (closes issue #16021) Reported by: Guggemand Patches:
- realtime-type-fix.patch uploaded by Guggemand (license 897)
- (altered by me slightly to avoid ref leaks) Tested by: Guggemand
- ........
-
-2010-05-24 19:30 +0000 [r265364] David Vossel <dvossel@digium.com>
-
- * main/channel.c, /: Merged revisions 265273 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r265273 |
- dvossel | 2010-05-24 11:10:09 -0500 (Mon, 24 May 2010) | 2 lines
- fixes segfault when using generic plc ........
-
-2010-05-24 18:30 +0000 [r265318] Tilghman Lesher <tlesher@digium.com>
-
- * main/asterisk.c, /: Merged revisions 265316 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r265316 |
- tilghman | 2010-05-24 13:19:08 -0500 (Mon, 24 May 2010) | 7 lines
- On systems with a LOT of RAM, a signed integer sometimes printed
- negative. (closes issue #16837) Reported by: jlpedrosa Patches:
- 20100504__issue16837.diff.txt uploaded by tilghman (license 14)
- ........
-
-2010-05-21 21:57 +0000 [r264998-265172] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_queue.c: Fix memory hogging behavior of app_queue. From
- reviewboard: This review request is for the patch on issue 17081.
- A user reported that he saw increasing numbers of allocations
- stemming from app_queue.c when he would run the "queue show" CLI
- command. The user reported that he was using approximately 40
- realtime queues and as he ran the CLI command more and more, the
- memory usage would shoot up. As it turns out, there was a memory
- leak and a separate usage of memory that, while not really a
- leak, was very irresponsible. Both memory problems can be
- attributed to the function init_queue(). When the "queue show"
- command is run, all realtime queues have the init_queue()
- function called on the in-memory queue. The idea is to place the
- queue in its default state and then overwrite options specified
- in the realtime backend as we read them. The first problem, the
- memory leak, had to do with the fact that the string field for
- the name of the first periodic announcement file was being
- re-created every time init_queue was called. This patch corrects
- the behavior by only calling ast_str_create if the memory has not
- already been allocated. The other problem is a bit more
- complicated. The majority of the strings in the call_queue
- structure were changed to use the ast_string_fields API for 1.6.0
- and beyond. init_queue resets all string fields on the queue to
- their default values. Then, later in the realtime queue loading
- process, these string fields are set to their configured values.
- For those unfamiliar with string fields, frequent resizing of a
- string like this is not what the string fields API is designed
- for. The result of this constant resizing is that as the queue
- gets loaded, eventually space for the string runs out and so a
- new memory pool, at twice the size of the previously allocated
- one, is created for the string fields. The reporter of issue
- 17081 wrote a script that ran the "queue show" CLI command 2100
- times. By the end, each of his 40 queues was taking about a
- megabyte of memory apiece just for their string fields. My fix
- for this problem is to revert the call_queue structure from using
- string fields. In my patch here, I have moved the queue back to
- using fixed-sized buffers. I ran the script provided by the
- reporter of 17081 and determined that I no longer saw the
- steadily-increasing memory usage that I had seen before applying
- the patch. (closes issue #17081) Reported by: wliegel Patches:
- 17081v2.patch uploaded by mmichelson (license 60) Tested by:
- wliegel, mmichelson Review:
- https://reviewboard.asterisk.org/r/651/
-
- * apps/app_queue.c, include/asterisk/file.h, /: Merged revisions
- 265090 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r265090 | mmichelson | 2010-05-21 16:08:51 -0500 (Fri, 21 May
- 2010) | 15 lines Merged revisions 265089 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r265089 | mmichelson | 2010-05-21 15:59:14 -0500 (Fri, 21 May
- 2010) | 8 lines Don't hang up on a queue caller if the file we
- attempt to play does not exist. This also fixes a documentation
- mistake in file.h that made my original attempt to correct this
- problem not work correctly. (closes issue #17061) Reported by:
- RoadKill ........ ................
-
- * /, channels/chan_sip.c: Merged revisions 265087 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r265087 |
- mmichelson | 2010-05-21 15:38:14 -0500 (Fri, 21 May 2010) | 7
- lines Be sure to set the sin_family on the proxy when allocating.
- (closes issue #17157) Reported by: stuarth ........
-
- * /, include/asterisk/channel.h: Merged revisions 265000 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r265000 | mmichelson | 2010-05-21 11:54:21 -0500
- (Fri, 21 May 2010) | 9 lines Merged revisions 264999 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r264999 | mmichelson | 2010-05-21 11:53:53 -0500 (Fri,
- 21 May 2010) | 3 lines Fix grammatical error in comment. ........
- ................
-
- * main/channel.c, main/autoservice.c, /,
- include/asterisk/channel.h: Merged revisions 264997 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r264997 | mmichelson | 2010-05-21 11:44:27 -0500
- (Fri, 21 May 2010) | 38 lines Merged revisions 264996 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r264996 | mmichelson | 2010-05-21 11:28:34 -0500 (Fri, 21 May
- 2010) | 32 lines Allow ast_safe_sleep to defer specific frames
- until after the sleep has concluded. From reviewboard Background:
- A Digium customer discovered a somewhat odd bug. The setup is
- that parties A and B are bridged, and party A places party B on
- hold. While party B is listening to hold music, he mashes a bunch
- of DTMF. Party A takes party B off hold while this is happening,
- but party B continues to hear hold music. I could reproduce this
- about 1 in 5 times. The issue: When DTMF features are enabled and
- a user presses keys, the channel that the DTMF is streamed to is
- placed in an ast_safe_sleep for 100 ms, the duration of the
- emulated tone. If an AST_CONTROL_UNHOLD frame is read from the
- channel during the sleep, the frame is dropped. Thus the unhold
- indication is never made to the channel that was originally
- placed on hold. The fix: Originally, I discussed with Kevin
- possible ways of fixing the specific problem reported. However,
- we determined that the same type of problem could happen in other
- situations where ast_safe_sleep() is used. Using autoservice as a
- model, I modified ast_safe_sleep_conditional() to defer specific
- frame types so they can be re-queued once the sleep has finished.
- I made a common function for determining if a frame should be
- deferred so that there are not two identical switch blocks to
- maintain. Review: https://reviewboard.asterisk.org/r/674/
- ........ ................
-
-2010-05-20 23:34 +0000 [r264829] Richard Mudgett <rmudgett@digium.com>
-
- * /, main/callerid.c: Merged revisions 264828 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r264828 | rmudgett | 2010-05-20 18:29:43 -0500 (Thu, 20 May 2010)
- | 13 lines Merged revisions 264820 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r264820 | rmudgett | 2010-05-20 18:23:21 -0500 (Thu, 20 May 2010)
- | 6 lines ast_callerid_parse() had a path that left name
- uninitialized. Several callers of ast_callerid_parse() do not
- initialize the name parameter before calling thus there is the
- potential to use an uninitialized pointer. ........
- ................
-
-2010-05-20 22:24 +0000 [r264753-264783] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c, /: Merged revisions 264779 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r264779 |
- tilghman | 2010-05-20 17:23:32 -0500 (Thu, 20 May 2010) | 8 lines
- Let ExtensionState resolve dynamic hints. (closes issue #16623)
- Reported by: tilghman Patches: 20100116__issue16623.diff.txt
- uploaded by tilghman (license 14) Tested by: lmadsen ........
-
- * apps/app_stack.c, /: Merged revisions 264752 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r264752 |
- tilghman | 2010-05-20 16:28:53 -0500 (Thu, 20 May 2010) | 7 lines
- Error message fix. (closes issue #17356) Reported by: kenner
- Patches: app_stack.c.diff uploaded by kenner (license 1040)
- ........
-
-2010-05-19 22:10 +0000 [r264453] Mark Michelson <mmichelson@digium.com>
-
- * include/asterisk/_private.h, include/asterisk/options.h,
- main/asterisk.c, main/loader.c, main/channel.c, /,
- channels/chan_sip.c: Merged revisions 264452 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r264452 |
- mmichelson | 2010-05-19 16:29:08 -0500 (Wed, 19 May 2010) | 86
- lines Fix transcode_via_sln option with SIP calls and improve PLC
- usage. From reviewboard: The problem here is a bit complex, so
- try to bear with me... It was noticed by a Digium customer that
- generic PLC (as configured in codecs.conf) did not appear to
- actually be having any sort of benefit when packet loss was
- introduced on an RTP stream. I reproduced this issue myself by
- streaming a file across an RTP stream and dropping approx. 5% of
- the RTP packets. I saw no real difference between when PLC was
- enabled or disabled when using wireshark to analyze the RTP
- streams. After analyzing what was going on, it became clear that
- one of the problems faced was that when running my tests, the
- translation paths were being set up in such a way that PLC could
- not possibly work as expected. To illustrate, if packets are lost
- on channel A's read stream, then we expect that PLC will be
- applied to channel B's write stream. The problem is that generic
- PLC can only be done when there is a translation path that moves
- from some codec to SLINEAR. When I would run my tests, I found
- that every single time, read and write translation paths would be
- set up on channel A instead of channel B. There appeared to be no
- real way to predict which channel the translation paths would be
- set up on. This is where Kevin swooped in to let me know about
- the transcode_via_sln option in asterisk.conf. It is supposed to
- work by placing a read translation path on both channels from the
- channel's rawreadformat to SLINEAR. It also will place a write
- translation path on both channels from SLINEAR to the channel's
- rawwriteformat. Using this option allows one to predictably set
- up translation paths on all channels. There are two problems with
- this, though. First and foremost, the transcode_via_sln option
- did not appear to be working properly when I was placing a SIP
- call between two endpoints which did not share any common
- formats. Second, even if this option were to work, for PLC to be
- applied, there had to be a write translation path that would go
- from some format to SLINEAR. It would not work properly if the
- starting format of translation was SLINEAR. The one-line change
- presented in this review request in chan_sip.c fixed the first
- issue for me. The problem was that in sip_request_call, the
- jointcapability of the outbound channel was being set to the
- format passed to sip_request_call. This is nativeformats of the
- inbound channel. Because of this, when
- ast_channel_make_compatible was called by app_dial, both channels
- already had compatibly read and write formats. Thus, no
- translation path was set up at the time. My change is to set the
- jointcapability of the sip_pvt created during sip_request_call to
- the intersection of the inbound channel's nativeformats and the
- configured peer capability that we determined during the earlier
- call to create_addr. Doing this got the translation paths set up
- as expected when using transcode_via_sln. The changes presented
- in channel.c fixed the second issue for me. First and foremost,
- when Asterisk is started, we'll read codecs.conf to see the value
- of the genericplc option. If this option is set, and ast_write is
- called for a frame with no data, then we will attempt to fill in
- the missing samples for the frame. The implementation uses a
- channel datastore for maintaining the PLC state and for creating
- a buffer to store PLC samples in. Even when we receive a frame
- with data, we'll call plc_rx so that the PLC state will have
- knowledge of the previous voice frame, which it can use as a
- basis for when it comes time to actually do a PLC fill-in. So,
- reviewers, now I ask for your help. First off, there's the one
- line change in chan_sip that I have put in. Is it right? By my
- logic it seems correct, but I'm sure someone can tell me why it
- is not going to work. This is probably the change I'm least
- concerned about, though. What concerns me much more is the set of
- changes in channel.c. First off, am I even doing it right? When I
- run tests, I can clearly see that when PLC is activated, I see a
- significant increase in RTP traffic where I would expect it to
- be. However, in my humble opinion, the audio sounds kind of
- crappy whenever the PLC fill-in is done. It sounds worse to me
- than when no PLC is used at all. I need someone to review the
- logic I have used to be sure that I'm not misusing anything. As
- far as I can see my pointer arithmetic is correct, and my use of
- AST_FRIENDLY_OFFSET should be correct as well, but I'm sure
- someone can point out somewhere where I've done something
- incorrectly. As I was writing this review request up, I decided
- to give the code a test run under valgrind, and I find that for
- some reason, calls to plc_rx are causing some invalid reads.
- Apparently I'm reading past the end of a buffer somehow. I'll
- have to dig around a bit to see why that is the case. If it's
- obvious to someone reviewing, speak up! Finally, I have one other
- proposal that is not reflected in my code review. Since without
- transcode_via_sln set, one cannot predict or control where a
- translation path will be up, it seems to me that the current
- practice of using PLC only when transcoding to SLINEAR is not
- useful. I recommend that once it has been determined that the
- method used in this code review is correct and works as expected,
- then the code in translate.c that invokes PLC should be removed.
- Review: https://reviewboard.asterisk.org/r/622/ ........
-
-2010-05-19 20:31 +0000 [r264405] David Vossel <dvossel@digium.com>
-
- * main/udptl.c, /: Merged revisions 264400 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r264400 |
- dvossel | 2010-05-19 15:30:33 -0500 (Wed, 19 May 2010) | 11 lines
- fixes infinite loop during udptl.c's decode_open_type When
- decode_length returns the length there is a check to see if that
- length is negative, if so the decode loop breaks as this means
- the limit has been reached. The problem here is that length is an
- unsigned int, so length can never be negative. This resulted in
- an infinite loop. (issue #17352) ........
-
-2010-05-19 20:27 +0000 [r264336-264388] Matthew Nicholson <mnicholson@digium.com>
-
- * main/udptl.c, /: Merged revisions 264379 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r264379 |
- mnicholson | 2010-05-19 15:26:27 -0500 (Wed, 19 May 2010) | 4
- lines Cast an unsigned int to a signed int when comparing it with
- 0. (AST-377) ........
-
- * apps/app_speech_utils.c, /: Merged revisions 264335 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r264335 | mnicholson | 2010-05-19 15:02:57 -0500
- (Wed, 19 May 2010) | 12 lines Merged revisions 264334 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r264334 | mnicholson | 2010-05-19 15:01:38 -0500 (Wed, 19 May
- 2010) | 5 lines Set quieted flag when receiving a dtmf tone
- during playback in speechbackground. (closes issue #16966)
- Reported by: asackheim ........ ................
-
-2010-05-19 19:25 +0000 [r264332] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 264331 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r264331 |
- dvossel | 2010-05-19 14:21:04 -0500 (Wed, 19 May 2010) | 13 lines
- fixes crash in check_rtp_timeout During deadlock avoidance the
- sip dialog pvt is locked and unlocked. When this occurs we have
- no guarantee the pvt's owner is still valid. We were trying to
- access the pvt's owner after this without checking to see if it
- still existed first. (closes issue #17271) Reported by: under
- Patches: check_rtp_timeout.diff uploaded by under (license 914)
- Tested by: dvossel ........
-
-2010-05-19 17:49 +0000 [r264205-264250] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/options.h, /, configure,
- include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
- 264249 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r264249 | tilghman | 2010-05-19 12:48:31 -0500 (Wed, 19 May 2010)
- | 24 lines Merged revisions 264248 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r264248 | tilghman | 2010-05-19 12:41:29 -0500 (Wed, 19 May 2010)
- | 17 lines Internal timing is now on by default, if you're using
- DAHDI 2.3 or above. The reason for ensuring DAHDI 2.3 or above is
- that this version ensures that a timer is always available,
- whereas in previous versions, it was possible for DAHDI to be
- loaded, but have no drivers to actually generate timing. If
- internal_timing was turned on in this circumstance, a complete
- lack of audio would result. This is the reason why
- internal_timing was not on by default. However, now that DAHDI
- ensures the availability of a timer, there is no reason for this
- setting to be off (and in fact, it solves a great many initial
- user problems). (closes issue #15932) Reported by: dimas Patches:
- 20100519__issue15932.diff.txt uploaded by tilghman (license 14)
- Tested by: tilghman ........ ................
-
- * main/dsp.c, /: Merged revisions 264204 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r264204 |
- tilghman | 2010-05-19 11:42:20 -0500 (Wed, 19 May 2010) | 9 lines
- Keep track of digit duration, when we're decoding inband to pass
- DTMF frames. (closes issue #17235) Reported by: frawd Patches:
- new_dtmf_dsp_len.patch uploaded by frawd (license 610)
- 20100518__issue17235.diff.txt uploaded by tilghman (license 14)
- Tested by: frawd ........
-
-2010-05-19 14:47 +0000 [r264115] David Vossel <dvossel@digium.com>
-
- * main/rtp.c, /: Merged revisions 264114 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r264114 |
- dvossel | 2010-05-19 09:38:02 -0500 (Wed, 19 May 2010) | 13 lines
- fixes crash during dtmf During the processing of Cisco dtmf the
- dtmf samples were not being calculated correctly. In an attempt
- to determine what sample rate was being used, a NULL frame was
- processed which caused a crash. This patch resolves this. (closes
- issue #17248) Reported by: falves11 Patches: issue_17248.diff
- uploaded by dvossel (license 671) ........
-
-2010-05-19 08:15 +0000 [r264032] Alec L Davis <sivad.a@paradise.net.nz>
-
- * /, configs/indications.conf.sample: Merged revisions 264031 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r264031 | alecdavis | 2010-05-19 20:09:14 +1200 (Wed, 19
- May 2010) | 8 lines fix incorrectly typed indications for [nz]
- stutter and dialrecall (closes issue #17359) Reported by:
- alecdavis Patches: bug17359.diff.txt uploaded by alecdavis
- (license 585) ........
-
-2010-05-19 06:41 +0000 [r263951] Tilghman Lesher <tlesher@digium.com>
-
- * main/dsp.c, /: Merged revisions 263950 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r263950 | tilghman | 2010-05-19 01:41:04 -0500 (Wed, 19 May 2010)
- | 15 lines Merged revisions 263949 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r263949 | tilghman | 2010-05-19 01:32:27 -0500 (Wed, 19 May 2010)
- | 8 lines Because progress is called multiple times, across
- several frames, we must persist states when detecting multitone
- sequences. (closes issue #16749) Reported by: dant Patches:
- dsp.c-bug16749-1.patch uploaded by dant (license 670) Tested by:
- dant ........ ................
-
-2010-05-18 22:49 +0000 [r263906] David Vossel <dvossel@digium.com>
-
- * main/strings.c, /: Merged revisions 263904 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r263904 |
- dvossel | 2010-05-18 17:48:51 -0500 (Tue, 18 May 2010) | 9 lines
- fixes segfault on logging (closes issue #17331) Reported by:
- under Patches: utils.diff uploaded by under (license 914)
- segfault_on_logging.diff uploaded by dvossel (license 671) Tested
- by: under, dvossel ........
-
-2010-05-18 19:41 +0000 [r263809] Jeff Peeler <jpeeler@digium.com>
-
- * apps/app_directory.c, /: Merged revisions 263807 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r263807 | jpeeler | 2010-05-18 14:27:34 -0500
- (Tue, 18 May 2010) | 17 lines Merged revisions 263769 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r263769 | jpeeler | 2010-05-18 13:54:58 -0500 (Tue, 18 May 2010)
- | 10 lines Modify directory name reading to be interrupted with
- operator or pound escape. In the case of accidentally entering
- the wrong first three letters for the reading, users could be
- very frustrated if the name listing is very long. This allows
- interrupting the reading by pressing 0 or #. 0 will attempt to
- execute a configured operator (o) extension and # will exit and
- proceed in the dialplan. ABE-2200 ........ ................
-
-2010-05-17 22:10 +0000 [r263642] Mark Michelson <mmichelson@digium.com>
-
- * /, main/devicestate.c: Merged revisions 263640 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r263640 | mmichelson | 2010-05-17 17:08:01 -0500 (Mon, 17 May
- 2010) | 16 lines Merged revisions 263639 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r263639 | mmichelson | 2010-05-17 17:00:28 -0500 (Mon, 17 May
- 2010) | 10 lines Fix logic error when checking for a devstate
- provider. When using strsep, if one of the list of specified
- separators is not found, it is the first parameter to strsep
- which is now NULL, not the pointer returned by strsep. This issue
- isn't especially severe in that the worst it is likely to do is
- waste some cycles when a device with no '/' and no ':' is passed
- to ast_device_state. ........ ................
-
-2010-05-17 19:37 +0000 [r263587-263590] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 263589 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r263589 | tilghman | 2010-05-17 14:31:15 -0500 (Mon, 17 May 2010)
- | 9 lines With IMAP backend, messages in INBOX were counted twice
- for MWI. (closes issue #17135) Reported by: edhorton Patches:
- 20100513__issue17135.diff.txt uploaded by tilghman (license 14)
- 17135_2.diff uploaded by ebroad (license 878) Tested by:
- edhorton, ebroad ........
-
- * main/app.c: Don't close 'n', just close 'above_n'. (closes issue
- #17345) Reported by: wdoekes
-
-2010-05-17 14:41 +0000 [r263376-263458] Leif Madsen <lmadsen@digium.com>
-
- * main/manager.c, /: Merged revisions 263457 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r263457 | lmadsen | 2010-05-17 09:37:35 -0500 (Mon, 17 May 2010)
- | 19 lines Recorded merge of revisions 263456 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r263456 | lmadsen | 2010-05-17 09:35:18 -0500 (Mon, 17 May 2010)
- | 11 lines Manager cookies are not compatible with RFC2109. The
- Version field in the cookies we're setting contain quotes around
- the version number which is not compatible with RFC2109 and
- breaks some implementations. (closes issue #17231) Reported by:
- ecarruda Patches: manager_rfc2109-trunk-v1.patch uploaded by
- ecarruda (license 559) manager_rfc2109-1.6.2-v1.patch uploaded by
- ecarruda (license 559) Tested by: ecarruda, russell ........
- ................
-
- * sounds/Makefile, /: Merged revisions 263375 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r263375 | lmadsen | 2010-05-17 09:05:33 -0500 (Mon, 17 May 2010)
- | 16 lines Merged revisions 263374 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r263374 | lmadsen | 2010-05-17 09:04:57 -0500 (Mon, 17 May 2010)
- | 8 lines Update link to new version of core sounds. The latest
- version of the core sounds files 1.4.19 now includes the missing
- queue-minute sound file which is called by app_queue but which
- has been missing. (closes issue #17123) Reported by: n8ideas
- ........ ................
-
-2010-05-17 13:03 +0000 [r263293] David Vossel <dvossel@digium.com>
-
- * CHANGES, channels/chan_dahdi.c: backport of DAHDI dynamic buffer
- policy dialstring option
-
-2010-05-15 23:41 +0000 [r263202] Paul Belanger <paul.belanger@polybeacon.com>
-
- * /, codecs/gsm/Makefile: Merged revisions 252488 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r252488 |
- tilghman | 2010-03-15 12:27:08 -0400 (Mon, 15 Mar 2010) | 9 lines
- Make the Makefile logic more explicit and move the Snow Leopard
- logic down to where it's not executed on non-Darwin systems.
- (closes issue #17028) Reported by: pabelanger Patches:
- issue17028_20100315.patch uploaded by seanbright (license 71)
- 20100315__issue17028.diff.txt uploaded by tilghman (license 14)
- Tested by: tilghman, pabelanger ........
-
-2010-05-13 22:13 +0000 [r263070] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 263069 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r263069 | rmudgett | 2010-05-13 17:01:36 -0500 (Thu, 13 May 2010)
- | 1 line Fix inverted logic in cli command: ss7 set debug on/off
- ........
-
-2010-05-13 15:36 +0000 [r262898] Russell Bryant <russell@digium.com>
-
- * channels/chan_console.c, /: Merged revisions 262897 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r262897 | russell | 2010-05-13 10:36:12 -0500 (Thu, 13 May 2010)
- | 4 lines Fix an off by one error that causes a crash. Thanks to
- Raymond Burke for pointing it out. ........
-
-2010-05-12 20:01 +0000 [r262801] Paul Belanger <paul.belanger@polybeacon.com>
-
- * main/loader.c, main/cli.c, /: Merged revisions 262800 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r262800 | pabelanger | 2010-05-12 15:59:16 -0400 (Wed,
- 12 May 2010) | 8 lines Notify CLI when modules is loaded /
- unloaded (closes issue #17308) Reported by: pabelanger Patches:
- cli.modules.patch uploaded by pabelanger (license 224) Tested by:
- pabelanger, russell ........
-
-2010-05-12 19:53 +0000 [r262797-262799] Leif Madsen <lmadsen@digium.com>
-
- * res/ael/pval.c, /: Merged revisions 262798 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r262798 |
- lmadsen | 2010-05-12 14:53:10 -0500 (Wed, 12 May 2010) | 7 lines
- Revert previous WARNING message removal. Marquis42 suggested a
- better method of doing what I wanted because I ended up removing
- the WARNING message for all instances when really I just wanted
- to remove it for the 'return' keyword, not everything. (issue
- #17145) ........
-
- * res/ael/pval.c, /: Merged revisions 262796 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r262796 |
- lmadsen | 2010-05-12 14:31:42 -0500 (Wed, 12 May 2010) | 4 lines
- Remove unnecessary WARNING message in ael/pval.c (closes issue
- #17145) Reported by: okrief ........
-
-2010-05-12 18:03 +0000 [r262746] David Vossel <dvossel@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 262744 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r262744 | dvossel | 2010-05-12 13:01:20 -0500 (Wed, 12 May 2010)
- | 17 lines Merged revisions 262662 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r262662 | dvossel | 2010-05-12 12:00:04 -0500 (Wed, 12 May 2010)
- | 11 lines fixes app_meetme dsp error We attempted to detect
- silence after translating a frame from signed linear. This caused
- a flooding of errors. To resolve this the code to detect silence
- was moved before the translation. (closes issue #17133) Reported
- by: jsdyer ........ ................
-
-2010-05-12 16:29 +0000 [r262516-262659] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_privacy.c: Merged revisions 262656 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r262656 |
- tilghman | 2010-05-12 11:23:26 -0500 (Wed, 12 May 2010) | 8 lines
- Ensure the arguments are initialized. Also miscellaneous CG
- cleanup. (closes issue #16576) Reported by: uxbod Patches:
- 20100505__issue16576.diff.txt uploaded by tilghman (license 14)
- Tested by: uxbod ........
-
- * /, include/asterisk/causes.h: Merged revisions 262513 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r262513 | tilghman | 2010-05-11 16:25:05 -0500 (Tue, 11
- May 2010) | 7 lines Move cause 200 to cause 26, as specified in
- Q.850. Also cleanup the formatting and add a few more that seem
- like good candidates. (closes issue #16157) Reported by: wimpy
- ........
-
-2010-05-11 19:58 +0000 [r262425] Jason Parker <jparker@digium.com>
-
- * /, res/Makefile: Merged revisions 262422 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r262422 | qwell | 2010-05-11 14:57:24 -0500 (Tue, 11 May 2010) |
- 18 lines Merged revisions 262421 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r262421 | qwell | 2010-05-11 14:55:42 -0500 (Tue, 11 May 2010) |
- 11 lines Use a less silly method for modifying a flex-generated
- file. The sed syntax that was used wasn't actually valid, causing
- some versions to choke. This is the method that is used in 1.6.x+
- for similar changes. (closes issue #16696) Reported by: bklang
- Patches: 16696-sedfix.diff uploaded by qwell (license 4) Tested
- by: qwell ........ ................
-
-2010-05-11 19:41 +0000 [r262415-262420] Paul Belanger <paul.belanger@polybeacon.com>
-
- * pbx/pbx_config.c, /: Merged revisions 262419 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r262419 |
- pabelanger | 2010-05-11 15:40:37 -0400 (Tue, 11 May 2010) | 8
- lines Improve logging by displaying line number (closes issue
- #16303) Reported by: dant Patches: issue16303.patch.v2 uploaded
- by pabelanger (license 224) Tested by: dant, lmadsen, pabelanger
- ........
-
- * /, channels/chan_sip.c: Merged revisions 262414 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r262414 |
- pabelanger | 2010-05-11 15:26:17 -0400 (Tue, 11 May 2010) | 8
- lines Improve logging information for misconfigured contexts
- (closes issue #17238) Reported by: pprindeville Patches:
- chan_sip-bug17238.patch uploaded by pprindeville (license 347)
- Tested by: pprindeville ........
-
-2010-05-11 17:25 +0000 [r262340] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c, /, Makefile.rules: Merged revisions 262330
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r262330 | tilghman | 2010-05-11 12:23:51 -0500
- (Tue, 11 May 2010) | 9 lines Merged revisions 262321 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r262321 | tilghman | 2010-05-11 12:22:07 -0500 (Tue, 11
- May 2010) | 2 lines Fix issue #17302 a slightly different way
- (mad props to Qwell) ........ ................
-
-2010-05-10 19:06 +0000 [r262237-262241] David Vossel <dvossel@digium.com>
-
- * /, apps/app_directed_pickup.c: Merged revisions 262240 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r262240 | dvossel | 2010-05-10 14:06:08 -0500 (Mon, 10
- May 2010) | 9 lines fixes PickupChan application (closes issue
- #16863) Reported by: schern Patches: app_directed_pickup.c.patch
- uploaded by schern (license 995) for_trunk.diff uploaded by
- cjacobsen (license 1029) Tested by: Graber, cjacobsen, lathama,
- rickead2000, dvossel ........
-
- * channels/chan_console.c, /: Merged revisions 262236 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r262236 | dvossel | 2010-05-10 13:36:10 -0500 (Mon, 10 May 2010)
- | 11 lines fixes crash in chan_console There is a race condition
- between console_hangup() and start_stream(). It is possible for
- console_hangup() to be called and then the stream thread to begin
- after the hangup. To avoid this a check in start_stream() to make
- sure the pvt-owner still exists while the pvt lock is held is
- made. If the owner is gone that means the channel hung up and
- start_stream should be aborted. ........
-
-2010-05-10 16:39 +0000 [r262155] Tilghman Lesher <tlesher@digium.com>
-
- * /, Makefile.rules: Merged revisions 262152 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r262152 | tilghman | 2010-05-10 11:36:25 -0500 (Mon, 10 May 2010)
- | 17 lines Merged revisions 262151 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r262151 | tilghman | 2010-05-10 11:34:21 -0500 (Mon, 10 May 2010)
- | 10 lines Allow compilation on Mac OS X 10.4 (Tiger) (closes
- issue #17297) Reported by: jcovert Patches:
- 20100506__issue17297.diff.txt uploaded by tilghman (license 14)
- (closes issue #17302) Reported by: jcovert ........
- ................
-
-2010-05-09 02:17 +0000 [r261916-262105] Tilghman Lesher <tlesher@digium.com>
-
- * autoconf/ast_ext_lib.m4, autoconf/ast_c_compile_check.m4,
- autoconf/ast_c_define_check.m4, /, configure,
- include/asterisk/autoconfig.h.in: Merged revisions 262102 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r262102 | tilghman | 2010-05-08 21:14:04 -0500 (Sat, 08
- May 2010) | 5 lines Cleanup a bit more by getting rid of useless
- version defines. Also make library detection use passed CFLAGS.
- (closes issue #17309) Reported by: stuarth ........
-
- * /, configure, configure.ac: Merged revisions 262048 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r262048 | tilghman | 2010-05-07 21:40:01 -0500 (Fri, 07 May 2010)
- | 2 lines Use CPPFLAGS to pass PTHREAD_CFLAGS for vpb only
- ........
-
- * /, funcs/func_odbc.c: Merged revisions 261917 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r261917 |
- tilghman | 2010-05-07 15:54:35 -0500 (Fri, 07 May 2010) | 8 lines
- Double free crash (closes issue #17245) Reported by:
- thedavidfactor Patches: 20100426__issue17245.diff.txt uploaded by
- tilghman (license 14) Tested by: murraytm ........
-
- * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
- Merged revisions 261913 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r261913 |
- tilghman | 2010-05-07 15:35:17 -0500 (Fri, 07 May 2010) | 14
- lines Use the detected pthread building flags in every place,
- instead of hardcoding -lpthread. We nicely detect the right flags
- on each system for building Asterisk with pthreads, then ignore
- it for every other build option that requires us to build with
- pthreads. This caused some items to return a false negative. Also
- cleanup some minor naming issues that caused "library library"
- redundancy in the output. (closes issue #17303) Reported by:
- stuarth Patches: 20100507__issue17303.diff.txt uploaded by
- tilghman (license 14) Tested by: stuarth ........
-
-2010-05-07 16:08 +0000 [r261868] Leif Madsen <lmadsen@digium.com>
-
- * UPGRADE-1.6.txt, /: Merged revisions 261867 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r261867 |
- lmadsen | 2010-05-07 11:05:24 -0500 (Fri, 07 May 2010) | 6 lines
- Update UPGRADE-1.6.txt stating insecure=very has been removed.
- (closes issue #17282) Reported by: stuarth Tested by: stuarth
- ........
-
-2010-05-06 20:13 +0000 [r261739] Jeff Peeler <jpeeler@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 261736 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r261736 | jpeeler | 2010-05-06 15:11:53 -0500
- (Thu, 06 May 2010) | 15 lines Merged revisions 261735 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r261735 | jpeeler | 2010-05-06 15:10:59 -0500 (Thu, 06 May 2010)
- | 8 lines Only allow the operator key to be accepted after
- leaving a voicemail. Or rather disallow the operator key from
- being accepted when not offered, such as after finishing a
- recording from within the mailbox options menu. ABE-2121 SWP-1267
- ........ ................
-
-2010-05-06 17:08 +0000 [r261612] Jason Parker <jparker@digium.com>
-
- * sounds/Makefile, /: Merged revisions 261609 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r261609 | qwell | 2010-05-06 12:06:40 -0500 (Thu, 06 May 2010) |
- 11 lines Merged revisions 261608 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r261608 | qwell | 2010-05-06 11:56:02 -0500 (Thu, 06 May 2010) |
- 4 lines Use the versioned MOH tarballs, now that we have them.
- This makes for more reproducibility. Prompted by a discussion in
- #asterisk-dev ........ ................
-
-2010-05-06 15:43 +0000 [r261563] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 261560 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r261560 |
- tilghman | 2010-05-06 10:39:10 -0500 (Thu, 06 May 2010) | 8 lines
- Permit more lines within a SIP body to be parsed. The example
- given within the related issue showed 120 lines, which was mostly
- a result of the body being XML. (closes issue #17179) Reported
- by: khw ........
-
-2010-06-01 Leif Madsen <lmadsen@digium.com>
-
- * Asterisk 1.6.2.8 Released.
-
-2010-05-26 Leif Madsen <lmadsen@digium.com>
-
- * Asterisk 1.6.2.8-rc2 Released.
-
-2010-05-26 10:56 -0500 [r265891] Matt Nicholson <mnicholson@digium.com>
-
- * Merged r265610 from 1.4:
-
- Don't mark the cdr records of unanswered queue calls with "NOANSWER".
- This restores the behavior prior to r258670.
-
- (closes issue #17334)
- Reported by: jvandal
- Patches:
- queue-cdr-fixes1.diff uploaded by mnicholson (license 96)
- Tested by: aragon, jvandal
-
-2010-05-06 Leif Madsen <lmadsen@digium.com>
-
- * Asterisk 1.6.2.8-rc1 Released
-
-2010-05-06 14:07 +0000 [r261498-261499] Russell Bryant <russell@digium.com>
-
- * tests/test_heap.c: Add test case that ensures the heap handles
- arbitrary removals properly. (issue #17277) Reported by:
- cappucinoking Patches: test_heap.diff uploaded by cappucinoking
- (license 1036) Tested by: cappucinoking, russell
-
- * /, main/heap.c: Merged revisions 261496 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r261496 |
- russell | 2010-05-06 08:58:07 -0500 (Thu, 06 May 2010) | 40 lines
- Fix handling of removing nodes from the middle of a heap. This
- bug surfaced in 1.6.2 and does not affect code in any other
- released version of Asterisk. It manifested itself as SIP qualify
- not happening when it should, causing peers to go unreachable.
- This was debugged down to scheduler entries sometimes not getting
- executed when they were supposed to, which was in turn caused by
- an error in the heap code. The problem only sometimes occurs, and
- it is due to the logic for removing an entry in the heap from an
- arbitrary location (not just popping off the top). The scheduler
- performs this operation frequently when entries are removed
- before they run (when ast_sched_del() is used). In a normal pop
- off of the top of the heap, a node is taken off the bottom,
- placed at the top, and then bubbled down until the max heap
- property is restored (see max_heapify()). This same logic was
- used for removing an arbitrary node from the middle of the heap.
- Unfortunately, that logic is full of fail. This patch fixes that
- by fully restoring the max heap property when a node is thrown
- into the middle of the heap. Instead of just pushing it down as
- appropriate, it first pushes it up as high as it will go, and
- _then_ pushes it down. Lastly, fix a minor problem in
- ast_heap_verify(), which is only used for debugging. If a parent
- and child node have the same value, that is not an error. The
- only error is if a parent's value is less than its children. A
- huge thanks goes out to cappucinoking for debugging this down to
- the scheduler, and then producing an ast_heap test case that
- demonstrated the breakage. That made it very easy for me to focus
- on the heap logic and produce a fix. Open source projects are
- awesome. (closes issue #16936) Reported by: ib2 Tested by:
- cappucinoking, crjw (closes issue #17277) Reported by:
- cappucinoking Patches: heap-fix.rev2.diff uploaded by russell
- (license 2) Tested by: cappucinoking, russell ........
-
-2010-05-06 07:43 +0000 [r261453] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 261451 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r261451 | tzafrir | 2010-05-06 10:27:31 +0300 (ה', 06 מאי 2010) |
- 4 lines When failing to configure, don't destroy 'cfg' twice
- Fixes a crash when some config section had an incorrect channel
- config. ........
-
-2010-05-05 19:08 +0000 [r261233-261315] Paul Belanger <paul.belanger@polybeacon.com>
-
- * /, channels/chan_sip.c: Merged revisions 261314 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r261314 | pabelanger | 2010-05-05 14:43:03 -0400 (Wed, 05 May
- 2010) | 19 lines Merged revisions 261274 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r261274 | pabelanger | 2010-05-05 12:42:22 -0400 (Wed, 05 May
- 2010) | 12 lines Registration fix for SIP realtime. Make sure
- realtime fields are not empty. (closes issue #17266) Reported by:
- Nick_Lewis Patches: chan_sip.c-realtime.patch uploaded by Nick
- Lewis (license 657) Tested by: Nick_Lewis, sberney Review:
- https://reviewboard.asterisk.org/r/643/ ........ ................
-
- * apps/app_queue.c, /: Merged revisions 261232 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r261232 |
- pabelanger | 2010-05-05 11:42:07 -0400 (Wed, 05 May 2010) | 10
- lines 'queue reset stats' erroneously clears wrapuptime
- configuration. Resets each member's lastcall to 0 now. (closes
- issue #17262, #16519) Reported by: rain Patches:
- wrapuptime_reset_fix.diff uploaded by rain (license 327) Tested
- by: rain ........
-
-2010-05-04 23:55 +0000 [r261098] Tilghman Lesher <tlesher@digium.com>
-
- * main/channel.c, /: Merged revisions 261095 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r261095 | tilghman | 2010-05-04 18:51:52 -0500 (Tue, 04 May 2010)
- | 18 lines Merged revisions 261093-261094 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r261093 | tilghman | 2010-05-04 18:36:53 -0500 (Tue, 04 May 2010)
- | 7 lines Protect against overflow, when calculating how long to
- wait for a frame. (closes issue #17128) Reported by: under
- Patches: d.diff uploaded by under (license 914) ........ r261094
- | tilghman | 2010-05-04 18:47:08 -0500 (Tue, 04 May 2010) | 2
- lines Add a tiny corner case to the previous commit ........
- ................
-
-2010-05-04 19:01 +0000 [r260927] Jeff Peeler <jpeeler@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 260924 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r260924 | jpeeler | 2010-05-04 13:51:28 -0500
- (Tue, 04 May 2010) | 18 lines Merged revisions 260923 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r260923 | jpeeler | 2010-05-04 13:46:46 -0500 (Tue, 04 May 2010)
- | 12 lines Voicemail transfer to operator should occur
- immediately, not after main menu. There were two scenarios in the
- advanced options that while using the operator=yes and review=yes
- options, the transfer occurred only after exiting the main menu
- (after sending a reply or leaving a message for an extension).
- Now after the audio is processed for the reply or message the
- transfer occurs immediately as expected. ABE-2107 ABE-2108
- ........ ................
-
-2010-05-04 16:58 +0000 [r260884] Matthew Nicholson <mnicholson@digium.com>
-
- * configs/sip.conf.sample, include/asterisk/frame.h,
- main/channel.c, /, channels/chan_sip.c: Merged revisions 254450
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r254450 | kpfleming | 2010-03-25 10:27:31 -0500 (Thu, 25
- Mar 2010) | 49 lines Improve handling of T.38 re-INVITEs that
- arrive before a T.38-capable application is executing on a
- channel. This patch addresses an issue found during working with
- end-users using res_fax. If an incoming call is answered in the
- dialplan, or jumps to the 'fax' extension due to reception of a
- CNG tone (with faxdetect enabled), and then the remote endpoint
- sends a T.38 re-INVITE, it is possible for the channel's T.38
- state to be 'T38_STATE_NEGOTIATING' when the application starts
- up. Unfortunately, even if the application wants to use T.38, it
- can't respond to the peer's negotiation request, because the
- AST_CONTROL_T38_PARAMETERS control frame that chan_sip sent
- originally has been lost, and the application needs the content
- of that frame to be able to formulate a reply. This patch adds a
- new 'request' type to AST_CONTROL_T38_PARAMETERS,
- AST_T38_REQUEST_PARMS. If the application sends this request,
- chan_sip will re-send the original control frame (with
- AST_T38_REQUEST_NEGOTIATE as the request type), and the
- application can respond as normal. If this occurs within the five
- second timeout in chan_sip, the automatic cancellation of the
- peer reinvite will be stopped, and the application will 'own' the
- negotiation process from that point onwards. This also improves
- the code path in chan_sip to allow sip_indicate(), when called
- for AST_CONTROL_T38_PARAMETERS, to be able to return a non-zero
- response, which should have been in place before since the
- control frame *can* fail to be processed properly. It also
- modifies ast_indicate() to return whatever result the channel
- driver returned for this control frame, rather than converting
- all non-zero results into '-1'. Finally, the new request type
- intentionally returns a positive value, so that an application
- that sends AST_T38_REQUEST_PARMS can know for certain whether the
- channel driver accepted it and will be replying with a control
- frame of its own, or whether it was ignored (if the
- sip_indicate()/ast_indicate() path had properly supported failure
- responses before, this would not be necessary). This patch also
- modifies res_fax to take advantage of the new request. In
- addition, this patch makes sip_t38_abort() actually lock the
- private structure before doing its work... bad programmer, no
- donut. This patch also enhances chan_sip's 'faxdetect' support to
- allow triggering on T.38 re-INVITEs received as well as CNG tone
- detection. Review: https://reviewboard.asterisk.org/r/556/
- ........
-
-2010-05-04 15:51 +0000 [r260746-260805] Jason Parker <jparker@digium.com>
-
- * /, build_tools/make_build_h: Merged revisions 260802 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r260802 | qwell | 2010-05-04 10:49:57 -0500
- (Tue, 04 May 2010) | 9 lines Merged revisions 260801 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r260801 | qwell | 2010-05-04 10:49:27 -0500 (Tue, 04 May
- 2010) | 1 line Fix fallout from removing from configure script.
- Pointed out by philipp64 on #asterisk-dev ........
- ................
-
- * /: Fix merge props
-
-2010-05-03 17:42 +0000 [r260743] Paul Belanger <paul.belanger@polybeacon.com>
-
- * Makefile, /: Merged revisions 260661-260662 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r260661 | pabelanger | 2010-05-03 12:41:30 -0400 (Mon, 03 May
- 2010) | 10 lines non-root make install PREFIX=/tmp fails. Prepend
- libdir when executing mkpkgconfig allowing non-root installs to
- work. (closes issue #17268) Reported by: pabelanger Patches:
- issue17268.patch uploaded by pabelanger (license 224) Tested by:
- pabelanger ........ r260662 | pabelanger | 2010-05-03 12:54:41
- -0400 (Mon, 03 May 2010) | 3 lines Should have removed /usr/lib/
- part. Thanks Qwell. ........
-
-2010-05-03 14:59 +0000 [r260571] Leif Madsen <lmadsen@digium.com>
-
- * doc/HOWTO_collect_debug_information.txt: Merged revisions 260570
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r260570 | lmadsen | 2010-05-03 09:58:23 -0500
- (Mon, 03 May 2010) | 9 lines Merged revisions 260569 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r260569 | lmadsen | 2010-05-03 09:57:39 -0500 (Mon, 03
- May 2010) | 1 line Minor typo pointed out by pabelanger on IRC.
- ........ ................
-
-2010-04-30 22:48 +0000 [r260441] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 260437 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r260437 | jpeeler | 2010-04-30 17:36:49 -0500
- (Fri, 30 Apr 2010) | 18 lines Merged revisions 260434 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r260434 | jpeeler | 2010-04-30 17:22:46 -0500 (Fri, 30 Apr 2010)
- | 11 lines Ensure channel state is not incorrectly set in the
- case of a very early answer. The needringing bit was being read
- in dahdi_read after answering thereby setting the state to
- ringing from up. This clears needringing upon answering so that
- is no longer possible. (closes issue #17067) Reported by: tzafrir
- Patches: needringing.diff uploaded by tzafrir (license 46)
- ........ ................
-
-2010-04-30 20:22 +0000 [r260373] Mark Michelson <mmichelson@digium.com>
-
- * res/res_musiconhold.c, /: Merged revisions 260346 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r260346 | mmichelson | 2010-04-30 15:11:02 -0500
- (Fri, 30 Apr 2010) | 24 lines Merged revisions 260345 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r260345 | mmichelson | 2010-04-30 15:08:15 -0500 (Fri, 30 Apr
- 2010) | 18 lines Fix potential crash from race condition due to
- accessing channel data without the channel locked. In
- res_musiconhold.c, there are several places where a channel's
- stream's existence is checked prior to calling ast_closestream on
- it. The issue here is that in several cases, the channel was not
- locked while checking the stream. The result was that if two
- threads checked the state of the channel's stream at
- approximately the same time, then there could be a situation
- where both threads attempt to call ast_closestream on the
- channel's stream. The result here is that the refcount for the
- stream would go below 0, resulting in a crash. I have added
- proper channel locking to res_musiconhold.c to ensure that we do
- not try to check chan->stream without the channel locked. A
- Digium customer has been using this patch for several weeks and
- has not had any crashes since applying the patch. ABE-2147
- ........ ................
-
-2010-04-30 06:22 +0000 [r260281-260303] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/app.c: Merged revisions 260292 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r260292 |
- tilghman | 2010-04-30 01:19:35 -0500 (Fri, 30 Apr 2010) | 13
- lines Don't allow file descriptors to go above 64k, when we're
- closing them in a fork(2). This saves time, when, even though the
- system allows the process limit to be that high, the practical
- limit is much lower. (closes issue #17223) Reported by:
- dbackeberg Patches: 20100423__issue17223.diff.txt uploaded by
- tilghman (license 14) Tested by: dbackeberg ........
-
- * configs/extensions.conf.sample, /: Merged revisions 260280 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r260280 | tilghman | 2010-04-30 00:23:56 -0500 (Fri, 30
- Apr 2010) | 7 lines Logic fixups for a sample FREENUM dialplan
- context. (closes issue #17263) Reported by: pprindeville Patches:
- freenum-dialplan.patch#3 uploaded by pprindeville (license 347)
- ........
-
-2010-04-29 23:13 +0000 [r260234] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 260231 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r260231 | rmudgett | 2010-04-29 17:44:14 -0500
- (Thu, 29 Apr 2010) | 33 lines Merged revisions 260195 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r260195 | rmudgett | 2010-04-29 17:11:47 -0500 (Thu, 29 Apr 2010)
- | 26 lines DTMF CallerID detection problems. The code handling
- DTMF CallerID drops digits on long CallerID numbers and may
- timeout waiting for the first ring with shorter numbers. The DTMF
- emulation mode was not turned off when processing DTMF CallerID.
- When the emulation code gets behind in processing the DTMF digits
- it can skip a digit. For shorter numbers, the timeout may have
- been too short. I increased it from 2 seconds to 4 seconds. Four
- seconds is a typical time between rings for many countries.
- (closes issue #16460) Reported by: sum Patches: issue16460.patch
- uploaded by rmudgett (license 664) issue16460_v1.6.2.patch
- uploaded by rmudgett (license 664) Tested by: sum, rmudgett
- Review: https://reviewboard.asterisk.org/r/634/ JIRA SWP-562 JIRA
- AST-334 JIRA SWP-901 ........ ................
-
-2010-04-29 18:18 +0000 [r260156] Tilghman Lesher <tlesher@digium.com>
-
- * configs/extensions.conf.sample, /: Merged revisions 260148 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r260148 | tilghman | 2010-04-29 13:15:57 -0500 (Thu, 29
- Apr 2010) | 2 lines Pattern match fail. ........
-
-2010-04-29 15:35 +0000 [r260051] David Vossel <dvossel@digium.com>
-
- * main/audiohook.c, /, include/asterisk/audiohook.h: Merged
- revisions 260050 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r260050 | dvossel | 2010-04-29 10:33:27 -0500 (Thu, 29 Apr 2010)
- | 21 lines Merged revisions 260049 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r260049 | dvossel | 2010-04-29 10:31:02 -0500 (Thu, 29 Apr 2010)
- | 14 lines Fixes crash in audiohook_write_list The middle_frame
- in the audiohook_write_list function was being freed if a
- audiohook manipulator returned a failure. This is incorrect
- logic. This patch resolves this and adds detailed descriptions of
- how this function should work and why manipulator failures must
- be ignored. (closes issue #17052) Reported by: dvossel Tested by:
- dvossel (closes issue #16196) Reported by: atis Review:
- https://reviewboard.asterisk.org/r/623/ ........ ................
-
-2010-04-28 22:36 +0000 [r259959] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 259957 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r259957 |
- mmichelson | 2010-04-28 17:34:15 -0500 (Wed, 28 Apr 2010) | 11
- lines Don't override peer context with domain context. (closes
- issue #17040) Reported by: pprindeville Patches:
- asterisk-1.6-bugid17040.patch uploaded by pprindeville (license
- 347) Tested by: pprindeville Review:
- https://reviewboard.asterisk.org/r/565/ ........
-
-2010-04-28 21:26 +0000 [r259899] David Vossel <dvossel@digium.com>
-
- * main/channel.c, channels/chan_local.c, /: Merged revisions 259870
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r259870 | dvossel | 2010-04-28 16:20:03 -0500
- (Wed, 28 Apr 2010) | 39 lines Merged revisions 259858 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r259858 | dvossel | 2010-04-28 16:16:03 -0500 (Wed, 28 Apr 2010)
- | 33 lines resolves deadlocks in chan_local Issue_1. In the
- local_hangup() 3 locks must be held at the same time... pvt,
- pvt->chan, and pvt->owner. Proper deadlock avoidance is done when
- the channel to hangup is the outbound chan_local channel, but
- when it is not the outbound channel we have an issue... We
- attempt to do deadlock avoidance only on the tech pvt, when both
- the tech pvt and the pvt->owner are locked coming into that loop.
- By never giving up the pvt->owner channel deadlock avoidance is
- not entirely possible. This patch resolves that by doing deadlock
- avoidance on both the pvt->owner and the pvt when trying to get
- the pvt->chan lock. Issue_2. ast_prod() is used in
- ast_activate_generator() to queue a frame on the channel and make
- the channel's read function get called. This function is used in
- ast_activate_generator() while the channel is locked, which
- mean's the channel will have a lock both from the generator code
- and the frame_queue code by the time it gets to chan_local.c's
- local_queue_frame code... local_queue_frame contains some of the
- same crazy deadlock avoidance that local_hangup requires, and
- this recursive lock prevents that deadlock avoidance from
- happening correctly. This patch removes ast_prod() from the
- channel lock so only one lock is held during the
- local_queue_frame function. (closes issue #17185) Reported by:
- schmoozecom Patches: issue_17185_v1.diff uploaded by dvossel
- (license 671) issue_17185_v2.diff uploaded by dvossel (license
- 671) Tested by: schmoozecom, GameGamer43 Review:
- https://reviewboard.asterisk.org/r/631/ ........ ................
-
-2010-04-28 21:09 +0000 [r259854] Leif Madsen <lmadsen@digium.com>
-
- * config.guess: Merged revisions 259853 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r259853 | lmadsen | 2010-04-28 16:08:34 -0500 (Wed, 28 Apr 2010)
- | 14 lines Merged revisions 259852 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r259852 | lmadsen | 2010-04-28 16:07:48 -0500 (Wed, 28 Apr 2010)
- | 6 lines Update config.guess. Updating config.guess because
- after installing Ubuntu Server 9.10 and running all the update
- scripts, running ./configure would not continue because it was
- unable to determine what kind of system I had. After updating
- config.guess things started working again. ........
- ................
-
-2010-04-28 20:34 +0000 [r259781-259851] Jason Parker <jparker@digium.com>
-
- * /, configure, configure.ac: Merged revisions 259848 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r259848 | qwell | 2010-04-28 15:32:14 -0500
- (Wed, 28 Apr 2010) | 9 lines Merged revisions 259847 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r259847 | qwell | 2010-04-28 15:30:21 -0500 (Wed, 28 Apr
- 2010) | 1 line Add AC_CONFIG_AUX_DIR to configure script, so
- systems without install can use install-sh from our source dir.
- ........ ................
-
- * makeopts.in, /: Merged revisions 259837 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r259837 | qwell | 2010-04-28 15:26:35 -0500 (Wed, 28 Apr 2010) |
- 9 lines Merged revisions 259833 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r259833 | qwell | 2010-04-28 15:25:36 -0500 (Wed, 28 Apr 2010) |
- 1 line Missed this when removing $ID ........ ................
-
- * Makefile, /, configure, configure.ac: Merged revisions 259760 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r259760 | qwell | 2010-04-28 14:19:54 -0500
- (Wed, 28 Apr 2010) | 14 lines Merged revisions 259748 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r259748 | qwell | 2010-04-28 14:17:38 -0500 (Wed, 28 Apr 2010) |
- 7 lines Remove usage of `id` since it isn't useful and was
- causing breakge. Solaris `id` doesn't support the -u argument.
- Instead of figuring out how to fix this to work on Solaris, I
- decided to check why it was necessary and where else it was used.
- It was only used in one place, and it hasn't been needed for a
- very long time (I question whether it was ever needed). ........
- ................
-
-2010-04-28 17:19 +0000 [r259681] Jeff Peeler <jpeeler@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 259672 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r259672 | jpeeler | 2010-04-28 12:18:43 -0500
- (Wed, 28 Apr 2010) | 11 lines Merged revisions 259664 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r259664 | jpeeler | 2010-04-28 12:13:29 -0500 (Wed, 28 Apr 2010)
- | 4 lines Do not play goodbye prompt after timeout of message
- review. ABE-2124 ........ ................
-
-2010-04-27 22:46 +0000 [r259616] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 259538 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r259538 | rmudgett | 2010-04-27 17:18:09 -0500
- (Tue, 27 Apr 2010) | 18 lines Merged revisions 259531 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r259531 | rmudgett | 2010-04-27 16:53:07 -0500 (Tue, 27 Apr 2010)
- | 11 lines DAHDI "WARNING" message is confusing and vague
- "WARNING[28406]: chan_dahdi.c:6873 ss_thread: CallerID feed
- failed: Success" Changed the warning to "Failed to decode
- CallerID on channel 'name'". The message before it is likely more
- specific about why the CallerID decode failed. SWP-501 AST-283
- ........ ................
-
-2010-04-27 21:50 +0000 [r259528] Leif Madsen <lmadsen@digium.com>
-
- * sounds/Makefile: Merged revisions 259527 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r259527 | lmadsen | 2010-04-27 16:49:36 -0500 (Tue, 27 Apr 2010)
- | 23 lines Merged revisions 259526 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r259526 | lmadsen | 2010-04-27 16:48:47 -0500 (Tue, 27 Apr 2010)
- | 15 lines Update sounds files. * Add additional sounds prompts
- for say_enumeration * Update the English conference sounds
- prompts so they are better quality and all sound more consistent
- * Clean up the core-sounds-XX.txt and extra-sounds-XX.txt files
- to include all present sound files Both core (en, fr, es) and
- extra (en, fr) sounds files have been updated. (closes issue
- #16200) Reported by: murf (closes issue #17137) Reported by:
- lmadsen ........ ................
-
-2010-04-27 21:25 +0000 [r259356-259486] Jason Parker <jparker@digium.com>
-
- * main/editline/configure.in, /, main/editline/configure,
- main/editline/Makefile.in: Merged revisions 259439 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r259439 | qwell | 2010-04-27 16:13:01 -0500 (Tue, 27 Apr 2010) |
- 5 lines Add gar to the check for AR for those silly OSes
- (Solaris) that don't have ar. autoconf2.13 couldn't handle
- AC_PROG_GREP, so I removed it. This is fine, since we don't need
- to use anything that the configure script doesn't. ........
-
- * /: Unblock revision 259439.
-
- * /, configure, configure.ac: Merged revisions 259353 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r259353 | qwell | 2010-04-27 14:31:55 -0500
- (Tue, 27 Apr 2010) | 12 lines Merged revisions 259352 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r259352 | qwell | 2010-04-27 14:29:26 -0500 (Tue, 27 Apr 2010) |
- 5 lines Support the silly OSes that don't have ar and strip.
- Since AC_PATH_TOOL is equiv to AC_CHECK_TOOL when path isn't
- specified, and AC_PATH_TOOLS doesn't exist, we'll just switch to
- AC_CHECK_TOOLS. ........ ................
-
-2010-04-27 19:03 +0000 [r259310] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged
- revisions 259307 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r259307 | rmudgett | 2010-04-27 13:29:33 -0500 (Tue, 27 Apr 2010)
- | 21 lines Merged revisions 259270 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r259270 | rmudgett | 2010-04-27 13:14:54 -0500 (Tue, 27 Apr 2010)
- | 14 lines hidecalleridname parameter in chan_dahdi.conf Issue
- #7321 implements a new chan_dahdi configuration option. However,
- a change mentioned in the issue was never implemented. This is
- the change that will allow the feature to work. I added a note to
- chan_dahdi.conf.sample about the feature. (closes issue #17143)
- Reported by: djensen99 Patches: diff.txt uploaded by djensen99
- (license NA) (One line change) Tested by: djensen99 ........
- ................
-
-2010-04-26 21:48 +0000 [r259103-259109] Mark Michelson <mmichelson@digium.com>
-
- * main/channel.c, /: Merged revisions 259105 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r259105 | mmichelson | 2010-04-26 16:45:13 -0500 (Mon, 26 Apr
- 2010) | 9 lines Merged revisions 259104 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r259104 | mmichelson | 2010-04-26 16:44:43 -0500 (Mon, 26 Apr
- 2010) | 3 lines Let compilation succeed warning-free when
- DONT_OPTIMIZE is turned off. ........ ................
-
- * main/channel.c, /: Merged revisions 259023 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r259023 | mmichelson | 2010-04-26 16:13:35 -0500 (Mon, 26 Apr
- 2010) | 19 lines Merged revisions 259018 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r259018 | mmichelson | 2010-04-26 16:03:08 -0500 (Mon, 26 Apr
- 2010) | 13 lines Prevent Newchannel manager events for dummy
- channels. No Newchannel manager event will be fired for channels
- that are allocated to not match a registered technology type.
- Thus bogus channels allocated solely for variable substitution or
- CDR operations do not result in a Newchannel event. (closes issue
- #16957) Reported by: atis Review:
- https://reviewboard.asterisk.org/r/601 ........ ................
-
-2010-04-26 16:00 +0000 [r258935] Leif Madsen <lmadsen@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 258934 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r258934 |
- lmadsen | 2010-04-26 10:59:34 -0500 (Mon, 26 Apr 2010) | 7 lines
- Small error in the T.140 RTP port verbose log. (closes issue
- #16988) Reported by: frawd Patches: chan_sip_sdp_verbose_fix.diff
- uploaded by frawd (license 610) Tested by: russell ........
-
-2010-04-25 18:14 +0000 [r258779] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_monitor.c, /: Merged revisions 258776 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r258776 | tilghman | 2010-04-25 13:12:14 -0500 (Sun, 25 Apr 2010)
- | 13 lines Merged revisions 258775 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r258775 | tilghman | 2010-04-25 13:09:05 -0500 (Sun, 25 Apr 2010)
- | 6 lines When StopMonitor is called, ensure that it will not be
- restarted by a channel event. (closes issue #16590) Reported by:
- kkm Patches: resmonitor-16590-trunk.239289.diff uploaded by kkm
- (license 888) ........ ................
-
-2010-04-22 22:15 +0000 [r258676] Matthew Nicholson <mnicholson@digium.com>
-
- * main/cdr.c, main/channel.c, /, main/features.c: Merged revisions
- 258671,258675 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r258671 | mnicholson | 2010-04-22 16:57:59 -0500 (Thu, 22 Apr
- 2010) | 32 lines Merged revisions 193391,258670 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r193391 | mnicholson | 2009-05-08 16:01:25 -0500 (Fri, 08 May
- 2009) | 8 lines Set the proper disposition on originated calls.
- (closes issue #14167) Reported by: jpt Patches:
- call-file-missing-cdr2.diff uploaded by mnicholson (license 96)
- Tested by: dlotina, rmartinez, mnicholson ........ r258670 |
- mnicholson | 2010-04-22 16:49:07 -0500 (Thu, 22 Apr 2010) | 11
- lines Fix broken CDR behavior. This change allows a CDR record
- previously marked with disposition ANSWERED to be set as BUSY or
- NO ANSWER. Additionally this change partially reverts r235635 and
- does not set the AST_CDR_FLAG_ORIGINATED flag on CDRs generated
- from ast_call(). To preserve proper CDR behavior, the
- AST_CDR_FLAG_DIALED flag is now cleared from all brige CDRs in
- ast_bridge_call(). (closes issue #16797) Reported by:
- VarnishedOtter Tested by: mnicholson ........ (closes issue
- #16222) Reported by: telles Tested by: mnicholson
- ................ r258675 | mnicholson | 2010-04-22 17:11:23 -0500
- (Thu, 22 Apr 2010) | 2 lines Fix previous commit.
- ................
-
-2010-04-22 21:58 +0000 [r258516-258672] Russell Bryant <russell@digium.com>
-
- * /, main/event.c: Merged revisions 258632 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk For 1.6.2, only
- merge the bug fixes, not the unit test. ........ r258632 |
- russell | 2010-04-22 16:06:53 -0500 (Thu, 22 Apr 2010) | 22 lines
- Add ast_event subscription unit test and fix some ast_event API
- bugs. This patch introduces another test in test_event.c that
- exercises most of the subscription related ast_event API calls. I
- made some minor additions to the existing event allocation test
- to increase API coverage by the test code. Finally, I made a list
- in a comment of API calls not yet touched by the test module as a
- to-do list for future test development. During the development of
- this test code, I discovered a number of bugs in the event API.
- 1) subscriptions to AST_EVENT_ALL were not handled appropriately
- in a couple of different places. The API allows a subscription to
- all event types, but with IE parameters, just as if it was a
- subscription to a specific event type. However, the parameters
- were being ignored. This affected ast_event_check_subscriber()
- and event distribution to subscribers. 2) Some of the logic in
- ast_event_check_subscriber() for checking subscriptions against
- query parameters was wrong. Review:
- https://reviewboard.asterisk.org/r/617/ ........
-
- * /, doc/tex/channelvariables.tex: Merged revisions 258515 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r258515 | russell | 2010-04-22 12:36:34 -0500 (Thu, 22
- Apr 2010) | 2 lines Add MEETMEBOOKID from r256019. ........
-
-2010-04-21 22:11 +0000 [r258436] Jeff Peeler <jpeeler@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 258433 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r258433 | jpeeler | 2010-04-21 16:56:09 -0500
- (Wed, 21 Apr 2010) | 15 lines Merged revisions 258432 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r258432 | jpeeler | 2010-04-21 16:45:36 -0500 (Wed, 21 Apr 2010)
- | 8 lines Fix looping forever when no input received in certain
- voicemail menu scenarios. Specifically, prompting for an
- extension (when leaving or forwarding a message) or when
- prompting for a digit (when saving a message or changing
- folders). ABE-2122 SWP-1268 ........ ................
-
-2010-04-21 19:44 +0000 [r258384-258386] Leif Madsen <lmadsen@digium.com>
-
- * doc/tex/asterisk.tex: Remove missed line in previous merge.
- (issue #17220)
-
- * configure: Forgot to merge the updated configure script. (issue
- #17220)
-
- * doc/tex/localchannel.tex, doc/tex/enum.tex, makeopts.in,
- doc/tex/asterisk.tex, Makefile, /, doc/tex/Makefile,
- configure.ac, doc/tex/phoneprov.tex, doc/tex, doc/tex/ael.tex,
- build_tools/prep_tarball: Merged revisions 258351 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r258351 | lmadsen | 2010-04-21 14:18:35 -0500 (Wed, 21 Apr 2010)
- | 20 lines Add ability to generate ASCII documentation from the
- TeX files. These changes add the ability to run 'make
- asterisk.txt' just like the existing 'make asterisk.pdf' commands
- to generate a text document from the TeX files we have in the
- doc/tex/ directory. I've also updated a few of the .tex files
- because they weren't properly escaping certain characters so they
- would show up as Unicode characters (like [U+021C]). Made changes
- to the configure scripts so it would detect the catdvi program
- which is required to convert the .dvi file generated by latex.
- I've also added a few lines to the build_tools/prep_tarball
- script so that the text documentation gets generated and added to
- future tarballs of Asterisk releases. (closes issue #17220)
- Reported by: lmadsen Patches: asterisk.txt.patch uploaded by
- lmadsen (license 10) asterisk.txt.patch-v4 uploaded by pabelanger
- (license 224) Tested by: lmadsen, pabelanger ........
-
-2010-04-21 18:19 +0000 [r258314] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 258305 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r258305 |
- dvossel | 2010-04-21 13:13:36 -0500 (Wed, 21 Apr 2010) | 12 lines
- fixes issue with double "sip:" in header field This is a clear
- mistake in logic. Future discussions about how to avoid having to
- handle uri's like this should take place in the future, but this
- fix needs to go in for now. (closes issue #15847) Reported by:
- ebroad Patches: doublesip.patch uploaded by ebroad (license 878)
- ........
-
-2010-04-20 19:03 +0000 [r258148-258150] Leif Madsen <lmadsen@digium.com>
-
- * /, configs/cli_aliases.conf.sample: Merged revisions 258149 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r258149 | lmadsen | 2010-04-20 14:02:49 -0500 (Tue, 20
- Apr 2010) | 1 line Add 'soft hangup' alias per Steve Johnson on
- asterisk-users. ........
-
- * configs/extensions.conf.sample, /: Merged revisions 258147 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r258147 | lmadsen | 2010-04-20 13:38:39 -0500 (Tue, 20
- Apr 2010) | 8 lines Add example dialplan for dialing ISN numbers
- (http://www.freenum.org). Minor tweaks and documentation added by
- me. (closes issue #17058) Reported by: pprindeville Patches:
- freenum.patch#5 uploaded by pprindeville (license 347) Tested by:
- lmadsen ........
-
-2010-04-20 18:04 +0000 [r258108] Jeff Peeler <jpeeler@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 258065 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r258065 | jpeeler | 2010-04-20 12:06:19 -0500
- (Tue, 20 Apr 2010) | 17 lines Merged revisions 258029 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r258029 | jpeeler | 2010-04-20 11:16:33 -0500 (Tue, 20 Apr 2010)
- | 11 lines Play correct prompt when voicemail store failure
- occurs after attempted forward. If a user's mailbox was full and
- a message was attempted to be forwarded to said box, warnings on
- the console would indicate failure. However, the played prompt
- was that of success (vm-msgsaved). Now storage failure is taken
- into account and the correct prompt (vm-mailboxfull) is played
- when appropriate. ABE-2123 SWP-1262 ........ ................
-
-2010-04-20 18:02 +0000 [r258107] Leif Madsen <lmadsen@digium.com>
-
- * contrib/scripts/sip-friends.sql, /: Merged revisions 258106 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r258106 | lmadsen | 2010-04-20 13:01:28 -0500 (Tue, 20
- Apr 2010) | 7 lines Add missing 'useragent' field to
- sip-friends.sql file. (closes issue #17171) Reported by: thehar
- Patches: sip-friends.patch uploaded by thehar (license 831)
- Tested by: pabelanger, thehar ........
-
-2010-04-19 21:58 +0000 [r257948-257950] Jason Parker <jparker@digium.com>
-
- * main/indications.c, /: Merged revisions 257949 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r257949 |
- qwell | 2010-04-19 16:57:56 -0500 (Mon, 19 Apr 2010) | 1 line
- Change log message to match severity. ........
-
- * main/indications.c, /: Merged revisions 257947 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r257947 |
- qwell | 2010-04-19 16:49:30 -0500 (Mon, 19 Apr 2010) | 6 lines
- Don't consider a missing indications.conf to be a critical error.
- There were many changes in revision 176627 which would avoid the
- error that a missing config would have caused. Other than this,
- there are no other config files (including asterisk.conf,
- surprisingly) that are required. ........
-
-2010-04-19 18:30 +0000 [r257850] Terry Wilson <twilson@digium.com>
-
- * /, main/features.c: Merged revisions 257810 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r257810 |
- twilson | 2010-04-19 12:57:41 -0500 (Mon, 19 Apr 2010) | 5 lines
- Fix incomplete CDR merge from r195881 Because res/res_features.c
- was removed and main/cdr.c added, these changes didn't make it to
- trunk and the 1.6.x branches ........
-
-2010-04-18 17:28 +0000 [r257771] Tilghman Lesher <tlesher@digium.com>
-
- * configs/cdr_odbc.conf.sample, /: Merged revisions 257768 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r257768 | tilghman | 2010-04-18 12:25:53 -0500 (Sun, 18
- Apr 2010) | 2 lines Removing unused configuration parameters
- ........
-
-2010-04-16 21:47 +0000 [r257740] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
-
- * apps/app_mixmonitor.c, /: Merged revisions 257713 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r257713 | dhubbard | 2010-04-16 16:22:30 -0500
- (Fri, 16 Apr 2010) | 28 lines Merged revisions 257686 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r257686 | dhubbard | 2010-04-16 16:15:43 -0500 (Fri, 16 Apr 2010)
- | 21 lines Make the mixmonitor thread process audio frames faster
- Mantis issue 17078 reports MixMonitor recordings have shorter
- durations than the call duration. This was because the mixmonitor
- thread was not processing frames from the audiohook fast enough.
- The mixmonitor thread would slowly fall behind the most recent
- audio frame and when the channel hangs up, the mixmonitor thread
- would exit without processing the same number of frames as the
- channel; leaving the mixmonitor recording shorter than actual
- call duration. This revision fixes this issue by moving the
- ast_audiohook_trigger_wait() and the subsequent audiohook.status
- check into the block where the ast_audiohook_read_frame()
- function returns NULL. (closes issue #17078) Reported by:
- geoff2010 Patches: dw-M17078.patch uploaded by dhubbard (license
- 733) Tested by: dhubbard, geoff2010 Review:
- https://reviewboard.asterisk.org/r/611/ ........ ................
-
-2010-04-15 21:34 +0000 [r257510-257597] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk/app.h, /, main/app.c: Merged revisions 257560
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r257560 | tilghman | 2010-04-15 16:26:19 -0500
- (Thu, 15 Apr 2010) | 13 lines Merged revisions 257544 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r257544 | tilghman | 2010-04-15 16:23:24 -0500 (Thu, 15 Apr 2010)
- | 6 lines Allow application options with arguments to contain
- parentheses, through a variety of escaping techniques. Fixes
- SWP-1194 (ABE-2143). Review:
- https://reviewboard.asterisk.org/r/604/ ........ ................
-
- * /, channels/chan_sip.c: Merged revisions 257493 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r257493 | tilghman | 2010-04-15 15:30:15 -0500 (Thu, 15 Apr 2010)
- | 20 lines Merged revisions 257467 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r257467 | tilghman | 2010-04-15 15:24:50 -0500 (Thu, 15 Apr 2010)
- | 13 lines Don't recreate peer, when responding to a repeated
- deregistration attempt. When a reply to a deregistration is lost
- in transmit, the client retries the deregistration. Previously,
- this would cause a realtime/autocreate peer to be loaded back
- into memory, after it had already been correctly purged. Instead,
- we just want to resend the reply without loading the peer.
- (closes issue #16908) Reported by: kkm Patches:
- 20100412__issue16908.diff.txt uploaded by tilghman (license 14)
- Tested by: kkm ........ ................
-
-2010-04-15 19:42 +0000 [r257344-257428] Leif Madsen <lmadsen@digium.com>
-
- * doc/backtrace.txt: Merged revisions 257427 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r257427 | lmadsen | 2010-04-15 14:41:05 -0500 (Thu, 15 Apr 2010)
- | 21 lines Merged revisions 257426 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r257426 | lmadsen | 2010-04-15 14:40:33 -0500 (Thu, 15 Apr 2010)
- | 13 lines Update backtrace.txt documentation. Update the
- backtrace.txt documentation so it conforms to the same layout as
- other documents we've been working on recently. Additionally, add
- a bunch of new information about gathering backtraces for crashes
- and deadlocks, along with ways of verifying your file before
- uploading it. Create a couple of one line commands for people to
- generate the files we need. (closes issue #17190) Reported by:
- lmadsen Patches: backtrace.txt.patch-2 uploaded by lmadsen
- (license 10) Tested by: lmadsen, pabelanger ........
- ................
-
- * doc/backtrace.txt: Merged revisions 257343 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r257343 | lmadsen | 2010-04-15 08:44:38 -0500 (Thu, 15 Apr 2010)
- | 9 lines Merged revisions 257342 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r257342 | lmadsen | 2010-04-15 08:41:45 -0500 (Thu, 15 Apr 2010)
- | 1 line Update address of the bug tracker. ........
- ................
-
-2010-04-14 23:00 +0000 [r257265] Tilghman Lesher <tlesher@digium.com>
-
- * configs/features.conf.sample, /, main/features.c: Merged
- revisions 257262 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r257262 |
- tilghman | 2010-04-14 17:57:35 -0500 (Wed, 14 Apr 2010) | 15
- lines Yet another issue where the conversion of the application
- delimiter to comma caused an issue. Application arguments within
- the feature map could possibly contain a comma, which conflicts
- with the syntax of the features.conf configuration file. This
- patch allows the argument to be wrapped in parentheses or quoted,
- to allow the application arguments to be interpreted as a single
- configuration parameter. (closes issue #16646) Reported by:
- pinga-fogo Patches: 20100414__issue16646.diff.txt uploaded by
- tilghman (license 14) Tested by: tilghman Review:
- https://reviewboard.asterisk.org/r/547/ ........
-
-2010-04-13 19:20 +0000 [r257210] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 257191 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r257191 |
- tilghman | 2010-04-13 14:17:48 -0500 (Tue, 13 Apr 2010) | 10
- lines Also unref the pvt when we delete the provisional keepalive
- job. (closes issue #16774) Reported by: kowalma Patches:
- 20100315__issue16774.diff.txt uploaded by tilghman (license 14)
- Tested by: falves11, jamicque Review:
- https://reviewboard.asterisk.org/r/591/ ........
-
-2010-04-13 18:43 +0000 [r257184] Matthew Nicholson <mnicholson@digium.com>
-
- * main/manager.c, /, configs/manager.conf.sample: Merged revisions
- 257146 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r257146 | mnicholson | 2010-04-13 13:10:30 -0500 (Tue, 13 Apr
- 2010) | 16 lines Merged revisions 257070 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r257070 | mnicholson | 2010-04-13 11:46:30 -0500 (Tue, 13 Apr
- 2010) | 9 lines Add an option to restore past broken behavor of
- the Events manager action Before r238915, certain values for the
- EventMask parameter of the Events action would result in no
- response being returned. This patch adds an option to restore
- that broken behavior. Also while fixing this bug I discovered
- that passing an empty EventMasks parameter would also result in
- no response being returned, this has been fixed as well while
- being preserved when the broken behavior is requested. (closes
- issue #17023) Reported by: nblasgen Review:
- https://reviewboard.asterisk.org/r/602/ ........ ................
-
-2010-04-13 16:38 +0000 [r257068] Tilghman Lesher <tlesher@digium.com>
-
- * cdr/cdr_sqlite3_custom.c, /: Merged revisions 257065 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r257065 | tilghman | 2010-04-13 11:33:21 -0500 (Tue, 13 Apr 2010)
- | 8 lines Ensure that we can have commas within cdr values.
- (closes issue #17001) Reported by: snuffy Patches:
- 20100412__issue17001.diff.txt uploaded by tilghman (license 14)
- Tested by: snuffy ........
-
-2010-04-12 17:30 +0000 [r256822-256902] Leif Madsen <lmadsen@digium.com>
-
- * doc/HOWTO_collect_debug_information.txt (added): Merged revisions
- 256901 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r256901 | lmadsen | 2010-04-12 12:29:53 -0500 (Mon, 12 Apr 2010)
- | 23 lines Merged revisions 256900 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r256900 | lmadsen | 2010-04-12 12:29:26 -0500 (Mon, 12 Apr 2010)
- | 15 lines Add How-To document on collecting debugging info for
- issues.asterisk.org Paul Belanger has been helping a lot with bug
- tracking recently and created this document that we can now point
- to when additional debugging information is required. This
- document will help those filing issues to know how to get the
- information required when filing their issues. This will make
- things easier on the developers. Initial text and changes by
- pabelanger. Tweaks and editing by myself. (closes issue #17159)
- Reported by: pabelanger Patches:
- HOWTO_collect_debug_information.txt.patch uploaded by lmadsen
- (license 10) Tested by: tzafrir, pabelanger, lmadsen ........
- ................
-
- * apps/app_voicemail.c, /: Merged revisions 256860 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r256860 | lmadsen | 2010-04-12 11:16:43 -0500 (Mon, 12 Apr 2010)
- | 3 lines Remove silly debug message that is not useful. (issue
- #17159) ........
-
- * /, main/logger.c: Merged revisions 256821 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r256821 |
- lmadsen | 2010-04-12 09:39:37 -0500 (Mon, 12 Apr 2010) | 8 lines
- CLI command logger set level auto complete. A simple patch to
- enable auto tab complete. (closes issue #17152) Reported by:
- pabelanger Patches: 0017152.patch uploaded by pabelanger (license
- 224) ........
-
-2010-04-08 22:03 +0000 [r256483] Tilghman Lesher <tlesher@digium.com>
-
- * main/app.c: Backport /proc/%d/fd method of closing file
- descriptors to 1.6.2.
-
-2010-04-06 19:40 +0000 [r256373] Tilghman Lesher <tlesher@digium.com>
-
- * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
- include/asterisk/lock.h: Merged revisions 256370 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r256370 | tilghman | 2010-04-06 14:28:42 -0500 (Tue, 06 Apr 2010)
- | 2 lines Mac OS X does not support comparing a mutex to its
- initializer. Create a test for this. ........
-
-2010-04-06 18:53 +0000 [r256268-256368] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_dahdi.c: CallerID channel DAHDI port FXS are empty
- after the first call. The bug is exposed if MFC/R2 support is
- built into asterisk (i.e., openr2.h is present in the include
- path). Code that unconditionally clears the CallerID name and
- number is included. Also fixed a malformed if test in mkintf()
- added by issue 15883. Converted the if statement to a switch
- statement for clarity. Regression of the issue 15883 fix. (closes
- issue #16968) Reported by: grecco Patches: issue16968.patch
- uploaded by rmudgett (license 664) (closes issue #16747) Reported
- by: viniciusfontes
-
- * channels/chan_dahdi.c, /: Merged revisions 256265 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r256265 | rmudgett | 2010-04-05 19:39:44 -0500
- (Mon, 05 Apr 2010) | 12 lines Merged revisions 256225 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r256225 | rmudgett | 2010-04-05 19:10:16 -0500 (Mon, 05 Apr 2010)
- | 5 lines DAHDI/PRI call to pri_channel_bridge() not protected by
- PRI lock. SWP-1231 ABE-2163 ........ ................
-
-2010-05-03 Leif Madsen <lmadsen@digium.com>
-
- * Asterisk 1.6.2.7 Released
-
-2010-04-29 Leif Madsen <lmadsen@digium.com>
-
- * Asterisk 1.6.2.7-rc3 Released
-
-2010-04-29 10:31 +0000 [r260053] David Vossel <dvossel@digium.com>
-
- * include/asterisk/audiohook.h, main/audiohook.c: Fixes crash in
- audiohook_write_list. (closes issue 0017052) Reported by: dvossel
- Tested by: dvossel. (closes issue 0016196) Reported by: atis.
- Review: https://reviewboard.asterisk.org/r/623/
-
-2010-04-28 10:31 +0000 [r259899] David Vossel <dvossel@digium.com>
-
- * channels/chan_local.c, main/channel.c: Resolves deadlocks in
- chan_local. (closes issue 0017185) Reported by: schmoozecom
- Patches: issue_17185_v1.diff uploaded by dvossel (license 671)
- issue_17185_v2.diff uploaded by dvossel (license 671) Tested
- by: schmoozecom, GameGamer43
- Review: https://reviewboard.asterisk.org/r/631/
-
-2010-04-13 Leif Madsen <lmadsen@digium.com>
-
- * Asterisk 1.6.2.7-rc2 Released
-
-2010-04-13 [r257210] Tilghman Lesher <tlesher@digium.com>
-
- Also unref the pvt when we delete the provisional keepalive job.
-
- (closes issue #16774)
- Reported by: kowalma
- Patches:
- 20100315__issue16774.diff.txt uploaded by tilghman (license 14)
- Tested by: falves11, jamicque
-
- Review: https://reviewboard.asterisk.org/r/591/
-
-2010-04-05 Leif Madsen <lmadsen@digium.com>
-
- * Asterisk 1.6.2.7-rc1 Released
-
-2010-04-05 15:15 +0000 [r256162] Leif Madsen <lmadsen@digium.com>
-
- * doc/tex/localchannel.tex, /: Merged revisions 256161 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r256161 | lmadsen | 2010-04-05 10:14:53 -0500 (Mon, 05 Apr 2010)
- | 1 line Fix for localchannel.tex to allow PDFs to be generated
- again. ........
-
-2010-04-02 23:56 +0000 [r256013-256020] Russell Bryant <russell@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 256019 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r256019 |
- russell | 2010-04-02 18:55:57 -0500 (Fri, 02 Apr 2010) | 10 lines
- Export MEETMEBOOKID and fix pin-less conferences with realtime
- conferences (closes issue #16866) Reported by: DEA Patches:
- rt-meetme-options.txt uploaded by DEA (license 3) Tested by: DEA
- Review: https://reviewboard.asterisk.org/r/582/ ........
-
- * 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 [r255955] 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 [r255676-255816] 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 17:49 +0000 [r255505] Leif Madsen <lmadsen@digium.com>
-
- * configs/sip.conf.sample, apps/app_dial.c: 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 20:58 +0000 [r255326-255413] 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 19:28 +0000 [r255023-255067] Leif Madsen <lmadsen@digium.com>
-
- * configs/sip.conf.sample, /: Merged revisions 255066 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r255066 | lmadsen | 2010-03-26 14:27:56 -0500 (Fri, 26 Mar 2010)
- | 6 lines Replace some documentation from 1.6.x back into trunk.
- This documentation associated wth tlsbindaddr is still useful so
- lets synchronize it between trunk and 1.6.x branches. (issue
- #17054) ........
-
- * 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 20:43 +0000 [r254770-254805] 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. ........
- ................
-
- * 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 20:09 +0000 [r254721] 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 17:47 +0000 [r254556] 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 ........ ................
-
-2010-03-25 17:21 +0000 [r254548] Sean Bright <sean@malleable.com>
-
- * channels/chan_sip.c: Initialize stream to avoid a compilation
- error.
-
-2010-03-25 17:12 +0000 [r254542] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_sip.c: Fix potential crashes from trying to
- reference nonexistent RTP streams.
-
-2010-03-25 16:26 +0000 [r254499] 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 16:22 +0000 [r254482] 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 15:21 +0000 [r254447] 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 17:19 +0000 [r254288] 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)
- ........ ................
-
-2010-03-23 22:05 +0000 [r254131] 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 ........
-
-2010-03-23 21:20 +0000 [r254068] Jeff Peeler <jpeeler@digium.com>
-
- * 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/ ........
-
-2010-03-22 19:55 +0000 [r253801] 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) ........
- ................
-
-2010-03-22 19:06 +0000 [r253714-253760] Tilghman Lesher <tlesher@digium.com>
-
- * /, contrib/scripts/dbsep.cgi: Merged revisions 253758 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r253758 | tilghman | 2010-03-22 14:05:27 -0500 (Mon, 22
- Mar 2010) | 2 lines Update query should be an UPDATE, not a
- SELECT. ........
-
- * /, contrib/scripts/dbsep.cgi: Merged revisions 253755 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r253755 | tilghman | 2010-03-22 13:58:48 -0500 (Mon, 22
- Mar 2010) | 4 lines Return the list for later manipulation. This
- fixes an issue with the update procedure. Debugging with
- mmichelson. ........
-
- * configs/dbsep.conf.sample, /, contrib/scripts/dbsep.cgi: Merged
- revisions 253712 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r253712 |
- tilghman | 2010-03-22 11:59:35 -0500 (Mon, 22 Mar 2010) | 2 lines
- Accomodate equal signs in DSNs and add documentation, based upon
- mmichelson's feedback. ........
-
-2010-03-20 17:33 +0000 [r253595-253620] Russell Bryant <russell@digium.com>
-
- * cdr/cdr_pgsql.c, main/stdtime/localtime.c, main/tcptls.c, /,
- main/features.c: Merged revisions 253540 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r253540 |
- russell | 2010-03-20 07:03:07 -0500 (Sat, 20 Mar 2010) | 2 lines
- Resolve more compiler warnings on FreeBSD. ........
-
- * apps/app_followme.c, apps/app_dial.c, /: Merged revisions 253538
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r253538 | russell | 2010-03-20 06:43:08 -0500 (Sat, 20
- Mar 2010) | 2 lines Resolve compiler warnings on FreeBSD.
- ........
-
- * /, pbx/pbx_dundi.c: Merged revisions 253537 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r253537 |
- russell | 2010-03-20 06:39:39 -0500 (Sat, 20 Mar 2010) | 2 lines
- Resolve a compiler warning on FreeBSD. ........
-
- * channels/chan_dahdi.c, /: Merged revisions 253536 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. ........
-
-2010-03-19 08:05 +0000 [r253492] Alec L Davis <sivad.a@paradise.net.nz>
-
- * main/astobj2.c, /: Merged revisions 253490 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r253490 |
- alecdavis | 2010-03-19 20:37:00 +1300 (Fri, 19 Mar 2010) | 19
- lines prevent segfault if bad magic number is encountered.
- internal_ao2_ref uses INTERNAL_OBJ which mzy report 'bad magic
- number', but internal_ao2_ref continues on, causing segfault.
- Although AO2_MAGIC number is checked by INTERNAL_OBJ before
- internal_ao2_ref is called, A02_MAGIC is being destroyed (or a
- wrong pointer) by the time internal_ao2_ref uses INTERNAL_OBJ.
- internal_ao2_ref now returns -1 if INTERNAL_OBJ encouters a bad
- magic number. (issue #17037) Reported by: alecdavis Patches:
- bug17037.diff.txt uploaded by alecdavis (license 585) Tested by:
- alecdavis ........
-
-2010-03-18 17:54 +0000 [r253257-253346] Leif Madsen <lmadsen@digium.com>
-
- * /, apps/app_userevent.c: Merged revisions 253345 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r253345 | lmadsen | 2010-03-18 12:52:35 -0500 (Thu, 18 Mar 2010)
- | 7 lines Change usage of pipe to comma in UserEvent docs. Change
- the example usage of pipe as a separator to comma in the
- UserEvent documentation. (closes issue #16961) Reported by:
- jlpedrosa ........
-
- * 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 00:41 +0000 [r253029-253033] Leif Madsen <lmadsen@digium.com>
-
- * main/xmldoc.c, /: Merged revisions 253032 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r253032 |
- lmadsen | 2010-03-16 19:40:51 -0500 (Tue, 16 Mar 2010) | 1 line
- Fix a typo. ........
-
- * 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 [r252978] 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 19:38 +0000 [r252850] Sean Bright <sean@malleable.com>
-
- * res/res_clialiases.c, /: Merged revisions 252848 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r252848 | seanbright | 2010-03-16 15:36:24 -0400 (Tue, 16 Mar
- 2010) | 10 lines Include an extra newline after "Aliased CLI
- command" to get back the prompt. The other issue mentioned in
- this bug will be more difficult to resolve since we have no idea
- (right now) of knowing if the command that is aliased has been
- installed yet. (issue #16978) Reported by: jw-asterisk Tested by:
- seanbright ........
-
-2010-03-16 19:02 +0000 [r252770] 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 [r252763] 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 [r252626] 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 21:54 +0000 [r252622] 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 20:53 +0000 [r252535] 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 05:04 +0000 [r252365-252444] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 252442 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r252442 |
- tilghman | 2010-03-14 23:25:35 -0500 (Sun, 14 Mar 2010) | 7 lines
- THIS IS NOT PYTHON. Indentation doesn't matter, only braces do.
- (closes issue #17025) Reported by: smurfix Patches: sip.patch
- uploaded by smurfix (license 547) ........
-
- * 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 17:48 +0000 [r252317] 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 00:32 +0000 [r252137-252178] Terry Wilson <twilson@digium.com>
-
- * main/rtp.c: Remove unusued field
-
- * configs/sip.conf.sample, include/asterisk/frame.h, 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: 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 22:05 +0000 [r252090] Moises Silva <moises.silva@gmail.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 252088 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r252088 | moy | 2010-03-12 16:57:40 -0500 (Fri, 12 Mar 2010) | 1
- line add missing mfcr2_skip_category setting ........
-
-2010-03-12 19:50 +0000 [r251994] 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 19:49 +0000 [r251991] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 251946 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r251946 | rmudgett | 2010-03-12 13:05:40 -0600 (Fri, 12 Mar 2010)
- | 1 line Doxegen this chan_dahdi lock. ........
-
-2010-03-11 21:08 +0000 [r251879-251887] 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 1.6.2.6 released
-
-2010-03-05 Leif Madsen <lmadsen@digium.com>
-
- * Asterisk 1.6.2.6-rc2 released
-
-2010-03-05 Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_voicemail.c: Merged revisions 250913 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r250913 | tilghman
- | 2010-03-04 22:37:36 -0600 (Thu, 04 Mar 2010) | 7 lines Missing quote in
- ODBC query. (closes issue #16953) Reported by: elguero Patches:
- app_voicemail-odbc-syntax-fix.diff uploaded by elguero (license 37)
- ........
-
-2010-03-04 Leif Madsen <lmadsen@digium.com>
-
- * Asterisk 1.6.2.6-rc1 released
-
-2010-03-03 21:24 +0000 [r250610] 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 19:13 +0000 [r250484] 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 18:05 +0000 [r250253-250396] 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 21:10 +0000 [r249953-250052] 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
- ........
-
- * apps/app_directory.c, /: Merged revisions 249950 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r249950 | lmadsen | 2010-03-02 14:49:48 -0500 (Tue, 02 Mar 2010)
- | 4 lines Fix literal values wrapped in documentation. (closes
- issue #16145) Reported by: tilghman ........
-
-2010-03-02 19:50 +0000 [r249952] Alec L Davis <sivad.a@paradise.net.nz>
-
- * UPGRADE-1.6.txt, main/editline/makelist.in, apps/app_echo.c,
- UPGRADE.txt: revert ability to exit echo app caused a regression,
- as only supported VOICE, not VIDEO etc. (issue #16880)
-
-2010-03-02 19:26 +0000 [r249916-249933] Leif Madsen <lmadsen@digium.com>
-
- * /, main/features.c: Merged revisions 249925 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r249925 |
- lmadsen | 2010-03-02 14:24:43 -0500 (Tue, 02 Mar 2010) | 6 lines
- Add missing description of the PARKINGLOT variable in XML
- documentation. (closes issue #16743) Reported by: snuffy Patches:
- parkingdoc.diff uploaded by snuffy (license 35) ........
-
- * /, pbx/pbx_dundi.c: Merged revisions 249912 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r249912 |
- lmadsen | 2010-03-02 14:21:19 -0500 (Tue, 02 Mar 2010) | 6 lines
- Convert some DUNDI functions to XML documentation. (closes issue
- #16798) Reported by: snuffy Patches: xml_dundi.diff uploaded by
- snuffy (license 35) ........
-
-2010-03-02 19:12 +0000 [r249895] David Vossel <dvossel@digium.com>
-
- * channels/chan_console.c, channels/chan_gtalk.c,
- 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: 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 19:09 +0000 [r249894] Leif Madsen <lmadsen@digium.com>
-
- * /, apps/app_confbridge.c: Merged revisions 249892 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r249892 | lmadsen | 2010-03-02 14:02:56 -0500 (Tue, 02 Mar 2010)
- | 1 line Fix several XML documentation validate errors. ........
-
-2010-03-02 09:05 +0000 [r249844] 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 (issue #16880) Reported by: alecdavis
- Patches: echo_exit_1-6-1.diff.txt uploaded by alecdavis (license
- 585) Tested by: alecdavis
-
-2010-03-01 19:40 +0000 [r249675] 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 18:47 +0000 [r249625] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 249623 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r249623 | tilghman | 2010-03-01 12:36:06 -0600 (Mon, 01 Mar 2010)
- | 2 lines Constify a bit of app_voicemail, to make ODBC and IMAP
- compile once again. ........
-
-2010-03-01 17:25 +0000 [r249580] 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 ........ ................
-
-2010-02-28 20:51 +0000 [r249407-249493] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 249491 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r249491 | tilghman | 2010-02-28 14:50:01 -0600 (Sun, 28 Feb 2010)
- | 5 lines Fix unit test that Alec Davis broke. (closes issue
- #16927) Reported by: alecdavis ........
-
- * apps/app_voicemail.c, include/asterisk/app.h, /: Merged revisions
- 249405 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r249405 |
- tilghman | 2010-02-28 01:10:22 -0600 (Sun, 28 Feb 2010) | 2 lines
- Properly document voicemail API documents. Also fix a crash
- reported via the -dev list. ........
-
-2010-02-27 23:04 +0000 [r249321] 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
-
-2010-02-27 14:10 +0000 [r249238] 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 18:49 +0000 [r249190] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 249187 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r249187 | tilghman | 2010-02-26 12:41:57 -0600 (Fri, 26 Feb 2010)
- | 18 lines Cleanups to fix bugs in the VM count API functions. -
- Urgent voicemails were not attached, because the attachment code
- looked in the wrong folder. - Urgent voicemails were sometimes
- counted twice when displaying the count of new messages. -
- Backends were inconsistent as to which voicemails each API
- counted. (closes issue #15654) Reported by: tomo1657 Patches:
- 20100225__issue15654.diff.txt uploaded by tilghman (license 14)
- Tested by: tilghman (closes issue #16448) Reported by: hevad
- Review: https://reviewboard.asterisk.org/r/525/ ........
-
-2010-02-26 17:06 +0000 [r249104] 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 23:12 +0000 [r248955] 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 [r248949] 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 21:25 +0000 [r248864] 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 18:52 +0000 [r248797] 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 21:29 +0000 [r248643] 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 16:37 +0000 [r248398] 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 19:07 +0000 [r248011] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_console.c, main/loader.c, /: Merged revisions
- 228798 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ 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 ........
-
-2010-02-19 19:00 +0000 [r248005] Moises Silva <moises.silva@gmail.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 248003 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r248003 | moy | 2010-02-19 13:38:34 -0500 (Fri, 19 Feb 2010) | 1
- line mfcr2 issue 0016844 - Fix portability bit fields and make
- mfcr2_immediate_accept work again, reported and patched by
- korihor ........
-
-2010-02-19 18:45 +0000 [r248004] 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
- ................ ................
-
-2010-02-19 17:41 +0000 [r247916] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 247915 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r247915 |
- dvossel | 2010-02-19 11:40:26 -0600 (Fri, 19 Feb 2010) | 7 lines
- handle_request_invite revise comment, fix coding guideline issues
- I'm working with this code right now trying to analyze a
- deadlock. This change is just to clean up a few things before I
- make a more complex patch. ........
-
-2010-02-18 23:15 +0000 [r247792-247845] 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)
- ........
-
- * /, 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 21:25 +0000 [r247737-247776] David Vossel <dvossel@digium.com>
-
- * /, bridges/bridge_softmix.c: Merged revisions 247770 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r247770 | dvossel | 2010-02-18 15:23:48 -0600 (Thu, 18 Feb 2010)
- | 9 lines fixes confbridge crash when no timing module is loaded.
- (closes issue #16471) Reported by: kjotte Patches: M16471.diff
- uploaded by junky (license 177) Tested by: kjotte, junky ........
-
- * apps/app_queue.c, /: Merged revisions 247736 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r247736 |
- dvossel | 2010-02-18 14:58:41 -0600 (Thu, 18 Feb 2010) | 7 lines
- fixes Queue with C option crash (closes issue #16475) Reported
- by: okrief Patches: queue_crash.diff uploaded by dvossel (license
- 671) ........
-
-2010-02-18 19:41 +0000 [r247653] 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 16:58 +0000 [r247506-247512] 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 04:21 +0000 [r247426] Russell Bryant <russell@digium.com>
-
- * sounds/Makefile, 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 21:32 +0000 [r246989-247337] Mark Michelson <mmichelson@digium.com>
-
- * include/asterisk/strings.h, main/strings.c, /: 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.
- ........ ................
-
- * main/strings.c, /: Merged revisions 247076 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r247076 |
- mmichelson | 2010-02-16 17:44:33 -0600 (Tue, 16 Feb 2010) | 12
- lines Add va_end calls to __ast_str_helper. According to the man
- page for stdarg(3), "Each invocation of va_copy() must be matched
- by a corresponding invocation of va_end() in the same function."
- There were several cases in __ast_str_helper where va_copy was
- not matched with a corresponding call to va_end. ........
-
- * 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 21:03 +0000 [r246900-246982] 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: fixes merge error with Monitor calculation fix
-
- * 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 23:45 +0000 [r246713] 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 23:33 +0000 [r246547] 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 19:08 +0000 [r246464] Jason Parker <jparker@digium.com>
-
- * main/channel.c: Fix some silly formatting that made my head hurt.
-
-2010-02-10 21:28 +0000 [r246199-246207] 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. ........
-
- * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
- res/res_agi.c: Merged revisions 246030 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r246030 |
- tilghman | 2010-02-10 10:01:28 -0600 (Wed, 10 Feb 2010) | 12
- lines Solaris doesn't like outputting a NULL to a %s in format
- strings. Detect all platforms that don't like that, either, and
- ensure that when documentation is missing, we pass a non-NULL
- pointer when outputting the corresponding documentation. (closes
- issue #16689) Reported by: bklang Patches:
- 20100209__issue16689__with_tests.diff.txt uploaded by tilghman
- (license 14) Review: https://reviewboard.asterisk.org/r/497/
- ........
-
-2010-02-10 17:51 +0000 [r246117] 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 16:58 +0000 [r246073] 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 [r245948-246025] 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 23:11 +0000 [r245794] 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 [r245732] 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 17:43 +0000 [r245728] Matthew Nicholson <mnicholson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 245727 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r245727 |
- mnicholson | 2010-02-09 11:40:04 -0600 (Tue, 09 Feb 2010) | 2
- lines This commit removes an extra newline in T.38 generated SDP
- packets. This bug was caused by the fix introduced in r243860.
- (closes issue #16766) Reported by: raivisr Patches:
- t38-sdp-newline-fix1.diff uploaded by mnicholson (license 96)
- Tested by: raivisr ........
-
-2010-02-09 16:26 +0000 [r245683] 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 23:47 +0000 [r245626] Russell Bryant <russell@digium.com>
-
- * /, main/event.c: Merged revisions 245624 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r245624 |
- russell | 2010-02-08 17:43:00 -0600 (Mon, 08 Feb 2010) | 5 lines
- Fix return value of get_ie_str() and get_ie_str_hash() for
- non-existent IE. I found this bug while developing a unit test
- for event allocation. Testing is awesome. ........
-
-2010-02-08 22:46 +0000 [r245581] Tilghman Lesher <tlesher@digium.com>
-
- * channels/Makefile, /, main/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 20:43 +0000 [r245500] 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 19:27 +0000 [r245097] 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 [r244930] 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 19:28 +0000 [r244555] Mark Michelson <mmichelson@digium.com>
-
- * main/sched.c, /: Merged revisions 244547 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r244547 |
- mmichelson | 2010-02-03 13:26:53 -0600 (Wed, 03 Feb 2010) | 3
- lines Initialize counters in ast_sched_report so that resulting
- data is not bogus. ........
-
-2010-02-03 18:47 +0000 [r244508] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_dahdi.c, /, main/ast_expr2f.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 22:29 +0000 [r244445] 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 20:35 +0000 [r244395] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_dial.c, /: Merged revisions 244393 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r244393 |
- tilghman | 2010-02-02 14:32:29 -0600 (Tue, 02 Feb 2010) | 18
- lines Properly respect GOSUB_RESULT as to what to do with the
- master channel. Previously, we would parse GOSUB_RESULT, but not
- actually do anything with it. (closes issue #16686) Reported by:
- bklang Patches: app_dial-respect-gosub_result.patch uploaded by
- bklang (license 919) (with modifications) ........
-
-2010-02-02 Leif Madsen <lmadsen@digium.com>
-
- * Release Asterisk 1.6.2.2
-
- * 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 1.6.2.1
-
-2010-01-08 Leif Madsen <lmadsen@digium.com>
-
- * Release Asterisk 1.6.2.1-rc1
-
-2010-01-07 21:17 +0000 [r238499] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_console.c, channels/chan_oss.c, main/poll.c,
- channels/chan_usbradio.c, include/asterisk/utils.h, /,
- channels/chan_sip.c, channels/chan_alsa.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 20:17 +0000 [r238362-238416] 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 ........
-
-2010-01-07 09:49 +0000 [r238349] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
-
- * configs/sip.conf.sample, /: Merged revisions 238313 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r238313 | tzafrir | 2010-01-07 11:14:57 +0200 (ה', 07 ינו 2010) |
- 2 lines Document the usefulness of explicit udp:// in the
- register string ........
-
-2010-01-06 21:48 +0000 [r238234] 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 20:40 +0000 [r238137-238185] 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 15:22 +0000 [r238013] 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 06:54 +0000 [r237969] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 237968 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r237968 |
- tilghman | 2010-01-06 00:53:23 -0600 (Wed, 06 Jan 2010) | 2 lines
- Whoa, duplicate setting (dead code). ........
-
-2010-01-05 23:10 +0000 [r237924] Kinsey Moore <kmoore@digium.com>
-
- * apps/app_test.c: Add a wait to ensure TestServer thinks it has
- finished sending the final digit. This was previously committed
- to 1.4, 1.6.0, 1.6.1, and trunk just after 1.6.2 was created (and
- missed). 1.6.2 also needs this patch to resolve the bug. (closes
- issue #16550) Reported by: opticron Patches: apptest.diff
- uploaded by opticron (license 267)
-
-2010-01-05 23:09 +0000 [r237840-237921] 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 17:25 +0000 [r237743] 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)
- ........ ................
-
-2010-01-05 16:09 +0000 [r237657] Michiel van Baak <michiel@vanbaak.info>
-
- * apps/app_mixmonitor.c, /: Merged revisions 237656 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r237656 | mvanbaak | 2010-01-05 17:08:12 +0100 (Tue, 05 Jan 2010)
- | 6 lines Make CLI command 'mixmonitor start|stop <channel> work
- again. (closes issue #16534) Reported by: jlaguilar Fix as
- suggested by jlaguilar in the bugreport ........
-
-2010-01-04 21:52 +0000 [r237409-237577] 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 16:50 +0000 [r237328] David Vossel <dvossel@digium.com>
-
- * apps/app_queue.c, /: Merged revisions 237327 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r237327 |
- dvossel | 2010-01-04 10:39:11 -0600 (Mon, 04 Jan 2010) | 10 lines
- app_queue segfaults if realtime field uniqueid is NULL (closes
- issue #16385) Reported by: haakon Patches: app_queue.c.patch
- uploaded by haakon (license 880) app_queue.c.patch_v2 uploaded by
- dvossel (license 671) Tested by: haakon ........
-
-2010-01-04 16:27 +0000 [r237326] 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 [r237322] 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 10:03 +0000 [r237139] 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 [r236985] 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 21:13 +0000 [r236899-236905] 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. ........
-
- * main/cli.c, /: Merged revisions 236893 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r236893 |
- jpeeler | 2009-12-30 14:34:41 -0600 (Wed, 30 Dec 2009) | 11 lines
- Fix compiling with LOW_MEMORY. Modified handle_verbose to be
- LOW_MEMORY aware. (closes issue #16381) Reported by:
- michael_iedema Patches: ast_complete_source_filename.patch
- uploaded by michael iedema (license 942) modified by me ........
-
-2009-12-30 17:56 +0000 [r236804-236850] 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 22:13 +0000 [r236716] 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 17:40 +0000 [r236670] 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 [r236513-236635] 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 [r236437] 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 15:32 +0000 [r236361] 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 18:27 +0000 [r236189-236303] 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 17:04 +0000 [r236064] 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 19:58 +0000 [r235944] 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 [r235826] 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 ........
- ................
-
-2009-12-20 08:58 +0000 [r235775] Alec L Davis <sivad.a@paradise.net.nz>
-
- * main/dsp.c: restarts busydetector (if enabled) when DTMF is
- received after call is bridged. (closes issue #16389) Reported
- by: alecdavis Tested by: alecdavis Patch
- dtmf_busydetector.diff2.txt uploaded by alecdavis (license 585)
-
-2009-12-18 23:04 +0000 [r235665] 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 [r235576-235659] 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, 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 23:22 +0000 [r235522] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 235521 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r235521 |
- file | 2009-12-17 19:21:07 -0400 (Thu, 17 Dec 2009) | 3 lines
- Remove some old code for going to the 'fax' extension when a T.38
- switchover occurs. This would have already happened when we
- detected the CNG tone so this was basically a noop. ........
-
-2009-12-17 Leif Madsen <lmadsen@digium.com>
-
- * Release Asterisk 1.6.2.0
-
-2009-12-09 Leif Madsen <lmadsen@digium.com>
-
- * Release Asterisk 1.6.2.0-rc8
-
-2009-12-08 18:33 +0000 [r233731] 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/ ........
-
-2009-12-08 18:04 +0000 [r233694] Russell Bryant <russell@digium.com>
-
- * formats/format_sln16.c, formats/format_wav_gsm.c,
- formats/format_siren7.c, formats/format_ilbc.c,
- formats/format_vox.c, formats/format_pcm.c,
- formats/format_h263.c, formats/format_g723.c,
- formats/format_h264.c, formats/format_siren14.c,
- formats/format_jpeg.c, formats/format_g726.c,
- formats/format_gsm.c, formats/format_g729.c, /,
- formats/format_sln.c, formats/format_wav.c,
- formats/format_ogg_vorbis.c: Merged revisions 233692 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r233692 | russell | 2009-12-08 12:00:16 -0600 (Tue, 08 Dec 2009)
- | 16 lines 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 fix applies to trunk, 1.6.1, and 1.6.2.
- The fix for 1.4 and 1.6.0 will require a different approach since
- the module load priority functionality is not present in the
- module API. (issue #16412) Reported by: jiddings ........
-
-2009-12-08 07:41 +0000 [r233689] TransNexus OSP Development <support@transnexus.com>
-
- * apps/app_osplookup.c: Fixed compile error with OSP Toolkit 3.6.
-
-2009-12-07 23:54 +0000 [r233615] 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 23:29 +0000 [r233473-233612] 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 [r233396] 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 21:55 +0000 [r233281] 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 21:07 +0000 [r233240] 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 17:36 +0000 [r233165] 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 17:23 +0000 [r233130] 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 15:57 +0000 [r233049] 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 [r232866] 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 15:14 +0000 [r232813] 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 00:20 +0000 [r232675-232678] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_musiconhold.c: Oops, really remove it this time
-
- * res/res_musiconhold.c, /: Recorded merge of 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 23:28 +0000 [r232658] David Vossel <dvossel@digium.com>
-
- * CHANGES, /, UPGRADE.txt: Merged revisions 232657 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r232657 | dvossel | 2009-12-02 17:27:45 -0600 (Wed, 02 Dec 2009)
- | 6 lines update CHANGES and UPGRADE.txt for early media behavior
- change between 1.6.1 and 1.6.2 (closes issue #16212) Reported by:
- miki ........
-
-2009-12-02 22:05 +0000 [r232579-232585] 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 17:11 +0000 [r232359] 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 17:01 +0000 [r232352] 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 16:43 +0000 [r232348] 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 15:43 +0000 [r232270] 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 14:55 +0000 [r232232] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 232230 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r232230 |
- file | 2009-12-02 10:54:28 -0400 (Wed, 02 Dec 2009) | 5 lines Fix
- a bug where a scheduled item ID would get retained on
- registrations in a certain scenario causing code to execute
- during reload that should not. (issue AST-263) ........
-
-2009-12-02 00:52 +0000 [r232094] 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 23:40 +0000 [r232011-232015] 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 22:03 +0000 [r231930] 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 21:21 +0000 [r231870] 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 Leif Madsen <lmadsen@digium.com>
-
- * Release Asterisk 1.6.2.0-rc7
-
-2009-12-01 15:48 +0000 [r231743] 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 21:59 +0000 [r231695-231696] 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. ........
-
- * pbx/pbx_config.c: Backport a tiny fix from trunk that makes GCC
- 4.4.x happier.
-
-2009-11-30 21:36 +0000 [r231689] 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 20:47 +0000 [r231605] 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 18:57 +0000 [r231505-231558] 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 [r231302] 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 [r231191] 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 20:36 +0000 [r231136] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_queue.c, /: Merged revisions 231134 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r231134 |
- tilghman | 2009-11-24 14:31:28 -0600 (Tue, 24 Nov 2009) | 7 lines
- Found a few places where queue refcounts were counted
- incorrectly. Also add debug statements. (closes issue #15982,
- closes issue #15984) Reported by: atis Patches:
- 20091111__issue15982.diff.txt uploaded by tilghman (license 14)
- Tested by: atis ........
-
-2009-11-24 18:54 +0000 [r231098] 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 15:48 +0000 [r230884] Joshua Colp <jcolp@digium.com>
-
- * configs/sip.conf.sample, /, channels/chan_sip.c: 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 15:38 +0000 [r230796-230880] 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 22:36 +0000 [r230727] 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 21:07 +0000 [r230629] 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 17:31 +0000 [r230510-230585] David Vossel <dvossel@digium.com>
-
- * main/audiohook.c, /, include/asterisk/audiohook.h: 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/ ........ ................
-
-2009-11-16 16:41 +0000 [r230250-230384] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, apps/app_fax.c: Merged revisions 230381 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r230381 |
- kpfleming | 2009-11-16 10:40:25 -0600 (Mon, 16 Nov 2009) | 1 line
- Fix another buglet in T.38 session teardown at the end of FAX
- sessions. ........
-
- * /, apps/app_fax.c: Merged revisions 230343 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r230343 |
- kpfleming | 2009-11-16 06:51:59 -0600 (Mon, 16 Nov 2009) | 2
- lines Ensure that only one end of a T.38 session initiates
- teardown at completion. ........
-
- * channels/chan_iax2.c, /: Merged revisions 230247 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r230247 | kpfleming | 2009-11-15 11:23:02 -0600
- (Sun, 15 Nov 2009) | 12 lines Merged revisions 230246 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r230246 | kpfleming | 2009-11-15 11:19:06 -0600 (Sun, 15 Nov
- 2009) | 6 lines Correct mistaken option name in error message.
- The configuration option for allowing hosts to make
- non-token-based calls is 'calltokenoptional', not
- 'calltokenignore'. (reported on asterisk-users) ........
- ................
-
-2009-11-13 22:01 +0000 [r229969-230148] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 230145 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r230145 | file | 2009-11-13 16:00:44 -0600 (Fri, 13 Nov 2009) |
- 15 lines Merged revisions 230144 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r230144 | file | 2009-11-13 16:00:19 -0600 (Fri, 13 Nov 2009) | 8
- lines Respect the maddr parameter in the Via header. (closes
- issue #14446) Reported by: frawd Patches: via_maddr.patch
- uploaded by frawd (license 610) Tested by: frawd ........
- ................
-
- * channels/chan_local.c, /: Merged revisions 230039 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r230039 | file | 2009-11-13 13:44:53 -0600 (Fri,
- 13 Nov 2009) | 16 lines Merged revisions 230038 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r230038 | file | 2009-11-13 13:44:07 -0600 (Fri, 13 Nov 2009) | 9
- lines Fix a crash caused by two threads thinking they should both
- free the chan_local private structure when only one should.
- (closes issue #15314) Reported by: sroberts Patches:
- Issue15314_Move_Nulling_owner.patch uploaded by davidw (license
- 780) Tested by: davidw, lottc ........ ................
-
- * configs/extensions.conf.sample, /, apps/app_chanisavail.c: Merged
- revisions 229966 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r229966 | file | 2009-11-13 11:20:26 -0600 (Fri, 13 Nov 2009) |
- 13 lines Merged revisions 229965 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r229965 | file | 2009-11-13 11:19:59 -0600 (Fri, 13 Nov 2009) | 6
- lines Document a limitation in the AVAILSTATUS variable from
- ChanIsAvail and provide a workaround for it that does not change
- existing behavior. (closes issue #14426) Reported by: macli
- ........ ................
-
-2009-11-13 Leif Madsen <lmadsen@digium.com>
-
- * Release Asterisk 1.6.2.0-rc6
-
-2009-11-13 15:57 +0000 [r229915] 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 23:31 +0000 [r229752] Jason Parker <jparker@digium.com>
-
- * channels/chan_oss.c, /: Merged revisions 229750 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r229750 |
- qwell | 2009-11-12 17:30:10 -0600 (Thu, 12 Nov 2009) | 1 line Fix
- mute toggling on OSS channels. ........
-
-2009-11-12 16:47 +0000 [r229671] 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 20:49 +0000 [r229570] David Ruggles <thedavidfactor@gmail.com>
-
- * doc/externalivr.txt: Merged revisions 229568 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r229568 |
- diruggles | 2009-11-11 15:47:06 -0500 (Wed, 11 Nov 2009) | 9
- lines Remove non-functional feature from ExternalIVR
- documentation Remove non-functional socket implementation of
- ExternalIVR from documentation (closes issue #16225) Reported by:
- thedavidfactor Patches: externalivr.txt.20091111.1542.patch
- uploaded by thedavidfactor (license 903) ........
-
-2009-11-11 19:56 +0000 [r229492-229502] 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 [r229366] 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 22:04 +0000 [r229359] 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 21:33 +0000 [r229354] 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 20:09 +0000 [r229285] 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 17:52 +0000 [r229232] 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 17:39 +0000 [r229231] 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 15:47 +0000 [r229101] Matthew Nicholson <mnicholson@digium.com>
-
- * UPGRADE-1.6.txt, main/editline/makelist.in, UPGRADE.txt: Reset
- props that were accidently deleted in 229088.
-
-2009-11-10 15:28 +0000 [r229094] David Vossel <dvossel@digium.com>
-
- * res/res_config_pgsql.c, /: Merged revisions 229093 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r229093 | dvossel | 2009-11-10 09:27:45 -0600 (Tue, 10 Nov 2009)
- | 11 lines fixes pgsql double free of threadstorage A thread
- storage variable was being freed incorrectly, which resulted in a
- double free if two queries were made in the same thread. (closes
- issue #16011) Reported by: cristiandimache Patches:
- issue16011.diff uploaded by dvossel (license 671) ........
-
-2009-11-10 15:16 +0000 [r229088] Matthew Nicholson <mnicholson@digium.com>
-
- * UPGRADE-1.6.txt, main/editline/makelist.in, channels/chan_sip.c,
- UPGRADE.txt: Reverted revision 202007. (closes issue #16175)
- Reported by: paul-tg Tested by: paul-tg
-
-2009-11-10 11:25 +0000 [r229078] 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 22:59 +0000 [r229017] Terry Wilson <twilson@digium.com>
-
- * channels/chan_local.c, /: Merged revisions 229015 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r229015 | twilson | 2009-11-09 16:50:22 -0600 (Mon, 09 Nov 2009)
- | 8 lines Don't crash when bridge->tech_pvt == NULL This is a
- similar solution to what is in place for chan_agent (closes issue
- #16003) Reported by: atis Tested by: twilson ........
-
-2009-11-09 22:17 +0000 [r229012] David Vossel <dvossel@digium.com>
-
- * channels/chan_sip.c: fixes segfault when transferring a queue
- caller In sip_hangup we attempted to lock p->owner after we set
- it to NULL. Thanks to fhackenberger for reporting the issue and
- submitting a patch. (closes issue #15848) Reported by:
- fhackenberger Patches: digium_bug_0015848 uploaded by
- fhackenberger (license 592) Tested by: fhackenberger, lmadsen,
- TomS, shin-shoryuken, dvossel
-
-2009-11-09 Leif Madsen <lmadsen@digium.com>
-
- * Release Asterisk 1.6.2.0-rc5
-
-2009-11-09 15:40 +0000 [r228900] 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 14:48 +0000 [r228859] 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 22:37 +0000 [r228694] 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 20:26 +0000 [r228649] 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 18:43 +0000 [r228551] 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 1.6.2.0-rc4
-
-2009-11-06 17:54 +0000 [r228504] 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. ........
-
-2009-11-06 17:24 +0000 [r228421-228447] David Vossel <dvossel@digium.com>
-
- * codecs/codec_ilbc.c, /: Merged revisions 228441 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r228441 |
- dvossel | 2009-11-06 11:22:31 -0600 (Fri, 06 Nov 2009) | 3 lines
- Fixes merging issue from 1.4, frame data is held in data.ptr in
- trunk ........
-
- * 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 16:44 +0000 [r228413] Joshua Colp <jcolp@digium.com>
-
- * /, 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 15:43 +0000 [r228269-228340] 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 22:13 +0000 [r228198] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 228196 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r228196 |
- tilghman | 2009-11-05 16:12:45 -0600 (Thu, 05 Nov 2009) | 2 lines
- Yet another error message in the dialplan (thanks,
- rmudgett/russellb) ........
-
-2009-11-05 21:27 +0000 [r228195] 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 21:27 +0000 [r228194] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_meetme.c: Merged revisions 228191 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r228191 |
- tilghman | 2009-11-05 15:24:21 -0600 (Thu, 05 Nov 2009) | 7 lines
- MEETME_INFO should not return a literal error message to the
- dialplan. (closes issue #15450) Reported by: JimVanM Patches:
- meetmeinfopatch.diff.txt uploaded by dbrooks (license 790) Tested
- by: JimVanM ........
-
-2009-11-05 19:42 +0000 [r228148] 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 19:20 +0000 [r228093] 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 ........ ................
-
-2009-11-05 17:14 +0000 [r228017] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_externalivr.c, /: Merged revisions 228015 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r228015 | tilghman | 2009-11-05 11:08:02 -0600 (Thu, 05 Nov 2009)
- | 4 lines Don't crash if no arguments are passed. (closes issue
- #16119) Reported by: thedavidfactor ........
-
-2009-11-04 23:53 +0000 [r227947] 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 21:09 +0000 [r227760-227831] 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 19:28 +0000 [r227733-227748] 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 20:01 +0000 [r227375] 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 [r227364-227371] 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 18:15 +0000 [r227280] 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 17:14 +0000 [r227239] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 227238 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r227238 |
- dvossel | 2009-11-03 11:12:52 -0600 (Tue, 03 Nov 2009) | 5 lines
- user.conf entries in SIP were not having their peer type set.
- (closes issue #16120) Reported by: jsmith ........
-
-2009-11-03 15:40 +0000 [r227170] 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 15:25 +0000 [r227165] 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 13:51 +0000 [r227156] Olle Johansson <oej@edvina.net>
-
- * Makefile, /, 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 21:06 +0000 [r226978] 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/
-
-2009-11-02 18:12 +0000 [r226893] 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 17:17 +0000 [r226815] 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 18:18 +0000 [r226540] Joshua Colp <jcolp@digium.com>
-
- * doc/tex/localchannel.tex, channels/chan_local.c, /: 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 21:32 +0000 [r226486] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
-
- * build_tools/get_documentation, /: remove empty awk pattern (//)
- Solaris 10 nawk doesn't like the empty pattern such as '//' for
- 'always'. Just remove that. No pattern at all always matches.
- Merged revisions 226453 via svnmerge from
- http://svn.digium.com/svn/asterisk/trunk
-
-2009-10-28 20:13 +0000 [r226379-226385] 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 18:06 +0000 [r226170-226308] 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 ........ ................
-
-2009-10-27 17:12 +0000 [r226101] Terry Wilson <twilson@digium.com>
-
- * res/res_http_post.c, /: Merged revisions 226099 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r226099 |
- twilson | 2009-10-27 11:48:54 -0500 (Tue, 27 Oct 2009) | 2 lines
- Don't prepend the URI prefix to the post directory ........
-
-2009-10-27 00:16 +0000 [r226055] 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 19:42 +0000 [r225914] Jeff Peeler <jpeeler@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 225912 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r225912 |
- jpeeler | 2009-10-26 14:40:26 -0500 (Mon, 26 Oct 2009) | 12 lines
- ACL check not present for verifying SIP INVITEs The ACL check in
- check_peer_ok was missing and has now been restored. The missing
- check allowed for calls to be made on prohibited networks where
- an ACL was defined in sip.conf and the allowguest option was set
- to off. See the AST security advisory below for more information.
- Merge code associated with AST-2009-007. (closes issue #16091)
- Reported by: thom4fun ........
-
-2009-10-26 15:56 +0000 [r225871] 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. (closes issue #16127)
-
-2009-10-23 14:46 +0000 [r225651] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 225650 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r225650 |
- dvossel | 2009-10-23 09:41:50 -0500 (Fri, 23 Oct 2009) | 3 lines
- Fixes an iterator memory leak and uninitialized memory ........
-
-2009-10-23 14:08 +0000 [r225585] 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 22:24 +0000 [r225516] Leif Madsen <lmadsen@digium.com>
-
- * README, /: Merged revisions 225515 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r225515 |
- lmadsen | 2009-10-22 17:24:03 -0500 (Thu, 22 Oct 2009) | 8 lines
- Update README documentation. Update the README documentation to
- correctly describe which CLI command you should use when
- attempting to get help from the CLI. (closes issue #16064)
- Reported by: thedavidfactor Patches: readme.patch uploaded by
- thedavidfactor (license 903) ........
-
-2009-10-22 21:55 +0000 [r225489] David Vossel <dvossel@digium.com>
-
- * apps/app_externalivr.c, include/asterisk/tcptls.h, main/tcptls.c,
- /, channels/chan_sip.c: Merged revisions 225445 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r225445 |
- dvossel | 2009-10-22 14:55:51 -0500 (Thu, 22 Oct 2009) | 50 lines
- SIP TCP/TLS: move client connection setup/write into tcp helper
- thread, various related locking/memory fixes. What this patch
- fixes 1.Moves sip TCP/TLS connection setup into the TCP helper
- thread: Connection setup takes awhile and before this it was
- being done while holding the monitor lock. 2.Moves TCP/TLS
- writing to the TCP helper thread: Through the use of a packet
- queue and an alert pipe, the TCP helper thread can now be woken
- up to write data as well as read data. 3.Locking error: sip_xmit
- returned an XMIT_ERROR without giving up the tcptls_session lock.
- This lock has been completely removed from sip_xmit and placed in
- the new sip_tcptls_write() function. 4.Memory leak: When creating
- a tcptls_client the tls_cfg was alloced but never freed unless
- the tcptls_session failed to start. Now the session_args for a
- sip client are an ao2 object which frees the tls_cfg on
- destruction. 5.Pointer to stack variable: During
- sip_prepare_socket the creation of a client's
- ast_tcptls_session_args was done on the stack and stored as a
- pointer in the newly created tcptls_session. Depending on the
- events that followed, there was a slight possibility that pointer
- could have been accessed after the stack returned. Given the new
- changes, it is always accessed after the stack returns which is
- why I found it. Notable code changes 1.I broke tcptls.c's
- ast_tcptls_client_start() function into two functions. One for
- creating and allocating the new tcptls_session, and a separate
- one for starting and handling the new connection. This allowed me
- to create the tcptls_session, launch the helper thread, and then
- establish the connection within the helper thread. 2.Writes to a
- tcptls_session are now done within the helper thread. This is
- done by using an alert pipe to wake up the thread if new data
- needs to be sent. The thread's sip_threadinfo object contains the
- alert pipe as well as the packet queue. 3.Since the threadinfo
- object contains the alert pipe, it must now be accessed outside
- of the helper thread for every write (queuing of a packet). For
- easy lookup, I moved the threadinfo objects from a linked list to
- an ao2_container. (closes issue #13136) Reported by: pabelanger
- Tested by: dvossel, whys (closes issue #15894) Reported by:
- dvossel Tested by: dvossel Review:
- https://reviewboard.asterisk.org/r/380/ ........
-
-2009-10-22 21:54 +0000 [r225488] 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 17:14 +0000 [r225363] 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 22:00 +0000 [r225035-225308] 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/sip.conf.sample, channels/chan_iax2.c,
- configs/iax.conf.sample, /, channels/chan_sip.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/ ........ ................
-
-2009-10-20 22:11 +0000 [r224859] Tilghman Lesher <tlesher@digium.com>
-
- * main/audiohook.c, funcs/func_speex.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 17:50 +0000 [r224777] 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 ........ ................
-
-2009-10-20 00:00 +0000 [r224674] 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 19:54 +0000 [r224571] 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 19:41 +0000 [r224563] Kevin P. Fleming <kpfleming@digium.com>
-
- * formats/format_siren14.c, /: Merged revisions 224562 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r224562 | kpfleming | 2009-10-19 14:40:26 -0500 (Mon, 19 Oct
- 2009) | 1 line Remove useless debugging message. ........
-
-2009-10-19 00:13 +0000 [r224447-224451] 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, and remove multiple goto's. This corrects an issue
- reported on the -users list. ........
-
- * configs/res_odbc.conf.sample, /: Merged revisions 224446 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r224446 | tilghman | 2009-10-18 18:41:30 -0500 (Sun, 18
- Oct 2009) | 2 lines Clarify that "forcecommit" is NOT an alias
- for "autocommit", but instead controls the default disposition of
- uncommitted transactions. ........
-
-2009-10-17 01:58 +0000 [r224334] Jeff Peeler <jpeeler@digium.com>
-
- * 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 20:58 +0000 [r224264] 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 15:58 +0000 [r224181] 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 23:55 +0000 [r223835] 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 21:01 +0000 [r223757] 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 14:37 +0000 [r223655] 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 17:32 +0000 [r223490] 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 23:12 +0000 [r223406] 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 21:01 +0000 [r223333] 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 18:53 +0000 [r223286] 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 18:29 +0000 [r223257] 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 17:55 +0000 [r223208] 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 [r223173] 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 17:09 +0000 [r223089-223133] 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 20:00 +0000 [r222883] Russell Bryant <russell@digium.com>
-
- * include/asterisk/frame.h, include/asterisk/file.h, main/frame.c,
- /, main/file.c: 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 19:41 +0000 [r222874] 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 16:51 +0000 [r222695-222802] 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 18:06 +0000 [r222549] Jason Parker <jparker@digium.com>
-
- * /, configs/queues.conf.sample: Merged revisions 222548 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r222548 | qwell | 2009-10-07 13:04:56 -0500 (Wed, 07 Oct
- 2009) | 5 lines Remove 'keepstats' queue option from sample
- config, as it's no longer used.
- https://reviewboard.asterisk.org/r/115/ (closes issue #15820)
- Reported by: kshumard ........
-
-2009-10-07 18:00 +0000 [r222547] Sean Bright <sean@malleable.com>
-
- * funcs/func_strings.c: Fix merge error.
-
-2009-10-07 17:45 +0000 [r222544] 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) ........ ................
-
-2009-10-06 23:59 +0000 [r222354-222466] 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, and 1.6.2) (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)
-
- * 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) ........
-
-2009-10-06 19:41 +0000 [r222311] Tilghman Lesher <tlesher@digium.com>
-
- * cdr/cdr_pgsql.c, res/res_config_pgsql.c, /: Merged revisions
- 222309 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r222309 |
- tilghman | 2009-10-06 14:31:39 -0500 (Tue, 06 Oct 2009) | 10
- lines Change schema query to involve the use of an optional
- schema parameter. This change is done in such a way as to allow
- the driver to continue to function with older databases which
- don't have these features. (closes issue #16000) Reported by:
- jamicque Patches: 20091002__issue16000.diff.txt uploaded by
- tilghman (license 14) 20091002__issue16000__1.6.1.diff.txt
- uploaded by tilghman (license 14) Tested by: jamicque ........
-
-2009-10-06 19:27 +0000 [r222304] Jeff Peeler <jpeeler@digium.com>
-
- * 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 19:22 +0000 [r222289] Tilghman Lesher <tlesher@digium.com>
-
- * res/ael/pval.c, /: Merged 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 ........
-
-2009-10-06 Leif Madsen <lmadsen@digium.com>
-
- * Released Asterisk 1.6.2.0-rc3
-
-2009-10-06 01:39 +0000 [r222113-222187] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/chan_console.c, res/res_musiconhold.c, apps/app_queue.c,
- channels/chan_iax2.c, main/astobj2.c, res/res_odbc.c,
- res/res_clialiases.c, /, channels/chan_sip.c,
- funcs/func_dialgroup.c, include/asterisk/astobj2.h,
- res/res_phoneprov.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/ ........ ................
-
- * configs/sip.conf.sample, main/udptl.c, /, channels/chan_sip.c,
- configs/udptl.conf.sample, UPGRADE.txt: Merged 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 17:35 +0000 [r222032] 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 17:01 +0000 [r221923-221974] 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. ........ ................
-
- * /, 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 01:35 +0000 [r221879] 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 00:07 +0000 [r221744-221780] 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 ........
- ................
-
- * /, 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 19:35 +0000 [r221698] 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 16:57 +0000 [r221660] 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 16:25 +0000 [r221622] 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 23:07 +0000 [r221477-221485] Matthew Nicholson <mnicholson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 221484 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r221484 |
- mnicholson | 2009-09-30 18:04:03 -0500 (Wed, 30 Sep 2009) | 2
- lines Cleaned up merge from r221432 ........
-
- * configs/sip.conf.sample, /, channels/chan_sip.c: 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 21:46 +0000 [r221371-221472] Matthias Nick <mnick@digium.com>
-
- * apps/app_queue.c, /: Merged revisions 221436 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r221436 |
- mnick | 2009-09-30 16:15:01 -0500 (Wed, 30 Sep 2009) | 2 lines
- Prevents from division by zero ........
-
- * 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 19:15 +0000 [r221304] Terry Wilson <twilson@digium.com>
-
- * configs/sip.conf.sample, main/rtp.c, /, channels/chan_sip.c,
- include/asterisk/rtp.h: 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 [r221204] 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 14:57 +0000 [r221089] 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 04:41 +0000 [r221027-221047] 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
- ........
-
- * include/asterisk/smdi.h, include/asterisk/optional_api.h
- (removed), apps/app_voicemail.c, include/asterisk/agi.h,
- include/asterisk/monitor.h: Remove optional_api from 1.6.2
- branch, since it is not currently working. This is a blocking
- issue for the 1.6.2 release. (closes issue #15914) Reported by:
- mbeckwell Branch:
- http://svn.digium.com/svn/asterisk/team/tilghman/optional_api_162
- Tested by: mbeckwell
-
- * /, 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 20:24 +0000 [r220905-220934] 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
-
- * /, apps/app_confbridge.c: Merged revisions 220904 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r220904 | mnicholson | 2009-09-29 14:49:02 -0500 (Tue, 29 Sep
- 2009) | 5 lines Fix options 'm' and 's'. They were swapped in the
- code. Also document the fact that app_confbridge does not
- automatically answer the channel. (closes issue #15964) Reported
- by: shrift ........
-
-2009-09-29 17:06 +0000 [r220836] 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 [r220725] 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. ........
- ................
-
-2009-09-28 19:11 +0000 [r220722] Jeff Peeler <jpeeler@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 220718 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r220718 |
- jpeeler | 2009-09-28 14:10:10 -0500 (Mon, 28 Sep 2009) | 10 lines
- Fix building of registration entry in build_peer when using
- callbackextension Check for remotesecret option was
- unintentionally always true, which therefore caused the secret
- option to never be used. Thanks to dvossel for pointing out the
- exact fix. (closes issue #15943) Reported by: tpsast ........
-
-2009-09-27 20:45 +0000 [r220632] Michiel van Baak <michiel@vanbaak.info>
-
- * funcs/func_callerid.c, /: Merged revisions 220629 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r220629 | mvanbaak | 2009-09-27 22:40:16 +0200 (Sun, 27 Sep 2009)
- | 3 lines add name argument for the CALLERID dialplan function to
- the xml documentation. Pointed out to me on IRC by snuff-home.
- Thanks ........
-
-2009-09-26 15:12 +0000 [r220589] 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 20:38 +0000 [r220369] 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 [r220292] 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 [r220103-220221] 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 08:43 +0000 [r220031] 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 ........ ................
-
-2009-09-24 07:45 +0000 [r219989] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_directory.c, /: Merged revisions 219987 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r219987 | tilghman | 2009-09-24 02:39:44 -0500 (Thu, 24 Sep 2009)
- | 8 lines Fix two possible crashes, one only in 1.6.1 and one in
- 1.6.1 forward. (closes issue #15739) Reported by: DLNoah, jeffg
- Patches: 20090914__issue15739.diff.txt uploaded by tilghman
- (license 14) 20090922__issue15739.diff.txt uploaded by tilghman
- (license 14) Tested by: DLNoah, jeffg ........
-
-2009-09-22 21:48 +0000 [r219821] 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 17:01 +0000 [r219722] 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 18:21 +0000 [r219669] 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 03:14 +0000 [r219590] 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 23:21 +0000 [r219452-219521] 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 [r219415] 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 22:38 +0000 [r219376] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 219371 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r219371 |
- dvossel | 2009-09-17 17:37:28 -0500 (Thu, 17 Sep 2009) | 9 lines
- fixes deadlock when performing directed pickup w Invite/replaces
- (closes issue #15340) Reported by: lmsteffan Patches:
- deadlock.patch uploaded by lmsteffan (license 779) Tested by:
- lmsteffan ........
-
-2009-09-17 22:37 +0000 [r219370] 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 22:06 +0000 [r219307] 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 [r219267] 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 1.6.2.0-rc2
-
-2009-09-17 15:38 +0000 [r219194] 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 [r219063] 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 19:40 +0000 [r218938] 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 19:29 +0000 [r218937] 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 19:25 +0000 [r218934] 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 13:38 +0000 [r218802] 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 22:46 +0000 [r218733] 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 19:24 +0000 [r218688] 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 16:18 +0000 [r218590] 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 16:06 +0000 [r218582] 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 15:59 +0000 [r218576] 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 ........ ................
-
-2009-09-15 15:42 +0000 [r218507-218575] 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 15:05 +0000 [r218503] 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 19:49 +0000 [r218364] Tilghman Lesher <tlesher@digium.com>
-
- * sounds/Makefile, apps/app_voicemail.c, /,
- configs/voicemail.conf.sample: 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 18:18 +0000 [r218300] Joshua Colp <jcolp@digium.com>
-
- * /, main/features.c: Merged revisions 218295 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r218295 |
- file | 2009-09-14 13:16:39 -0500 (Mon, 14 Sep 2009) | 2 lines Do
- not attempt to add a parking extension if an error occurred while
- reading the configuration. ........
-
-2009-09-14 15:20 +0000 [r218238] 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 22:12 +0000 [r218219] 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
-
-2009-09-13 05:59 +0000 [r218151] Moises Silva <moises.silva@gmail.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 218150 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r218150 | moy | 2009-09-13 01:51:46 -0400 (Sun, 13 Sep 2009) | 1
- line get rid of mfcr2 monitor thread condition, is problematic
- ........
-
-2009-09-11 06:00 +0000 [r217926-218055] Tilghman Lesher <tlesher@digium.com>
-
- * main/pbx.c, /: Merged revisions 218050 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r218050 |
- tilghman | 2009-09-11 00:58:11 -0500 (Fri, 11 Sep 2009) | 3 lines
- Check the origination priority for more matches, not the current
- priority. Found by Pavel Troller on the -dev list. ........
-
- * 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) ........
- ................
-
- * channels/chan_iax2.c, contrib/scripts/iax-friends.sql, /,
- channels/chan_sip.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 21:21 +0000 [r217821] David Vossel <dvossel@digium.com>
-
- * 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 19:56 +0000 [r217739] 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 18:40 +0000 [r217665] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 216805 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r216805 |
- oej | 2009-09-07 18:08:08 +0200 (MÃ¥n, 07 Sep 2009) | 2 lines
- Since it's possible to have more than 999 calls, I'm changing the
- call counter roof to something higher. ........
-
-2009-09-10 18:19 +0000 [r217647] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_config_odbc.c, /, configure,
- include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
- 217638 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r217638 |
- tilghman | 2009-09-10 13:17:14 -0500 (Thu, 10 Sep 2009) | 4 lines
- Verify support for wide ODBC character types before using them.
- (closes issue #15870) Reported by: nic_bellamy ........
-
-2009-09-10 15:14 +0000 [r217632] Moises Silva <moises.silva@gmail.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 217524 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r217524 | moy | 2009-09-09 17:48:04 -0400 (Wed, 09 Sep 2009) | 1
- line ast_log replaced for ast_verbose in MFCR2 event
- notifications ........
-
-2009-09-10 12:09 +0000 [r217594] 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 20:37 +0000 [r217519] 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
- https://origsvn.digium.com/svn/asterisk/trunk
-
-2009-09-09 10:58 +0000 [r217369] 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 22:20 +0000 [r217299] Sean Bright <sean@malleable.com>
-
- * /, apps/app_meetme.c: Merged revisions 217286 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r217286 |
- seanbright | 2009-09-08 18:17:08 -0400 (Tue, 08 Sep 2009) | 4
- lines Fix compilation of app_meetme. Reported by ebroad in
- #asterisk-bugs ........
-
-2009-09-08 20:33 +0000 [r217217] 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 16:39 +0000 [r217077] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, configure, 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 15:36 +0000 [r217036] 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 14:27 +0000 [r216994] 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 16:43 +0000 [r216647-216845] 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 ........
-
- * CHANGES, UPGRADE.txt: Add docs
-
- * configs/sip.conf.sample, apps/app_playback.c, main/pbx.c, /,
- channels/chan_sip.c, apps/app_disa.c: 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 19:42 +0000 [r216598] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 216594 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r216594 |
- dvossel | 2009-09-04 14:32:07 -0500 (Fri, 04 Sep 2009) | 7 lines
- 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 Review:
- https://reviewboard.asterisk.org/r/354/ ........
-
-2009-09-04 19:32 +0000 [r216597] 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 17:53 +0000 [r216550-216553] Tilghman Lesher <tlesher@digium.com>
-
- * /, include/asterisk/lock.h: Merged revisions 216551 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r216551 | tilghman | 2009-09-04 12:50:21 -0500 (Fri, 04 Sep 2009)
- | 2 lines Fix trunk breakage. ........
-
- * UPGRADE-1.6.txt, main/pbx.c, /: 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 15:11 +0000 [r216469-216509] 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 ........ ................
-
- * /, include/asterisk/lock.h: Merged revisions 216437 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r216437 | mvanbaak | 2009-09-04 16:00:38 +0200 (Fri, 04 Sep 2009)
- | 2 lines make sure canlog is set so we can compile with
- DEBUG_THREADS enabled on OpenBSD ........
-
-2009-09-04 13:57 +0000 [r216267-216436] Russell Bryant <russell@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 216368 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r216368 |
- russell | 2009-09-04 08:14:25 -0500 (Fri, 04 Sep 2009) | 12 lines
- Do not treat every SIP peer as if they were configured with
- insecure=port. There was a problem in the function responsible
- for doing peer matching by IP address and port number such that
- during the second pass for checking for a peer configured with
- insecure=port, it would end up treating every peer as if it had
- been configured that way. These changes fix the logic in the peer
- IP and port comparison callback to handle insecure=port checking
- properly. This problem was introduced when SIP peers were
- converted to astobj2. Many thanks to dvossel for noticing this
- while working on another peer matching issue. ........
-
- * 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 06:14 +0000 [r216225] 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 19:44 +0000 [r216014-216099] Russell Bryant <russell@digium.com>
-
- * /, 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 18:42 +0000 [r216007] David Vossel <dvossel@digium.com>
-
- * channels/chan_iax2.c, 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: 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 14:21 +0000 [r215887-215929] Olle Johansson <oej@edvina.net>
-
- * /, channels/chan_sip.c: Merged revisions 215891 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r215891 |
- oej | 2009-09-03 15:02:41 +0200 (Tor, 03 Sep 2009) | 10 lines Add
- known internal IP address when autodomain=yes (closes issue
- #14573) Reported by: pj Patches: sip-internip-autodomain1.diff
- uploaded by mnicholson (license 96) modified by oej Tested by: pj
- ........
-
- * main/rtp.c, channels/chan_sip.c: Fix bad reports in "sip show
- channelstats". Not directly mergeable in svn trunk, needs more
- tests, therefore committed directly to 1.6.2. (closes issue
- #15819) Reported by: klaus3000 Patches:
- asterisk-1.6.2-beta4-sipshowchannelstats-patch-0.2.txt uploaded
- by klaus3000 (license 65) Tested by: klaus3000, oej
-
-2009-09-03 06:02 +0000 [r215841] Michiel van Baak <michiel@vanbaak.info>
-
- * doc/manager_1_1.txt, /: Merged revisions 215838 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r215838 |
- mvanbaak | 2009-09-03 07:57:23 +0200 (Thu, 03 Sep 2009) | 5 lines
- Document that SIPshowpeer and SKINNYshowline now include the
- configured parkinglot in their response. Prodded by snuff-work on
- #asterisk-dev IRC channel ........
-
-2009-09-03 03:44 +0000 [r215802] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 215801 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r215801 |
- tilghman | 2009-09-02 22:43:51 -0500 (Wed, 02 Sep 2009) | 5 lines
- Default the callback extension to "s". This is a regression.
- (closes issue #15764) Reported by: elguero Change-type: bugfix
- ........
-
-2009-09-03 00:34 +0000 [r215795] Terry Wilson <twilson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 215758 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r215758 | twilson | 2009-09-02 18:31:04 -0500 (Wed, 02 Sep 2009)
- | 25 lines Merged revisions 215682 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r215682 | twilson | 2009-09-02 16:41:22 -0500 (Wed, 02 Sep 2009)
- | 18 lines Re-send non-100 provisional responses to prevent
- cancellation From section 13.3.1.1 of RFC 3261: If the UAS
- desires an extended period of time to answer the INVITE, it will
- need to ask for an "extension" in order to prevent proxies from
- canceling the transaction. A proxy has the option of canceling a
- transaction when there is a gap of 3 minutes between responses in
- a transaction. To prevent cancellation, the UAS MUST send a
- non-100 provisional response at every minute, to handle the
- possibility of lost provisional responses. (closes issue #11157)
- Reported by: rjain Tested by: twilson Review:
- https://reviewboard.asterisk.org/r/315/ ........ ................
-
-2009-09-02 21:46 +0000 [r215683] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 215681 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r215681 |
- dvossel | 2009-09-02 16:39:31 -0500 (Wed, 02 Sep 2009) | 10 lines
- port string to int conversion using sscanf There are several
- instances where a port is parsed from a uri or some other source
- and converted to an int value using atoi(), if for some reason
- the port string is empty, then a standard port is used. This
- logic is used over and over, so I created a function to handle it
- in a safer way using sscanf(). ........
-
-2009-09-02 21:37 +0000 [r215647-215680] Michiel van Baak <michiel@vanbaak.info>
-
- * /, channels/chan_sip.c, channels/chan_skinny.c: Merged revisions
- 215665 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r215665 |
- mvanbaak | 2009-09-02 23:23:17 +0200 (Wed, 02 Sep 2009) | 5 lines
- add Parkinglot info to sip show peer <foo> and skinny show line
- <foo> If we had this from the start, debugging the 'parking not
- using configured parkinglot' bug would have been easier. ........
-
- * /, main/features.c: Merged revisions 215622 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r215622 |
- mvanbaak | 2009-09-02 22:21:51 +0200 (Wed, 02 Sep 2009) | 4 lines
- - lock channel before looking for a channel variable - Init the
- parkings list member of struct parkinglot. Thanks Sean for the
- explanation why this should be here. ........
-
-2009-09-02 18:52 +0000 [r215569-215570] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/Makefile, main/app.c: Merged revisions 215567 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r215567 | tilghman | 2009-09-02 13:37:25 -0500 (Wed, 02
- Sep 2009) | 9 lines Close up to the soft open file limit (same on
- Linux, but varies drastically on OS X). Also, a Makefile fix for
- Darwin (OS X). (closes issue #14542) Reported by: jtodd Patches:
- 20090901__issue14542.diff.txt uploaded by tilghman (license 14)
- Tested by: jtodd, tilghman Change-type: bugfix ........
-
- * /, channels/chan_sip.c: Merged revisions 215222 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r215222 |
- tilghman | 2009-09-01 16:19:40 -0500 (Tue, 01 Sep 2009) | 3 lines
- Fix register such that lines with a transport string, but without
- an authuser, parse correctly. (AST-228) ........
-
-2009-09-02 17:35 +0000 [r215523] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 215522 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r215522 |
- dvossel | 2009-09-02 12:26:40 -0500 (Wed, 02 Sep 2009) | 11 lines
- SIP uri parsing cleanup Now, the scheme passed to parse_uri can
- either be a single scheme, or a list of schemes ',' delimited.
- This gets rid of the whole problem of having to create two
- buffers and calling parse_uri twice to check for separate
- schemes. Review: https://reviewboard.asterisk.org/r/343/ ........
-
-2009-09-02 16:35 +0000 [r215512] Michiel van Baak <michiel@vanbaak.info>
-
- * /, channels/chan_skinny.c: Merged revisions 215479 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r215479 | mvanbaak | 2009-09-02 18:20:23 +0200 (Wed, 02 Sep 2009)
- | 3 lines like in chan_sip's sip_new skinny should copy the
- configured parkinglot from a line to the newly created channel.
- This makes callparking honor the configured parkinglot for skinny
- lines as well. ........
-
-2009-09-02 16:09 +0000 [r215467] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 215466 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r215466 |
- dvossel | 2009-09-02 11:08:00 -0500 (Wed, 02 Sep 2009) | 16 lines
- SIP support for keep-alive event keep-alive events are used by
- Sipura/Linksys for NAT keepalive. There currently don't appear to
- be any problems with NAT, but everytime a keep-alive event is
- received, Asterisk responds with a "489 Bad event". This error
- may indicate to a user that NAT problems exist just because this
- even is not supported. Now, rather than respond with an error,
- the packet is consumed and a "200 ok" is sent just to indicate we
- received the packet. (issue #15084) Patches:
- chan_sip.keepalive.v1.diff uploaded by IgorG (license 20)
- ........
-
-2009-09-02 16:07 +0000 [r215465] Michiel van Baak <michiel@vanbaak.info>
-
- * /, channels/chan_sip.c: Merged revisions 215462 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r215462 |
- mvanbaak | 2009-09-02 17:56:46 +0200 (Wed, 02 Sep 2009) | 12
- lines Honor configured parkinglot when parking and retrieving
- parked calls Thank oej for pointing out the fact that sip_new did
- not copy parkinglot from the peer into the newly created channel.
- (closes issue #15538) Reported by: gracedman Patches:
- 2009090100_sipnewparkinglot-161.diff.txt uploaded by mvanbaak
- (license 7) With mod by me to also fix callparking as well (this
- uploaded patch only fixed retrieving a parked call) Tested by:
- gracedman, mvanbaak ........
-
-2009-09-02 01:49 +0000 [r215376] Dwayne M. Hubbard <dwayne.hubbard@gmail.com>
-
- * /, apps/app_softhangup.c: Merged revisions 215338 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r215338 | dhubbard | 2009-09-01 20:16:59 -0500
- (Tue, 01 Sep 2009) | 18 lines Merged revisions 215270 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r215270 | dhubbard | 2009-09-01 18:04:52 -0500 (Tue, 01 Sep 2009)
- | 12 lines Use strrchr() so SoftHangup will correctly truncate
- multi-hyphen channel names In general channel names are in the
- form Foo/Bar-Z, but the channel name could have multiple hyphens
- and look like Foo/B-a-r-Z. Use strrchr to truncate the channel
- name at the last hyphen. (closes issue #15810) Reported by:
- dhubbard Patches: dw-softhangup-1.4.patch uploaded by dhubbard
- (license 733) ........ ................
-
-2009-09-01 20:00 +0000 [r215165] Kevin P. Fleming <kpfleming@digium.com>
-
- * main/frame.c, /: Merged revisions 215161 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r215161 |
- kpfleming | 2009-09-01 14:50:48 -0500 (Tue, 01 Sep 2009) | 3
- lines Ensure that frame dumps of AST_CONTROL_T38_PARAMETERS
- frames are properly decoded. ........
-
-2009-08-31 16:22 +0000 [r214822-214960] Tilghman Lesher <tlesher@digium.com>
-
- * channels/chan_local.c, /: Merged revisions 214945 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r214945 | tilghman | 2009-08-31 11:18:33 -0500
- (Mon, 31 Aug 2009) | 14 lines Merged revisions 214940 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r214940 | tilghman | 2009-08-31 11:16:52 -0500 (Mon, 31 Aug 2009)
- | 7 lines Also unlock the "other" channel, when returning, due to
- glare. (closes issue #15787) Reported by: tim_ringenbach Patches:
- chan_local.diff uploaded by tim ringenbach (license 540) Tested
- by: tim_ringenbach ........ ................
-
- * Makefile, /: Merged revisions 214898 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r214898 |
- tilghman | 2009-08-30 17:10:35 -0500 (Sun, 30 Aug 2009) | 2 lines
- Force Darwin on ppc platforms to compile with a target level that
- supports aliasing. ........
-
- * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
- pbx/pbx_lua.c: Merged revisions 214819 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r214819 |
- tilghman | 2009-08-30 01:43:04 -0500 (Sun, 30 Aug 2009) | 4 lines
- If lua is detected with the lua5.1 prefix (or not), adjust the
- include path accordingly. Based upon feedback to a release
- announcement on the -users list. See
- http://lists.digium.com/pipermail/asterisk-users/2009-August/236954.html
- ........
-
-2009-08-29 Leif Madsen <lmadsen@digium.com>
-
- * Asterisk 1.6.2.0-rc1 released.
-
-2009-08-28 20:17 +0000 [r214707] Tilghman Lesher <tlesher@digium.com>
-
- * main/channel.c, /: Merged revisions 214702 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r214702 | tilghman | 2009-08-28 15:14:39 -0500 (Fri, 28 Aug 2009)
- | 15 lines Merged revisions 214701 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r214701 | tilghman | 2009-08-28 15:13:32 -0500 (Fri, 28 Aug 2009)
- | 8 lines Modify comment to be a bit more accurate. We have kept
- this comment around long enough, that it's pretty clear that
- we're keeping the code, because changing the code would require a
- pretty fundamental architectural shift. We've also taken
- criticism in some quarters, because it was believed that it was
- referring to the code being nasty. No, the code isn't nasty, just
- the operation itself is rather odd. Fixed for eternity (probably
- not). ........ ................
-
-2009-08-28 20:05 +0000 [r214700] Kevin P. Fleming <kpfleming@digium.com>
-
- * makeopts.in, Makefile, /, configure,
- include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
- 214696 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r214696 |
- kpfleming | 2009-08-28 15:01:21 -0500 (Fri, 28 Aug 2009) | 9
- lines Ensure that CFLAGS and/or LDFLAGS provided to configure
- script are preserved. Cross-compilation environments want to
- provide 'defaults' for compiler and linker options, and
- frequently do this by specifying CFLAGS and LDFLAGS in the
- environment or as command-line arguments to the configure script.
- This patch modifies the configure script and Makefile to preserve
- these settings and ensure they are used in the build process.
- ........
-
-2009-08-28 18:43 +0000 [r214653] Mark Michelson <mmichelson@digium.com>
-
- * /, include/asterisk/sched.h: Merged revisions 214650 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r214650 | mmichelson | 2009-08-28 13:41:23 -0500 (Fri, 28 Aug
- 2009) | 3 lines Fix some incorrect documentation of sched_thread
- functions. ........
-
-2009-08-27 21:49 +0000 [r214202-214521] Tilghman Lesher <tlesher@digium.com>
-
- * autoconf/libcurl.m4 (added), /, configure,
- include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
- 214518 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r214518 | tilghman | 2009-08-27 16:46:46 -0500 (Thu, 27 Aug 2009)
- | 14 lines Merged revisions 214517 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r214517 | tilghman | 2009-08-27 16:45:34 -0500 (Thu, 27 Aug 2009)
- | 7 lines Use autoconf to detect libcurl, as this enables
- cross-compilation checks, something we didn't allow before.
- (closes issue #15714) Reported by: pprindeville Patches:
- 20090813__issue15714.diff.txt uploaded by tilghman (license 14)
- Tested by: pprindeville ........ ................
-
- * main/manager.c, /: Merged revisions 214514 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r214514 |
- tilghman | 2009-08-27 16:26:37 -0500 (Thu, 27 Aug 2009) | 7 lines
- Ensure that we check for the special value
- CONFIG_STATUS_FILEINVALID. (closes issue #15786) Reported by:
- a_villacis Patches:
- asterisk-1.6.2.0-beta4-manager-fix-crash-on-include-nonexistent-file.patch
- uploaded by a villacis (license 660) (Plus a few of my own, to
- catch the remaining places within manager.c where it could have
- been a problem) ........
-
- * autoconf/ast_ext_lib.m4, /, configure,
- include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
- 214466 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r214466 | tilghman | 2009-08-27 12:28:01 -0500 (Thu, 27 Aug 2009)
- | 9 lines Merged revisions 214436 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r214436 | tilghman | 2009-08-27 11:53:58 -0500 (Thu, 27 Aug 2009)
- | 2 lines One more build system change, to make the descriptions
- look better, if we have better information. ........
- ................
-
- * autoconf/ast_ext_lib.m4, /, configure,
- include/asterisk/autoconfig.h.in: Merged revisions 214360 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r214360 | tilghman | 2009-08-27 11:12:03 -0500
- (Thu, 27 Aug 2009) | 10 lines Merged revisions 214357 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r214357 | tilghman | 2009-08-27 11:03:50 -0500 (Thu, 27 Aug 2009)
- | 3 lines Make autoheader descriptions render correctly in our
- autoconfig.h file. (Figured out while working with issue #14906)
- ........ ................
-
- * /, channels/chan_sip.c: Merged revisions 214199 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r214199 |
- tilghman | 2009-08-26 11:53:03 -0500 (Wed, 26 Aug 2009) | 6 lines
- Typo fix ("SIP/2.0 XXX" is 11 chars, not 10) (closes issue
- #15362) Reported by: klaus3000 Patches:
- chan_sip.c_logmessagefix_patch.txt uploaded by klaus3000 (license
- 65) ........
-
-2009-08-26 16:39 +0000 [r214196] David Vossel <dvossel@digium.com>
-
- * main/channel.c, /: Merged revisions 214195 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r214195 | dvossel | 2009-08-26 11:38:53 -0500 (Wed, 26 Aug 2009)
- | 25 lines Merged revisions 214194 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r214194 | dvossel | 2009-08-26 11:36:42 -0500 (Wed, 26 Aug 2009)
- | 19 lines ast_write() ignores ast_audiohook_write() results In
- ast_write(), if a channel has a list of audiohooks, those lists
- are written to and the resulting frame is what ast_write() should
- continue with. The problem was the returned audiohook frame was
- not being handled at all, and the original frame passed into it
- did not contain the mixed audio, so essentially audio was being
- lost. One result of this was chan_spy's whisper mode no longer
- worked. To complicate the issue, frames passed into ast_write may
- either be a single frame, or a list of frames. So, as the list of
- frames is processed in the audiohook_write, the returned frames
- had to be added to a new list. (closes issue #15660) Reported by:
- corruptor Tested by: dvossel ........ ................
-
-2009-08-25 22:43 +0000 [r213903-214155] Tilghman Lesher <tlesher@digium.com>
-
- * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
- Merged revisions 214152 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r214152 |
- tilghman | 2009-08-25 17:39:51 -0500 (Tue, 25 Aug 2009) | 4 lines
- Not all versions of gnu-linux use glibc, which contains iconv.
- Some (especially embedded systems) don't have iconv at all.
- (closes issue #15169) Reported by: pprindeville ........
-
- * /, main/say.c: Merged revisions 214071 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r214071 | tilghman | 2009-08-25 14:32:48 -0500 (Tue, 25 Aug 2009)
- | 17 lines Merged revisions 214068-214069 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r214068 | tilghman | 2009-08-25 14:26:50 -0500 (Tue, 25 Aug 2009)
- | 6 lines Fix pronunciation of German dates. (closes issue
- #15273) Reported by: Benjamin Kluck Patches: say_c.patch uploaded
- by Benjamin Kluck (license 803) ........ r214069 | tilghman |
- 2009-08-25 14:28:42 -0500 (Tue, 25 Aug 2009) | 2 lines I should
- always compile before committing... ........ ................
-
- * /, pbx/pbx_dundi.c: Merged revisions 213975 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r213975 |
- tilghman | 2009-08-25 01:51:12 -0500 (Tue, 25 Aug 2009) | 6 lines
- DUNDILOOKUP function in 1.6 should use comma delimiters. (closes
- issue #15322) Reported by: chappell Patches:
- dundilookup-0015322.patch uploaded by chappell (license 8)
- ........
-
- * main/pbx.c, /: Merged revisions 213971 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r213971 | tilghman | 2009-08-25 01:35:37 -0500 (Tue, 25 Aug 2009)
- | 14 lines Merged revisions 213970 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r213970 | tilghman | 2009-08-25 01:34:44 -0500 (Tue, 25 Aug 2009)
- | 7 lines Improve error message by informing user exactly which
- function is missing a parethesis. (closes issue #15242) Reported
- by: Nick_Lewis Patches: pbx.c-funcparenthesis.patch2 uploaded by
- dbrooks (license 790) pbx.c-funcparenthesis-1.4.diff uploaded by
- loloski (license 68) ........ ................
-
- * Makefile, /: Merged revisions 213904 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r213904 |
- tilghman | 2009-08-24 21:54:07 -0500 (Mon, 24 Aug 2009) | 6 lines
- The DTD should be installed in the same path as the rest of the
- XML documentation. (closes issue #15344) Reported by: tzafrir
- Patches: makefile_appdocs_dtd.diff uploaded by tzafrir (license
- 46) ........
-
- * Makefile, /: Merged revisions 213900 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r213900 | tilghman | 2009-08-24 21:41:17 -0500 (Mon, 24 Aug 2009)
- | 11 lines Merged revisions 213899 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r213899 | tilghman | 2009-08-24 21:40:22 -0500 (Mon, 24 Aug 2009)
- | 4 lines Use the default runlevels for Debian derivatives,
- instead of making up our own. (closes issue #14730) Reported by:
- pkempgen ........ ................
-
-2009-08-24 16:49 +0000 [r213836] Jeff Peeler <jpeeler@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 213833 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r213833 | jpeeler | 2009-08-24 11:43:57 -0500 (Mon, 24 Aug 2009)
- | 14 lines Fix storage of greetings when using IMAP_STORAGE The
- store macro was not getting called preventing storage of IMAP
- greetings at all. This has been corrected along with fixing
- checking if the imapgreetings option is turned on to store the
- greeting in IMAP. Lastly, the attachment filename was incorrectly
- using the full path instead of just the basename, which was
- causing problems with retrieval of the greeting. (closes issue
- #14950) Reported by: noahisaac (closes issue #15729) Reported by:
- lmadsen ........
-
-2009-08-24 04:54 +0000 [r213791] Moises Silva <moises.silva@gmail.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 213790 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r213790 | moy | 2009-08-24 00:46:28 -0400 (Mon, 24 Aug 2009) | 1
- line improve handling of openr2_chan_disconnect_call API failure,
- unlikely, but happened on openr2 library bug ........
-
-2009-08-21 22:54 +0000 [r213739] Tilghman Lesher <tlesher@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 213738 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r213738 |
- tilghman | 2009-08-21 17:36:39 -0500 (Fri, 21 Aug 2009) | 2 lines
- Clarifying comments in sip_register, and removing a dead section
- ........
-
-2009-08-21 22:23 +0000 [r213721] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 213716 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r213716 |
- dvossel | 2009-08-21 17:22:11 -0500 (Fri, 21 Aug 2009) | 10 lines
- Register request line contains wrong address when user domain and
- register host differ (closes issue #15539) Reported by:
- Nick_Lewis Patches: chan_sip.c-registraraddr.patch uploaded by
- Nick (license 657) register_domain_fix_1.6.2 uploaded by dvossel
- (license 671) Tested by: Nick_Lewis, dvossel ........
-
-2009-08-21 21:44 +0000 [r213698] Kevin P. Fleming <kpfleming@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 213697 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r213697 | kpfleming | 2009-08-21 16:39:51 -0500 (Fri, 21 Aug
- 2009) | 12 lines Ensure that realtime mailboxes properly report
- status on subscription. This patch modifies app_voicemail's
- response to mailbox status subscriptions (via the internal event
- system) to ensure that a subscription triggers an explicit poll
- of the mailbox, so the subscriber can get an immediate cached
- event with that status. Previously, the cache was only populated
- with the status of non-realtime mailboxes. (closes issue #15717)
- Reported by: natmlt ........
-
-2009-08-21 21:12 +0000 [r213636] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 213635 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r213635 |
- dvossel | 2009-08-21 16:02:50 -0500 (Fri, 21 Aug 2009) | 5 lines
- fixes sip register parsing when user@domain is used (issue
- #15008) (issue #15672) ........
-
-2009-08-21 16:55 +0000 [r213563] Tilghman Lesher <tlesher@digium.com>
-
- * include/asterisk.h, /: Merged revisions 213560 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r213560 | tilghman | 2009-08-21 11:53:52 -0500 (Fri, 21 Aug 2009)
- | 14 lines Merged revisions 213559 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r213559 | tilghman | 2009-08-21 11:52:53 -0500 (Fri, 21 Aug 2009)
- | 7 lines Permit DEBUG_FD_LEAKS to be used with C++ source files.
- (closes issue #15698) Reported by: slavon Patches:
- 20090817__issue15698.diff.txt uploaded by tilghman (license 14)
- Tested by: slavon, tilghman ........ ................
-
-2009-08-21 16:06 +0000 [r213497] Jason Parker <jparker@digium.com>
-
- * /, configs/queues.conf.sample: Merged revisions 213494 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r213494 | qwell | 2009-08-21 11:04:21 -0500
- (Fri, 21 Aug 2009) | 12 lines Merged revisions 213493 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r213493 | qwell | 2009-08-21 11:03:21 -0500 (Fri, 21 Aug 2009) |
- 5 lines Clarify queues.conf comments to specify that variables
- should be set in the dialplan. (closes issue #15755) Reported by:
- trendboy ........ ................
-
-2009-08-21 04:25 +0000 [r213475-213481] Moises Silva <moises.silva@gmail.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 213454 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r213454 | moy | 2009-08-21 00:09:26 -0400 (Fri, 21 Aug 2009) | 1
- line increment the mfcr2 monitor count when clearing the call
- request ........
-
- * channels/chan_dahdi.c, /: Merged revisions 213216 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r213216 | moy | 2009-08-19 23:26:59 -0400 (Wed, 19 Aug 2009) | 1
- line fixed bug caused by calling ast_request without calling
- ast_call on an R2 channel, ie, CHANISAVAIL ........
-
-2009-08-21 03:53 +0000 [r213453] Terry Wilson <twilson@digium.com>
-
- * main/loader.c, /: Merged revisions 213450 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r213450 |
- twilson | 2009-08-20 22:48:54 -0500 (Thu, 20 Aug 2009) | 2 lines
- Make LOAD_ORDER actually work ........
-
-2009-08-20 21:50 +0000 [r213413] Jeff Peeler <jpeeler@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 213404 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r213404 | jpeeler | 2009-08-20 16:33:11 -0500 (Thu, 20 Aug 2009)
- | 12 lines Fix greeting retrieval from IMAP Properly check for
- the current voicemail state and if it doesn't exist, create it.
- (closes issue #14597) Reported by: wtca Patches: 14597_v2.patch
- uploaded by mmichelson (license 60) Tested by: jpeeler ........
-
-2009-08-20 20:37 +0000 [r213350] Matthew Nicholson <mnicholson@digium.com>
-
- * /, main/features.c: Merged revisions 213327 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r213327 |
- mnicholson | 2009-08-20 15:29:32 -0500 (Thu, 20 Aug 2009) | 7
- lines Fix a crash by checking the proper pointer for validity
- before deferencing it. (closes issue #15751) Reported by: atis
- Patches: ast_bridge_call_peer_cdr.patch uploaded by atis (license
- 242) ........
-
-2009-08-19 22:41 +0000 [r213182] Jason Parker <jparker@digium.com>
-
- * main/alaw.c, main/ulaw.c, /: Merged revisions 213179 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r213179 | qwell | 2009-08-19 17:38:46 -0500 (Wed, 19 Aug 2009) |
- 5 lines Fix compile when certain G711 menuselect options are
- enabled. (closes issue #15697) Reported by: slavon ........
-
-2009-08-19 21:25 +0000 [r213128] David Vossel <dvossel@digium.com>
-
- * apps/app_mixmonitor.c, /: Merged revisions 213113 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r213113 | dvossel | 2009-08-19 16:21:00 -0500
- (Wed, 19 Aug 2009) | 14 lines Merged revisions 213103 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r213103 | dvossel | 2009-08-19 16:18:37 -0500 (Wed, 19 Aug 2009)
- | 8 lines Fixes memory leak caused by incorrectly freeing
- mixmonitor (closes issue #15699) Reported by: edantie Patches:
- mixmonitor.patch uploaded by edantie (license 862) ........
- ................
-
-2009-08-19 21:22 +0000 [r213095-213117] Tilghman Lesher <tlesher@digium.com>
-
- * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions
- 213098 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r213098 |
- tilghman | 2009-08-19 16:05:17 -0500 (Wed, 19 Aug 2009) | 9 lines
- Better parsing for the "register" line Allows characters that are
- otherwise used as delimiters to be used within certain fields
- (like the secret). (closes issue #15008, closes issue #15672)
- Reported by: tilghman Patches: 20090818__issue15008.diff.txt
- uploaded by tilghman (license 14) Tested by: lmadsen, tilghman
- ........
-
- * /, channels/chan_sip.c: Merged revisions 213093 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r213093 |
- tilghman | 2009-08-19 15:29:41 -0500 (Wed, 19 Aug 2009) | 7 lines
- If we have realtime caching enabled, 'sip reload' must purge
- users/peers, even if the config files haven't changed. (closes
- issue #12869) Reported by: bcnit Patches:
- 20090819__issue12869__2.diff.txt uploaded by tilghman (license
- 14) Tested by: lasko ........
-
-2009-08-19 15:35 +0000 [r213047] Russell Bryant <russell@digium.com>
-
- * /, main/features.c: Merged revisions 213046 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r213046 |
- russell | 2009-08-19 10:32:18 -0500 (Wed, 19 Aug 2009) | 4 lines
- Don't blow up on a NULL cdr. Reported in #asterisk-dev. ........
-
-2009-08-18 20:34 +0000 [r212931-212944] Kevin P. Fleming <kpfleming@digium.com>
-
- * /: Merged revisions 212939 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r212939 |
- kpfleming | 2009-08-18 15:33:34 -0500 (Tue, 18 Aug 2009) | 1 line
- Remove some accidentally-committed properties. ........
-
- * sounds/Makefile, doc/tex/asterisk.tex, CREDITS, /,
- UPGRADE-1.4.txt, sounds/sounds.xml, build_tools/prep_tarball:
- Merged revisions 212922 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r212922 |
- kpfleming | 2009-08-18 15:29:37 -0500 (Tue, 18 Aug 2009) | 6
- lines Convert this branch to Opsound music-on-hold. For more
- details:
- http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/
- ........
-
-2009-08-18 19:28 +0000 [r212866] Tilghman Lesher <tlesher@digium.com>
-
- * /, configs/extconfig.conf.sample: Merged revisions 212857 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r212857 | tilghman | 2009-08-18 14:25:09 -0500 (Tue, 18
- Aug 2009) | 4 lines Make the default extconfig.conf match entries
- with the sample res_mysql.conf. This eliminates a future source
- of possible confusion with the configuration of 1.6.1 and higher.
- ........
-
-2009-08-18 16:56 +0000 [r212769] Richard Mudgett <rmudgett@digium.com>
-
- * channels/misdn/isdn_lib.c, /: Merged revisions 212758 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r212758 | rmudgett | 2009-08-18 11:29:47 -0500
- (Tue, 18 Aug 2009) | 9 lines Merged revisions 212727 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r212727 | rmudgett | 2009-08-18 11:00:56 -0500 (Tue, 18
- Aug 2009) | 1 line Removed some deadwood and added some doxygen
- comments. ........ ................
-
-2009-08-18 16:41 +0000 [r212767] Sean Bright <sean@malleable.com>
-
- * main/manager.c, /: Merged revisions 212764 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r212764 | seanbright | 2009-08-18 12:38:36 -0400 (Tue, 18 Aug
- 2009) | 18 lines Merged revisions 212763 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r212763 | seanbright | 2009-08-18 12:36:00 -0400 (Tue, 18 Aug
- 2009) | 11 lines Delay the creation of temporary files until we
- have a valid manager command to handle. Without this patch,
- asterisk creates a temporary file before determining if the
- specified command is valid. If invalid, we weren't properly
- cleaning up the file. (closes issue #15730) Reported by: zmehmood
- Patches: M15730.diff uploaded by junky (license 177) Tested by:
- zmehmood ........ ................
-
-2009-08-17 20:01 +0000 [r212631] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 212627 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r212627 | tilghman | 2009-08-17 14:57:42 -0500 (Mon, 17 Aug 2009)
- | 4 lines Check the return value of opendir(3), or we may crash.
- (closes issue #15720) Reported by: tobias_e ........
-
-2009-08-17 18:56 +0000 [r212580-212584] Sean Bright <sean@malleable.com>
-
- * /, channels/chan_agent.c: Merged revisions 212581 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r212581 | seanbright | 2009-08-17 14:50:24 -0400 (Mon, 17 Aug
- 2009) | 5 lines Correct spelling of AGENTACCEPTDTMF in
- chan_agent. (closes issue #15668) Reported by: davidw ........
-
- * main/logger.c: Merged revisions 212574 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r212574 |
- seanbright | 2009-08-17 14:18:16 -0400 (Mon, 17 Aug 2009) | 8
- lines Correct the return value check for ast_safe_system. The
- logic here was reversed as ast_safe_system returns -1 on error
- and not on success. Fix suggested by reporter. (closes issue
- #15667) Reported by: loic ........
-
-2009-08-17 16:52 +0000 [r212509] Jeff Peeler <jpeeler@digium.com>
-
- * channels/misdn_config.c, /: Merged revisions 212506 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r212506 | jpeeler | 2009-08-17 11:50:45 -0500
- (Mon, 17 Aug 2009) | 19 lines Merged revisions 212498 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r212498 | jpeeler | 2009-08-17 11:34:56 -0500 (Mon, 17 Aug 2009)
- | 12 lines Fix segfault when reloading chan_misdn. If more ports
- were specified than configured in misdn.conf a reload would crash
- asterisk. The problem was the unconfigured port was using data
- from the previously configured port. When the data for an
- unconfigured port was freed a crash would result from the double
- free. (closes issue #12113) Reported by: agupta Patches:
- bug12113.patch uploaded by jpeeler (license 325) ........
- ................
-
-2009-08-17 15:51 +0000 [r212434] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 212431 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r212431 | rmudgett | 2009-08-17 10:42:51 -0500
- (Mon, 17 Aug 2009) | 16 lines Merged revisions 212430 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 Fix
- uninitialized variable causing random MWI indications. (closes
- issue #15727) Reported by: doda Patches: dahdi_changes.patch
- uploaded by doda (license 853) ........ r212430 | rmudgett |
- 2009-08-17 10:36:28 -0500 (Mon, 17 Aug 2009) | 1 line Fix
- uninitialized variable. ........ ................
-
-2009-08-14 17:37 +0000 [r212250] Tilghman Lesher <tlesher@digium.com>
-
- * funcs/func_curl.c, /: Merged revisions 212249 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r212249 |
- tilghman | 2009-08-14 12:36:40 -0500 (Fri, 14 Aug 2009) | 2 lines
- Add SSL_VERIFYPEER, as requested on the -users list ........
-
-2009-08-13 15:47 +0000 [r212116] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 212113 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r212113 |
- kpfleming | 2009-08-13 10:46:25 -0500 (Thu, 13 Aug 2009) | 3
- lines Ensure that T38FaxVersion is put into outgoing SDP in the
- proper case. ........
-
-2009-08-13 13:56 +0000 [r212070] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 212067 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r212067 |
- file | 2009-08-13 10:51:04 -0300 (Thu, 13 Aug 2009) | 2 lines
- Check an actual populated variable when seeing if we need to do
- video or not. ........
-
-2009-08-13 11:47 +0000 [r212030] Gavin Henry <ghenry@suretecsystems.com>
-
- * contrib/scripts/asterisk.ldap-schema,
- contrib/scripts/asterisk.ldif, /: Merged revisions 212027 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r212027 | ghenry | 2009-08-13 12:37:12 +0100 (Thu, 13
- Aug 2009) | 6 lines Fixed typo (closes issue #15710) Reported by:
- suretec ........
-
-2009-08-12 23:16 +0000 [r211951-211959] Matthew Nicholson <mnicholson@digium.com>
-
- * apps/app_queue.c, /: Merged revisions 211957 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r211957 | mnicholson | 2009-08-12 18:14:36 -0500 (Wed, 12 Aug
- 2009) | 17 lines Merged revisions 211953 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r211953 | mnicholson | 2009-08-12 18:04:02 -0500 (Wed, 12 Aug
- 2009) | 10 lines This patch adds additional checking when
- generating queue log TRANSFER events. The additional checks
- prevent generation of false TRANSFER events in certain
- situations. (closes issue #14536) Reported by: aragon Patches:
- queue-log-xfer-fix1.diff uploaded by mnicholson (license 96)
- Tested by: aragon, mnicholson ........ ................
-
- * /, channels/chan_sip.c: Merged revisions 211876 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r211876 |
- mnicholson | 2009-08-12 14:53:14 -0500 (Wed, 12 Aug 2009) | 11
- lines Make asterisk handle 423 Interval Too Short messages
- better. This change uses separate values for the acceptable
- minimum expiry provided by the 423 error and the expiry value
- stored in the configuration file. Previously, the value pulled
- from the configuration file would be overwritten. (closes issue
- #14366) Reported by: Nick_Lewis Patches: sip-expiry-fix1.diff
- uploaded by mnicholson (license 96) chan_sip.c-reqexpiry.patch
- uploaded by Nick (license 657) Tested by: mnicholson ........
-
-2009-08-12 16:21 +0000 [r211785] Gavin Henry <ghenry@suretecsystems.com>
-
- * res/res_config_ldap.c, contrib/scripts/asterisk.ldap-schema,
- contrib/scripts/asterisk.ldif, /: Merged revisions 211767 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r211767 | ghenry | 2009-08-12 17:00:46 +0100 (Wed, 12
- Aug 2009) | 33 lines Added three new attributes and applied a
- patch to res_config_ldap.c attributetype (
- AstAccountSubscribeContext NAME 'AstAccountSubscribeContext' DESC
- 'Asterisk subscribe context' EQUALITY caseIgnoreMatch SUBSTR
- caseIgnoreSubstringsMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.15)
- attributetype ( AstAccountIpAddr NAME 'AstAccountIpAddr' DESC
- 'Asterisk aaccount IP address' EQUALITY caseIgnoreMatch SUBSTR
- caseIgnoreSubstringsMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.15)
- attributetype ( AstAccountUserAgent NAME 'AstAccountUserAgent'
- DESC 'Asterisk account user context' EQUALITY caseIgnoreMatch
- SUBSTR caseIgnoreSubstringsMatch SYNTAX
- 1.3.6.1.4.1.1466.115.121.1.15) and patch
- fix_empty_attributes_1.6.1.4_v2.patch (closes issue #13725)
- Reported by: macogeek Patches:
- fix_empty_attributes_1.6.1.4_v2.patch uploaded by xvisor (license
- 863) Tested by: suretec ........
-
-2009-08-10 19:51 +0000 [r211580-211585] 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
- ........ ................
-
- * apps/app_queue.c, apps/app_talkdetect.c, agi/eagi-sphinx-test.c,
- res/res_config_curl.c, channels/chan_usbradio.c,
- channels/chan_misdn.c, res/snmp/agent.c, apps/app_sms.c,
- apps/app_verbose.c, apps/app_stack.c, apps/app_mixmonitor.c,
- main/asterisk.c, main/dsp.c, main/timing.c,
- doc/CODING-GUIDELINES, funcs/func_speex.c, main/frame.c,
- utils/muted.c, apps/app_meetme.c, apps/app_alarmreceiver.c,
- cdr/cdr_pgsql.c, res/res_musiconhold.c, channels/chan_iax2.c,
- apps/app_followme.c, main/enum.c, main/indications.c,
- res/res_config_sqlite.c, channels/misdn_config.c, utils/frame.c,
- main/cli.c, pbx/pbx_loopback.c, channels/chan_phone.c,
- funcs/func_enum.c, res/res_smdi.c, channels/chan_skinny.c,
- funcs/func_odbc.c, apps/app_minivm.c, res/res_agi.c,
- res/res_config_ldap.c, apps/app_adsiprog.c,
- funcs/func_dialplan.c, main/pbx.c, main/dnsmgr.c,
- funcs/func_sprintf.c, funcs/func_timeout.c, channels/chan_sip.c,
- apps/app_privacy.c, res/res_limit.c, apps/app_waitforsilence.c,
- codecs/codec_speex.c, agi/eagi-test.c, apps/app_morsecode.c,
- funcs/func_cut.c, channels/chan_oss.c, main/netsock.c,
- apps/app_waitforring.c, funcs/func_channel.c, apps/app_macro.c,
- pbx/pbx_dundi.c, utils/extconf.c, pbx/pbx_config.c,
- apps/app_chanspy.c, res/res_odbc.c, apps/app_voicemail.c,
- apps/app_dahdibarge.c, funcs/func_rand.c, apps/app_readfile.c, /,
- apps/app_record.c, main/utils.c, cdr/cdr_adaptive_odbc.c,
- res/res_http_post.c, main/config.c, res/ael/pval.c, main/cdr.c,
- main/channel.c, channels/chan_dahdi.c, pbx/pbx_spool.c,
- main/manager.c, apps/app_setcallerid.c, apps/app_osplookup.c,
- main/features.c, main/http.c, channels/xpmr/xpmr.c,
- apps/app_rpt.c, channels/chan_mgcp.c, res/res_config_pgsql.c,
- channels/chan_agent.c, funcs/func_math.c, apps/app_waituntil.c,
- apps/app_disa.c, main/acl.c, apps/app_originate.c,
- channels/iax2-provision.c: AST-2009-005
-
-2009-08-10 14:15 +0000 [r211350] 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 [r211235-211278] 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 20:18 +0000 [r211122] 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 18:20 +0000 [r211051] 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 13:10 +0000 [r210995] 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 21:47 +0000 [r210911-210917] 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 17:49 +0000 [r210820] 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 20:43 +0000 [r210686] 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 18:56 +0000 [r210565-210566] Leif Madsen <lmadsen@digium.com>
-
- * /: 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 ........ ................
-
- * 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 14:55 +0000 [r210191-210241] 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 ........ ................
-
- * configs/sip.conf.sample, configs/skinny.conf.sample, main/rtp.c,
- channels/chan_mgcp.c, doc/chan_sip-perf-testing.txt,
- contrib/scripts/realtime_pgsql.sql, /, channels/chan_sip.c,
- channels/chan_skinny.c, configs/mgcp.conf.sample,
- doc/res_config_sqlite.txt, doc/tex/phoneprov.tex, UPGRADE.txt,
- configs/res_ldap.conf.sample: Merged revisions 210190 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r210190 | kpfleming | 2009-08-03 15:48:48 -0500 (Mon, 03
- Aug 2009) | 11 lines Rename 'canreinvite' option to
- 'directmedia', with backwards compatibility. It is clear from
- multiple mailing list, forum, wiki and other sorts of posts that
- users don't really understand the effects that the 'canreinvite'
- config option actually has, and that in some cases they think
- that setting it to 'no' will actually cause various other
- features (T.38, MOH, etc.) to not work properly, when in fact
- this is not the case. This patch changes the proper name of the
- option to what it should have been from the beginning
- ('directmedia'), but preserves backwards compatibility for
- existing configurations. ........
-
-2009-08-01 11:33 +0000 [r209837-209906] 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 ........
- ................
-
- * /, main/event.c: Merged revisions 209835 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r209835 |
- russell | 2009-08-01 05:43:40 -0500 (Sat, 01 Aug 2009) | 6 lines
- Fix ast_event_queue_and_cache() to actually do the cache() part.
- (closes issue #15624) Reported by: ffossard Tested by: russell
- ........
-
-2009-08-01 01:34 +0000 [r209816] Kevin P. Fleming <kpfleming@digium.com>
-
- * pbx/pbx_config.c, channels/misdn/isdn_lib.c, utils/frame.c,
- main/pbx.c, /, main/Makefile, channels/misdn/ie.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 22:01 +0000 [r209715] 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 18:51 +0000 [r209594] David Brooks <dbrooks@digium.com>
-
- * channels/chan_console.c, include/asterisk/abstract_jb.h,
- apps/app_forkcdr.c, channels/chan_dahdi.c,
- contrib/init.d/rc.debian.asterisk, /, apps/app_sms.c,
- codecs/lpc10/pitsyn.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 ........
-
-2009-07-30 14:40 +0000 [r209518] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 209516 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r209516 |
- mmichelson | 2009-07-30 09:38:21 -0500 (Thu, 30 Jul 2009) | 8
- lines Fix a crash that can result if text codecs are allowed but
- textsupport is disabled. (closes issue #15596) Reported by:
- fabled Patches: sip-red.patch uploaded by fabled (license 448)
- ........
-
-2009-07-28 Leif Madsen <lmadsen@digium.com>
-
- * Release Asterisk 1.6.2.0-beta4
-
-2009-07-28 00:19 +0000 [r209328] 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 [r209265-209282] 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 20:58 +0000 [r209238] Mark Michelson <mmichelson@digium.com>
-
- * main/rtp.c, /: Merged revisions 209235 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r209235 |
- mmichelson | 2009-07-27 15:54:54 -0500 (Mon, 27 Jul 2009) | 5
- lines Gracefully handle malformed RTP text packets. AST-2009-004
- ........
-
-2009-07-27 20:33 +0000 [r209234] David Brooks <dbrooks@digium.com>
-
- * res/res_jabber.c, main/loader.c, channels/chan_dahdi.c,
- channels/chan_vpb.cc, res/res_smdi.c, /,
- include/asterisk/module.h, main/features.c, res/res_agi.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 20:23 +0000 [r209135-209222] 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 16:07 +0000 [r209063] David Brooks <dbrooks@digium.com>
-
- * apps/app_rpt.c, 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 15:40 +0000 [r209059] 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 01:23 +0000 [r208927] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_iax2.c, /, main/translate.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 ........ ................
-
-2009-07-26 14:07 +0000 [r208889] Michiel van Baak <michiel@vanbaak.info>
-
- * contrib/scripts/install_prereq, /: Merged revisions 208886 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r208886 | mvanbaak | 2009-07-26 16:00:52 +0200 (Sun, 26
- Jul 2009) | 2 lines add OpenBSD to the install_prereq script
- ........
-
-2009-07-25 12:31 +0000 [r208816-208853] Michiel van Baak <michiel@vanbaak.info>
-
- * contrib/scripts/install_prereq, /: Merged revisions 208848 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r208848 | mvanbaak | 2009-07-25 14:28:38 +0200 (Sat, 25
- Jul 2009) | 2 lines libxml2-dev is needed as well by default.
- ........
-
- * main/cli.c, /, configs/cli_aliases.conf.sample: Merged revisions
- 208813 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r208813 |
- mvanbaak | 2009-07-25 14:03:25 +0200 (Sat, 25 Jul 2009) | 10
- lines add default alias reload to run module reload. Requiring
- 'module reload' to reload everything, including core etc makes
- russell very unhappy. The default configuration already loads the
- 'friendly' aliases template. Added 'reload=module reload' to that
- template. Also removed the comment in main/cli.c that reload
- should come back. ........
-
-2009-07-25 06:26 +0000 [r208755] Jeff Peeler <jpeeler@digium.com>
-
- * channels/chan_iax2.c, /, channels/chan_skinny.c,
- main/translate.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 21:13 +0000 [r208695-208710] Russell Bryant <russell@digium.com>
-
- * /, pbx/pbx_dundi.c: Merged revisions 208709 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r208709 |
- russell | 2009-07-24 16:12:43 -0500 (Fri, 24 Jul 2009) | 2 lines
- Remove trailing whitespace. ........
-
- * main/cli.c, /: Merged revisions 208706 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r208706 |
- russell | 2009-07-24 15:54:37 -0500 (Fri, 24 Jul 2009) | 6 lines
- Note that "reload" needs to be added back. I keep getting annoyed
- at having to type "module reload" to reload everything, so I'm
- adding a note that we need to add "reload" back. "module reload"
- doesn't really make sense as the command to reload everything,
- including the core. ........
-
- * main/cli.c, /: Merged revisions 208693 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r208693 |
- russell | 2009-07-24 15:25:23 -0500 (Fri, 24 Jul 2009) | 2 lines
- Don't log a warning for something that does not affect operation.
- ........
-
-2009-07-24 19:42 +0000 [r208664] Mark Michelson <mmichelson@digium.com>
-
- * /: Fixing trunk-blocked property.
-
-2009-07-24 18:56 +0000 [r208596] 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 18:32 +0000 [r208591] 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 15:06 +0000 [r208551] 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. ........
-
-2009-07-24 14:39 +0000 [r208545] Michiel van Baak <michiel@vanbaak.info>
-
- * contrib/scripts/install_prereq, /: Merged revisions 208542 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r208542 | mvanbaak | 2009-07-24 16:35:49 +0200 (Fri, 24
- Jul 2009) | 13 lines use aptitude for debian based systems The
- function to check wether we need to install packages was using
- dpkg-query which was gives wrong output on Debian 5 Also, the
- apt-get has been replaced with aptitude because aptitude is now
- the preferred way to handle packages on Debian (closes issue
- #15570) Reported by: mvanbaak Patches:
- 2009072400_installprereq-aptitude.diff uploaded by mvanbaak
- (license 7) ........
-
-2009-07-23 22:31 +0000 [r208501] Kevin P. Fleming <kpfleming@digium.com>
-
- * include/asterisk/frame.h, 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: 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 19:36 +0000 [r208391] 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 19:25 +0000 [r208387] 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 [r208266-208320] 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 ........ ................
-
-2009-07-22 21:46 +0000 [r208116] Jason Parker <jparker@digium.com>
-
- * /, apps/app_festival.c: Merged revisions 208113 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r208113 |
- qwell | 2009-07-22 16:43:57 -0500 (Wed, 22 Jul 2009) | 9 lines
- Restore an int declaration on PPC platforms. This x is one crafty
- little bugger... It was used for 2 different things (one of which
- was only done on PPC) in 1.4. One of the uses were removed in
- trunk, and with it went the declaration. (closes issue #14038)
- Reported by: ffloimair ........
-
-2009-07-22 16:52 +0000 [r207949-208053] Tilghman Lesher <tlesher@digium.com>
-
- * /, res/res_realtime.c: Merged revisions 208052 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r208052 |
- tilghman | 2009-07-22 11:49:42 -0500 (Wed, 22 Jul 2009) | 7 lines
- Clarify documentation on 'realtime update2' to show more than one
- condition. (closes issue #15357) Reported by: snuffy Patches:
- bug_fix_doc_update2.diff uploaded by snuffy (license 35)
- (slightly modified by me) ........
-
- * /, 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 22:23 +0000 [r207930] Russell Bryant <russell@digium.com>
-
- * doc/CODING-GUIDELINES, /: Merged revisions 207925 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r207925 | russell | 2009-07-21 17:22:18 -0500 (Tue, 21 Jul 2009)
- | 4 lines Note that we use tabs instead of spaces for
- indentation. I'm surprised this was never actually in here...
- ........
-
-2009-07-21 20:30 +0000 [r207785-207862] 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 r207638, this approach could
- potentially block for an unacceptable amount of time.
-
-2009-07-21 14:32 +0000 [r207727] 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 13:56 +0000 [r207685] Kevin P. Fleming <kpfleming@digium.com>
-
- * channels/Makefile, doc/video_console.txt, Makefile, agi/Makefile,
- codecs/Makefile, utils/Makefile, funcs/Makefile,
- codecs/lpc10/Makefile, main/db1-ast/Makefile, /, main/Makefile,
- codecs/gsm/Makefile, Makefile.moddir_rules, Makefile.rules,
- pbx/Makefile, res/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 04:51 +0000 [r207638] 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 22:14 +0000 [r207523] 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 16:41 +0000 [r207364] 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 04:19 +0000 [r207327] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 207317 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r207317 | tilghman | 2009-07-17 23:16:44 -0500 (Fri, 17 Jul 2009)
- | 3 lines Flag field in wrong position. Reported by "Hoggins!" on
- asterisk-dev list. ........
-
-2009-07-18 03:50 +0000 [r207315] Richard Mudgett <rmudgett@digium.com>
-
- * channels/misdn/isdn_lib.c, channels/chan_misdn.c: 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 22:31 +0000 [r207226-207257] Tilghman Lesher <tlesher@digium.com>
-
- * /, doc/voicemail_odbc_postgresql.txt: Merged revisions 207255 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r207255 | tilghman | 2009-07-17 17:29:50 -0500 (Fri, 17
- Jul 2009) | 2 lines Add flag here, too (as requested by jsmith)
- ........
-
- * /, doc/tex/odbcstorage.tex, UPGRADE.txt: Merged revisions 207224
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r207224 | tilghman | 2009-07-17 17:04:43 -0500 (Fri, 17
- Jul 2009) | 2 lines Document the "flag" field in the
- voicemessages table. ........
-
-2009-07-17 19:40 +0000 [r207104-207159] 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 17:52 +0000 [r206869-207030] 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 ........ ................
-
- * /, funcs/func_timeout.c: Merged revisions 206877 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r206877 | dvossel | 2009-07-16 16:45:14 -0500 (Thu, 16 Jul 2009)
- | 6 lines TIMEOUT(absolute) returned negative value. (closes
- issue #15513) Reported by: ys ........
-
- * 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 16:53 +0000 [r206811] 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 22:04 +0000 [r206770] 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 21:50 +0000 [r206765] 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 20:20 +0000 [r206703] 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 16:04 +0000 [r206639] 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 20:26 +0000 [r206598] Tilghman Lesher <tlesher@digium.com>
-
- * /, apps/app_meetme.c, contrib/scripts/meetme.sql: Merged
- 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 19:49 +0000 [r206565] 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 14:59 +0000 [r206389] 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 01:59 +0000 [r206373] 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
- ........ ................
-
-2009-07-13 23:27 +0000 [r206281] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 206280 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r206280 |
- dvossel | 2009-07-13 18:26:51 -0500 (Mon, 13 Jul 2009) | 9 lines
- dns lookup of peername rather than peer's host in
- transmit_register() (closes issue #15052) Reported by: fsantulli
- Patches: chan_sip_bug_15052_[20090626204511].patch uploaded by
- fsantulli (license 818) Tested by: fsantulli ........
-
-2009-07-13 16:24 +0000 [r206187] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 206185 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r206185 | tilghman | 2009-07-13 11:23:07 -0500 (Mon, 13 Jul 2009)
- | 2 lines Remove reference to non-existent help file ........
-
-2009-07-10 21:46 +0000 [r205986] 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 18:45 +0000 [r205942] 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 17:54 +0000 [r205882] Mark Michelson <mmichelson@digium.com>
-
- * /, 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 16:47 +0000 [r205841] 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 16:01 +0000 [r205781] Kevin P. Fleming <kpfleming@digium.com>
-
- * /, apps/app_fax.c: Merged revisions 205780 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r205780 |
- kpfleming | 2009-07-10 11:00:44 -0500 (Fri, 10 Jul 2009) | 11
- lines Eliminate extraneous LOG_DEBUG messages generated by
- app_fax. The transmit_audio() and transmit_t38() functions in
- app_fax have processing loops that are supposed to wait for
- frames to arrive on the channel and then handle them, but they
- also have short timeouts so that the loops can have watchdog
- timers and do other required processing. This commit changes the
- loops to not actually call ast_read() and attempt to process the
- returned frame unless a frame actually arrived, eliminating
- hundreds of LOG_DEBUG messages and slightly improving
- performance. ........
-
-2009-07-10 15:58 +0000 [r205779] 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 15:36 +0000 [r205773] 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 23:56 +0000 [r205731] 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 21:27 +0000 [r205699] Kevin P. Fleming <kpfleming@digium.com>
-
- * include/asterisk/frame.h, /, channels/chan_sip.c, apps/app_fax.c:
- 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 16:19 +0000 [r205595-205603] 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. ........ ................
-
- * channels/chan_iax2.c, include/asterisk/frame.h, main/rtp.c, /:
- 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 08:34 +0000 [r205535] 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 22:15 +0000 [r205411-205413] David Vossel <dvossel@digium.com>
-
- * include/asterisk/pbx.h, include/asterisk/devicestate.h,
- main/pbx.c, /, main/devicestate.c: 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. ........
- ................
-
- * /, main/devicestate.c: Merged revisions 205410 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r205410 |
- dvossel | 2009-07-08 17:02:54 -0500 (Wed, 08 Jul 2009) | 3 lines
- missing comma in devstatestring array ........
-
-2009-07-08 19:28 +0000 [r205353] 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 18:22 +0000 [r205302] 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 18:18 +0000 [r205287] David Brooks <dbrooks@digium.com>
-
- * /, main/features.c: Merged revisions 205254 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r205254 |
- dbrooks | 2009-07-08 12:26:26 -0500 (Wed, 08 Jul 2009) | 8 lines
- Fixes Park() argument handling Park() was not respecting the
- arguments passed to it. Any extension/context/priority given to
- it was being ignored. This patch remedies this. (closes issue
- #15380) Reported by: DLNoah ........
-
-2009-07-08 17:00 +0000 [r205223] Tilghman Lesher <tlesher@digium.com>
-
- * main/say.c: oops, fixing build
-
-2009-07-08 16:55 +0000 [r205217] 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 16:30 +0000 [r205207] 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 15:57 +0000 [r205148-205154] 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. ........
-
- * include/asterisk/_private.h, res/res_jabber.c, main/asterisk.c,
- /, main/Makefile, res/res_crypto.c, main/ssl.c (added): 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. ........
-
-2009-07-06 13:41 +0000 [r204951] 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 22:06 +0000 [r204838] 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 16:12 +0000 [r204711] 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/ ........ ................
-
-2009-06-30 21:30 +0000 [r204611] 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 18:55 +0000 [r204478] 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 [r204473] 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 ........
- ................
-
-2009-06-30 17:22 +0000 [r204442] Russell Bryant <russell@digium.com>
-
- * configs/res_config_sqlite.conf (removed),
- configs/res_config_sqlite.conf.sample (added), /: Merged
- revisions 204440 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r204440 |
- russell | 2009-06-30 12:22:16 -0500 (Tue, 30 Jun 2009) | 2 lines
- Rename res_config_sqlite.conf to res_config_sqlite.conf.sample
- (missing .sample). ........
-
-2009-06-29 22:53 +0000 [r204250-204304] 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 09:55 +0000 [r203961] Russell Bryant <russell@digium.com>
-
- * CHANGES, /: Merged revisions 203960 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r203960 |
- russell | 2009-06-27 04:51:45 -0500 (Sat, 27 Jun 2009) | 2 lines
- Minor tweaks and spelling fixes for CHANGES and UPGRADE.txt.
- ........
-
-2009-06-27 01:24 +0000 [r203941] 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 22:14 +0000 [r203857] 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 21:27 +0000 [r203782-203828] 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 20:18 +0000 [r203731] 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 19:49 +0000 [r203715] Russell Bryant <russell@digium.com>
-
- * include/asterisk/devicestate.h, main/pbx.c, /,
- main/devicestate.c: Merged revisions 203702 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r203702 |
- russell | 2009-06-26 14:31:14 -0500 (Fri, 26 Jun 2009) | 5 lines
- Make invalid hints report Unavailable instead of Idle. (closes
- issue #14413) Reported by: pj ........
-
-2009-06-26 19:48 +0000 [r203712] 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 19:42 +0000 [r203709] 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 ........
-
-2009-06-26 19:38 +0000 [r203705] Joshua Colp <jcolp@digium.com>
-
- * configs/sip.conf.sample, include/asterisk/frame.h, main/rtp.c,
- main/channel.c, main/frame.c, /, channels/chan_sip.c,
- apps/app_fax.c: 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. ........
-
-2009-06-25 21:46 +0000 [r203445] 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 21:21 +0000 [r203400] 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 21:08 +0000 [r203379] 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 ........ ................
-
-2009-06-25 19:27 +0000 [r203276] Jason Parker <jparker@digium.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 203258 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r203258 | qwell | 2009-06-25 14:22:46 -0500 (Thu, 25 Jun 2009) |
- 10 lines Unmute when we get a dtmfup (we muted on dtmfdown)
- event. This would occasionally cause one-way audio when using
- hardware DTMF detection. (closes issue #14761) Reported by:
- tzafrir Patches: v1-14761.patch uploaded by dimas (license 88)
- Tested by: tzafrir, dimas ........
-
-2009-06-25 16:08 +0000 [r203119] Russell Bryant <russell@digium.com>
-
- * /, 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 21:27 +0000 [r203077] 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 18:30 +0000 [r202970] 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 18:11 +0000 [r202928] 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 23:58 +0000 [r202842] Sean Bright <sean@malleable.com>
-
- * doc/tex/cdrdriver.tex, /, doc/tex/billing.tex: Merged revisions
- 202840-202841 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r202840 |
- seanbright | 2009-06-23 19:53:45 -0400 (Tue, 23 Jun 2009) | 1
- line Remove some trailing whitespace before making content
- changes. ........ r202841 | seanbright | 2009-06-23 19:57:07
- -0400 (Tue, 23 Jun 2009) | 1 line Change some section names in
- the CDR tex documentation. ........
-
-2009-06-23 22:47 +0000 [r202805] Russell Bryant <russell@digium.com>
-
- * doc/tex/cdrdriver.tex, /: Merged revisions 202804 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r202804 | russell | 2009-06-23 17:47:26 -0500 (Tue, 23 Jun 2009)
- | 2 lines Clean up section hierarchy for the CDR chapter.
- ........
-
-2009-06-23 22:12 +0000 [r202765] 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 16:33 +0000 [r202673] 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 20:19 +0000 [r202495-202511] 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 ........ ................
-
- * /, 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 16:31 +0000 [r202473] 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 15:37 +0000 [r202411] David Vossel <dvossel@digium.com>
-
- * main/loader.c, /, include/asterisk/module.h: Merged revisions
- 202410 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r202410 |
- dvossel | 2009-06-22 10:33:35 -0500 (Mon, 22 Jun 2009) | 5 lines
- attempting to load running modules Modules placed in the priority
- heap for loading were not properly removed from the linked list.
- This resulted in some modules attempting to load twice. ........
-
-2009-06-22 15:17 +0000 [r202340-202346] 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 16:16 +0000 [r202261-202265] 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 [r202186] 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 21:08 +0000 [r202007] 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 20:26 +0000 [r201995] 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 15:49 +0000 [r201785-201906] Tilghman Lesher <tlesher@digium.com>
-
- * res/res_config_odbc.c, /: Merged revisions 201904 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r201904 | tilghman | 2009-06-19 10:47:55 -0500 (Fri, 19 Jun 2009)
- | 4 lines Fix 2 typos and add support for wide character types.
- Reported by Benny Amorsen via the asterisk-users mailing list.
- http://lists.digium.com/pipermail/asterisk-users/2009-June/233622.html
- ........
-
- * /, 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 16:44 +0000 [r201679] David Vossel <dvossel@digium.com>
-
- * channels/misdn/isdn_lib.c, utils/conf2ael.c, main/ast_expr2.c,
- utils/stereorize.c, main/ast_expr2f.c, res/ael/ael_lex.c,
- utils/ael_main.c, utils/extconf.c, channels/xpmr/xpmr.c,
- pbx/pbx_config.c, res/res_config_ldap.c, apps/app_rpt.c,
- main/asterisk.c, codecs/gsm/src/gsm_destroy.c, /,
- channels/h323/ast_h323.cxx: 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 15:40 +0000 [r201614] 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)
- ........ ................
-
-2009-06-18 15:23 +0000 [r201595] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 201570 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r201570 |
- dvossel | 2009-06-18 10:16:05 -0500 (Thu, 18 Jun 2009) | 11 lines
- parsing extension correctly from sip register lines If a
- transport type was specified, but no extension, parsing of the
- extension would return whatever was after the transport rather
- than defaulting to 's'. (closes issue #15111) Reported by: ffs
- Patches: chan_sip.c_register-parser.patch uploaded by ffs
- (license 730) Tested by: ffs, dvossel ........
-
-2009-06-17 21:33 +0000 [r201533] Tilghman Lesher <tlesher@digium.com>
-
- * apps/app_voicemail.c, /: Merged revisions 201531 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r201531 | tilghman | 2009-06-17 16:31:39 -0500 (Wed, 17 Jun 2009)
- | 7 lines Initialize additional variables, to prevent a possible
- crash. (closes issue #15186) Reported by: ajohnson Patches:
- 20090528__issue15186.diff.txt uploaded by tilghman (license 14)
- Tested by: ajohnson ........
-
-2009-06-17 20:12 +0000 [r201461-201465] 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 20:01 +0000 [r201447-201454] David Vossel <dvossel@digium.com>
-
- * doc/datastores.txt, /: Merged revisions 201453 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r201453 |
- dvossel | 2009-06-17 15:00:51 -0500 (Wed, 17 Jun 2009) | 3 lines
- ast_channel_datastore_alloc is no longer used. updating
- datastores.txt to reflect that. ........
-
- * 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 19:49 +0000 [r201446] 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 15:25 +0000 [r201360] David Vossel <dvossel@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 201344 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r201344 |
- dvossel | 2009-06-17 10:20:26 -0500 (Wed, 17 Jun 2009) | 16 lines
- SIP registry ref count error During a sip reload, the list of
- sip_registry objects are supposed to be traversed, unlinked, and
- destroyed, but destruction never takes place due to a ref
- counting error. This causes a memory leak when registry items are
- removed from sip.conf and reloaded. While the registries are
- removed from the global list, they are not removed from the
- scheduler. Because of this, SIP register attempts continue to be
- sent out for the item even though it may no longer be in the
- .conf. (closes issue #15295) Reported by: amorsen Review:
- https://reviewboard.asterisk.org/r/282/ ........
-
-2009-06-17 12:06 +0000 [r201265] 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 22:30 +0000 [r201224] 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 19:47 +0000 [r200990-201097] Kevin P. Fleming <kpfleming@digium.com>
-
- * include/asterisk/frame.h, apps/app_chanspy.c,
- apps/app_mixmonitor.c, main/channel.c, main/autoservice.c,
- main/frame.c, /, apps/app_meetme.c, main/slinfactory.c,
- include/asterisk/linkedlists.h, main/file.c,
- include/asterisk/channel.h: Merged revisions 201056 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/ ........
- ................
-
- * /, configure, autoconf/ast_gcc_attribute.m4, configure.ac: Merged
- revisions 201090 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ 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. ........
-
- * /, 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 16:28 +0000 [r200984] 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 16:05 +0000 [r200948] 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 12:55 +0000 [r200842] Eliel C. Sardanons <eliels@gmail.com>
-
- * res/res_smdi.c, /: Merged revisions 200841 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r200841 |
- eliel | 2009-06-16 08:32:00 -0400 (Tue, 16 Jun 2009) | 6 lines
- Show the interface name on error, if it is not found. If the
- smdiport specified is not found, show the interface name instead
- of '(null)'. ........
-
-2009-06-16 02:41 +0000 [r200807] Moises Silva <moises.silva@gmail.com>
-
- * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged
- revisions 200799 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r200799 |
- moy | 2009-06-15 21:24:30 -0500 (Mon, 15 Jun 2009) | 2 lines keep
- backwards compatible chan_dahdi with older openr2 versions by not
- using the new skip category feature unless supported ........
-
-2009-06-16 01:30 +0000 [r200690-200765] 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: Merged revisions 200689 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r200689 |
- kpfleming | 2009-06-15 15:42:38 -0500 (Mon, 15 Jun 2009) | 11
- 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 15:23 +0000 [r200517] 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 ........ ................
-
-2009-06-14 06:33 +0000 [r200512] Moises Silva <moises.silva@gmail.com>
-
- * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /,
- build_tools/menuselect-deps.in: Merged revisions 200477 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r200477 | moy | 2009-06-14 01:13:48 -0500 (Sun, 14 Jun
- 2009) | 3 lines added openr2 to menuselect-deps.in, recent commit
- in menuselect made me realize this was never done but was working
- anyways also added support for skip category request feature of
- openr2 and updated chan_dahdi.conf.sample ........
-
-2009-06-12 19:08 +0000 [r200364] 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 ........ ................
-
-2009-06-12 02:20 +0000 [r200198-200255] Sean Bright <sean@malleable.com>
-
- * contrib/init.d/rc.debian.asterisk, /: Merged revisions 200254 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r200254 | seanbright | 2009-06-11 22:20:19 -0400 (Thu,
- 11 Jun 2009) | 5 lines Call chgrp instead of chown when setting
- run directory group ownership. (issue #13153) Reported by:
- pabelanger ........
-
- * 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#. ........
-
- * /: Undo block of revision 199782 (will be merging it momentarily)
-
-2009-06-11 21:35 +0000 [r200172] Terry Wilson <twilson@digium.com>
-
- * main/rtp.c: Don't access rtp->rtcp->* if rtp->rtcp is null
-
-2009-06-11 21:18 +0000 [r200154] 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 Leif Madsen <lmadsen@digium.com>
-
- * Release Asterisk 1.6.2.0-beta3
-
-2009-06-11 12:19 +0000 [r200051] Leif Madsen <lmadsen@digium.com>
-
- * build_tools/make_version_h, /, build_tools/make_version_c: 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 20:37 +0000 [r199998] 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 20:18 +0000 [r199966] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 199958 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r199958 |
- mmichelson | 2009-06-10 15:15:48 -0500 (Wed, 10 Jun 2009) | 6
- lines Only try to use the invite_branch on outgoing INVITEs with
- auth credentials. I have added a comment to the code to help ease
- understanding of the logic here as well. ........
-
-2009-06-10 16:13 +0000 [r199860] Sean Bright <sean.bright@gmail.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 20:48 +0000 [r199744-199819] 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 ........
-
- * main/loader.c, /, res/res_timing_pthread.c,
- include/asterisk/module.h, res/res_timing_dahdi.c,
- res/res_timing_timerfd.c: Merged revisions 199743 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r199743 | dvossel | 2009-06-09 11:22:04 -0500 (Tue, 09 Jun 2009)
- | 11 lines module load priority This patch adds the option to
- give a module a load priority. The value represents the order in
- which a module's load() function is initialized. The lower the
- value, the higher the priority. The value is only checked if the
- AST_MODFLAG_LOAD_ORDER flag is set. If the AST_MODFLAG_LOAD_ORDER
- flag is not set, the value will never be read and the module will
- be given the lowest possible priority on load. Since some modules
- are reliant on a timing interface, the timing modules have been
- given a high load priorty. (closes issue #15191) Reported by:
- alecdavis Tested by: dvossel Review:
- https://reviewboard.asterisk.org/r/262/ ........
-
-2009-06-08 19:39 +0000 [r199634] Sean Bright <sean.bright@gmail.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. ........ ................
-
-2009-06-08 17:42 +0000 [r199591] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Recorded merge of revisions 199588 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r199588 | mmichelson | 2009-06-08 12:32:04 -0500 (Mon,
- 08 Jun 2009) | 9 lines Fix a deadlock that could occur when
- setting rtp stats on SIP calls. (closes issue #15143) Reported
- by: cristiandimache Patches: 15143.patch uploaded by mmichelson
- (license 60) Tested by: cristiandimache ........
-
-2009-06-06 21:39 +0000 [r199369] Russell Bryant <russell@digium.com>
-
- * Makefile, /: Merged revisions 199368 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r199368 |
- russell | 2009-06-06 16:38:54 -0500 (Sat, 06 Jun 2009) | 2 lines
- Switch from "echo -n" to printf. On my mac, the -n was just
- getting printed out. ........
-
-2009-06-05 21:25 +0000 [r199299] David Vossel <dvossel@digium.com>
-
- * include/asterisk/devicestate.h, /, main/devicestate.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 13:52 +0000 [r199230] 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 19:15 +0000 [r199140] 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 ........
- ................
-
-2009-06-04 14:53 +0000 [r199054] Sean Bright <sean.bright@gmail.com>
-
- * include/asterisk/_private.h, main/asterisk.c, main/loader.c, /:
- Merged revisions 199051 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r199051 | seanbright | 2009-06-04 10:31:24 -0400 (Thu, 04 Jun
- 2009) | 47 lines Merged revisions 199022 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r199022 | seanbright | 2009-06-04 10:14:57 -0400 (Thu, 04 Jun
- 2009) | 40 lines Safely handle AMI connections/reload requests
- that occur during startup. During asterisk startup, a lock on the
- list of modules is obtained by the primary thread while each
- module is initialized. Issue 13778 pointed out a problem with
- this approach, however. Because the AMI is loaded before other
- modules, it is possible for a module reload to be issued by a
- connected client (via Action: Command), causing a deadlock. The
- resolution for 13778 was to move initialization of the manager to
- happen after the other modules had already been lodaded. While
- this fixed this particular issue, it caused a problem for users
- (like FreePBX) who call AMI scripts via an #exec in a
- configuration file (See issue 15189). The solution I have come up
- with is to defer any reload requests that come in until after the
- server is fully booted. When a call comes in to ast_module_reload
- (from wherever) before we are fully booted, the request is added
- to a queue of pending requests. Once we are done booting up, we
- then execute these deferred requests in turn. Note that I have
- tried to make this a bit more intelligent in that it will not
- queue up more than 1 request for the same module to be reloaded,
- and if a general reload request comes in ('module reload') the
- queue is flushed and we only issue a single deferred reload for
- the entire system. As for how this will impact existing
- installations - Before 13778, a reload issued before module
- initialization was completed would result in a deadlock. After
- 13778, you simply couldn't connect to the manager during startup
- (which causes problems with #exec-that-calls-AMI configuration
- files). I believe this is a good general purpose solution that
- won't negatively impact existing installations. (closes issue
- #15189) (closes issue #13778) Reported by: p_lindheimer Patches:
- 06032009_15189_deferred_reloads.diff uploaded by seanbright
- (license 71) Tested by: p_lindheimer, seanbright Review:
- https://reviewboard.asterisk.org/r/272/ ........ ................
-
-2009-06-03 15:24 +0000 [r198827-198886] David Vossel <dvossel@digium.com>
-
- * main/channel.c, /, main/features.c, include/asterisk/channel.h:
- Merged revisions 198856 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r198856 |
- dvossel | 2009-06-02 16:17:49 -0500 (Tue, 02 Jun 2009) | 10 lines
- Generic call forward api, ast_call_forward() The function
- ast_call_forward() forwards a call to an extension specified in
- an ast_channel's call_forward string. After an ast_channel is
- called, if the channel's call_forward string is set this function
- can be used to forward the call to a new channel and terminate
- the original one. I have included this api call in both
- channel.c's ast_request_and_dial() and feature.c's
- feature_request_and_dial(). App_dial and app_queue already
- contain call forward logic specific for their application and
- options. (closes issue #13630) Reported by: festr Review:
- https://reviewboard.asterisk.org/r/271/ ........
-
- * channels/chan_iax2.c, /: Merged revisions 198824 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r198824 | dvossel | 2009-06-02 12:55:35 -0500 (Tue, 02 Jun 2009)
- | 8 lines fixes issue with channels not going down after transfer
- Iax2 currently does not support native bridging if the timeoutms
- value is set. We check for that in iax2_bridge, but then set
- timeoutms to 0 by default. If the timeoutms is not provided it is
- set to -1. By setting timeoutms to 0 it is processed causing a
- bridging retry loop. (closes issue #15216) Reported by: oxymoron
- Tested by: dvossel ........
-
-2009-06-02 13:51 +0000 [r198794] Joshua Colp <jcolp@digium.com>
-
- * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions
- 198791 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r198791 |
- file | 2009-06-02 10:48:06 -0300 (Tue, 02 Jun 2009) | 5 lines
- Correct documentation for the register line, specifically where
- the domain should be specified. (closes issue #14367) Reported
- by: Nick_Lewis ........
-
-2009-06-01 21:04 +0000 [r198730] Russell Bryant <russell@digium.com>
-
- * channels/iax2-parser.c, /: Merged revisions 198729 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r198729 | russell | 2009-06-01 16:03:18 -0500 (Mon, 01 Jun 2009)
- | 2 lines Tell the IAX2 parser about more control frame types.
- ........
-
-2009-06-01 18:44 +0000 [r198629] Tilghman Lesher <tlesher@digium.com>
-
- * /, contrib/scripts/meetme.sql: Merged revisions 198626 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r198626 | tilghman | 2009-06-01 13:40:35 -0500 (Mon, 01
- Jun 2009) | 2 lines Add information for new meetme realtime
- fields ........
-
-2009-05-31 17:53 +0000 [r198471] Tilghman Lesher <tlesher@digium.com>
-
- * /, funcs/func_strings.c: Merged revisions 198470 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r198470 | tilghman | 2009-05-31 12:52:28 -0500 (Sun, 31 May 2009)
- | 2 lines Fix documentation for FIELDQTY. ........
-
-2009-05-31 01:48 +0000 [r198440] Eliel C. Sardanons <eliels@gmail.com>
-
- * /, res/res_timing_dahdi.c: Merged revisions 198437 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r198437 | eliel | 2009-05-30 21:22:15 -0400 (Sat, 30 May 2009) |
- 11 lines Avoid a crash when res_timing_dahdi is unloaded but
- wasn't properly loaded. if dahdi_test_timer() fails,
- timing_funcs_handle remains NULL causing a crash when calling
- ast_unregister_timing_interface() with a NULL pointer. (closes
- issue #15234) Reported by: eliel Patches: timing_dahdi1.diff
- uploaded by eliel (license 64) ........
-
-2009-05-31 01:21 +0000 [r198436] Russell Bryant <russell@digium.com>
-
- * res/res_smdi.c, /: Merged revisions 198312 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r198312 | russell | 2009-05-29 22:43:23 -0500 (Fri, 29 May 2009)
- | 12 lines Merged revisions 198311 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r198311 | russell | 2009-05-29 22:42:46 -0500 (Fri, 29 May 2009)
- | 5 lines Fix a crash that occurred when MWI SMDI messages
- expired. (closes issue #14561) Reported by: cmoss28 ........
- ................
-
-2009-05-30 20:22 +0000 [r198297-198397] Sean Bright <sean.bright@gmail.com>
-
- * res/res_jabber.c, /: Merged revisions 198375 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r198375 |
- seanbright | 2009-05-30 16:11:33 -0400 (Sat, 30 May 2009) | 13
- lines Properly terminate the receive buffer before sending to
- iksemel. aji_io_recv takes the maximum number of bytes to read
- (instead of the total buffer size), so we have to subtract 1 from
- our buffer size. Without this, when we receive packets that are
- larger than our buffer, iksemel will choke and things get wonky.
- (closes issue #15232) Reported by: lp0 Patches:
- 05302009_res_jabber.c.patch uploaded by seanbright (license 71)
- Tested by: seanbright, lp0 ........
-
- * res/res_jabber.c, /: Merged revisions 198371 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r198371 | seanbright | 2009-05-30 15:38:58 -0400 (Sat, 30 May
- 2009) | 19 lines Merged revisions 198370 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r198370 | seanbright | 2009-05-30 15:36:20 -0400 (Sat, 30 May
- 2009) | 12 lines Properly terminate AMI JabberSend response
- messages. The response message (either Error or Success) needs an
- extra trailing \r\n after the fields to inform the client that
- the message is complete. (closes issue #14876) Reported by: srt
- Patches: 05302009_1.4_res_jabber.c.diff uploaded by seanbright
- (license 71) asterisk_14876.patch uploaded by srt (license 378)
- trunk-14876-2.diff uploaded by phsultan (license 73) ........
- ................
-
- * apps/app_dial.c, /: Merged revisions 198285 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r198285 | seanbright | 2009-05-29 23:26:06 -0400 (Fri, 29 May
- 2009) | 15 lines Merged revisions 198251 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r198251 | seanbright | 2009-05-29 22:46:41 -0400 (Fri, 29 May
- 2009) | 8 lines Treat an empty FORWARD_CONTEXT the same way we
- treat a missing one. (closes issue #15056) Reported by:
- p_lindheimer Patches: 05292009_bug15056.diff uploaded by
- seanbright (license 71) Tested by: p_lindheimer ........
- ................
-
-2009-05-30 02:35 +0000 [r198250] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 198248 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r198248 |
- file | 2009-05-29 23:31:48 -0300 (Fri, 29 May 2009) | 2 lines
- When removing all packets from a dialog we also need to free the
- data if present. ........
-
-2009-05-29 23:05 +0000 [r198148-198188] Russell Bryant <russell@digium.com>
-
- * /, configs/modules.conf.sample: Merged revisions 198186 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ........ r198186 | russell | 2009-05-29 18:04:31 -0500 (Fri, 29
- May 2009) | 2 lines Suggesting that only a single timing module
- be loaded is no longer necessary. ........
-
- * /, res/res_timing_pthread.c: Merged revisions 198183 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r198183 | russell | 2009-05-29 17:33:31 -0500 (Fri, 29 May 2009)
- | 2 lines Improve handling of trying to ACK too many timer
- expirations. ........
-
- * /, res/res_timing_pthread.c: Merged revisions 198146 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r198146 | russell | 2009-05-29 15:06:59 -0500 (Fri, 29 May 2009)
- | 38 lines Resolve issues with choppy sound when using
- res_timing_pthread. The situation that caused this problem was
- when continuous mode was being turned on and off while a rate was
- set for a timing interface. A very easy way to replicate this bug
- was to do a Playback() from behind a Local channel. In this
- scenario, a rate gets set on the channel for doing file playback.
- At the same time, continuous mode gets turned on and off about
- every 20 ms as frames get queued on to the PBX side channel from
- the other side of the Local channel. Essentially, this module
- treated continuous mode and a set rate as mutually exclusive
- states for the timer to be in. When I dug deep enough, I observed
- the following pattern: 1) Set timer to tick every 20 ms. 2) Wait
- almost 20 ms ... 3) Continuous mode gets turned on for a queued
- up frame 4) Continuous mode gets turned off 5) The timer goes
- back to its tick per 20 ms. state but starts counting at 0 ms. 6)
- Goto step 2. Sometimes, res_timing_pthread would make it 20 ms
- and produce a timer tick, but not most of the time. This is what
- produced the choppy sound (or sometimes no sound at all). Now,
- the module treats continuous mode and a set rate as completely
- independent timer modes. They can be enabled and disabled
- independently of each other and things work as expected. (closes
- issue #14412) Reported by: dome Patches: issue14412.diff.txt
- uploaded by russell (license 2) issue14412-1.6.1.0.diff.txt
- uploaded by russell (license 2) Tested by: DennisD, russell
- ........
-
-2009-05-29 19:26 +0000 [r198111] Eliel C. Sardanons <eliels@gmail.com>
-
- * CREDITS, /: Merged revisions 198083 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r198083 |
- eliel | 2009-05-29 15:18:35 -0400 (Fri, 29 May 2009) | 3 lines
- Apply anti-spam obfuscation to an email address. ........
-
-2009-05-29 19:14 +0000 [r198075] Matthew Nicholson <mnicholson@digium.com>
-
- * main/cdr.c, main/channel.c, /, include/asterisk/cdr.h: Merged
- revisions 198072 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r198072 | mnicholson | 2009-05-29 14:04:24 -0500 (Fri, 29 May
- 2009) | 21 lines Merged revisions 198068 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r198068 | mnicholson | 2009-05-29 13:53:01 -0500 (Fri, 29 May
- 2009) | 15 lines Use AST_CDR_NOANSWER instead of AST_CDR_NULL as
- the default CDR disposition. This change also involves the
- addition of an AST_CDR_FLAG_ORIGINATED flag that is used on
- originated channels to distinguish: them from dialed channels.
- (closes issue #12946) Reported by: meral Patches: null-cdr2.diff
- uploaded by mnicholson (license 96) Tested by: mnicholson,
- dbrooks (closes issue #15122) Reported by: sum Tested by: sum
- ........ ................
-
-2009-05-29 18:40 +0000 [r198066] Joshua Colp <jcolp@digium.com>
-
- * /, main/file.c: Merged revisions 198064 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r198064 |
- file | 2009-05-29 15:39:04 -0300 (Fri, 29 May 2009) | 2 lines Fix
- a memory leak of the write buffer when writing a file. ........
-
-2009-05-29 18:18 +0000 [r198008] Sean Bright <sean.bright@gmail.com>
-
- * Makefile, /: Merged revisions 198000 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r198000 | seanbright | 2009-05-29 14:15:15 -0400 (Fri, 29 May
- 2009) | 15 lines Merged revisions 197998 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r197998 | seanbright | 2009-05-29 14:14:12 -0400 (Fri, 29 May
- 2009) | 8 lines Fix 'make config' target for Slackware. There was
- a missing semi-colon after the echo statement in the Makefile
- that was causing problems for some users. Fix suggested by
- reporter. (closes issue #15225) Reported by: pdavis ........
- ................
-
-2009-05-29 16:29 +0000 [r197994] Russell Bryant <russell@digium.com>
-
- * /, res/res_timing_pthread.c: Merged revisions 197960 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r197960 | russell | 2009-05-29 11:15:30 -0500 (Fri, 29 May 2009)
- | 2 lines Trim trailing whitespace so that I can work on this bug
- without it bothering me. :-) ........
-
-2009-05-28 23:54 +0000 [r197894] Leif Madsen <lmadsen@digium.com>
-
- * apps/app_mixmonitor.c, /: Merged revisions 197828 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r197828 | lmadsen | 2009-05-28 18:04:00 -0400 (Thu, 28 May 2009)
- | 8 lines Update documentation in MixMonitor. Updated the
- MixMonitor documentation for the 'b' option so that it is more
- obvious that you must not optimize away the Local channel when
- using this option. (closes issue #14829) Reported by: licedey
- Tested by: mmichelson, licedey, lmadsen ........
-
-2009-05-28 18:50 +0000 [r197703] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c, /: Merged revisions 197697 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r197697 | file | 2009-05-28 15:45:11 -0300 (Thu, 28 May 2009) | 2
- lines Fix a bug where the trunkmtu setting was not set to the
- default value of 1240 on load but was on reload. ........
-
-2009-05-28 16:15 +0000 [r197625] Eliel C. Sardanons <eliels@gmail.com>
-
- * /, channels/chan_sip.c: Merged revisions 197621 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r197621 | eliel | 2009-05-28 12:01:48 -0400 (Thu, 28 May 2009) |
- 19 lines Merged revisions 197562 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r197562 | eliel | 2009-05-28 11:21:32 -0400 (Thu, 28 May 2009) |
- 13 lines Use the address we already know when reloading a peer
- with nat=yes. If we already have an address for a peer, and we
- are reloading the sip configuration, try to use that address to
- contact the peer, instead of getting it from the Contact. (closes
- issue #15194) Reported by: ibc Patches: sip.patch uploaded by
- eliel (license 64) Tested by: manwe ........ ................
-
-2009-05-28 15:44 +0000 [r197548-197619] Mark Michelson <mmichelson@digium.com>
-
- * main/rtp.c, /, channels/chan_sip.c, include/asterisk/rtp.h:
- Merged revisions 197606 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r197606 | mmichelson | 2009-05-28 10:32:19 -0500 (Thu, 28 May
- 2009) | 22 lines Recorded merge of revisions 197588 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/branches/1.4
- ........ r197588 | mmichelson | 2009-05-28 10:27:49 -0500 (Thu,
- 28 May 2009) | 16 lines Allow for media to arrive from an
- alternate source when responding to a reinvite with 491. When we
- receive a SIP reinvite, it is possible that we may not be able to
- process the reinvite immediately since we have also sent a
- reinvite out ourselves. The problem is that whoever sent us the
- reinvite may have also sent a reinvite out to another party, and
- that reinvite may have succeeded. As a result, even though we are
- not going to accept the reinvite we just received, it is
- important for us to not have problems if we suddenly start
- receiving RTP from a new source. The fix for this is to grab the
- media source information from the SDP of the reinvite that we
- receive. This information is passed to the RTP layer so that it
- will know about the alternate source for media. Review:
- https://reviewboard.asterisk.org/r/252 ........ ................
-
- * main/audiohook.c, apps/app_chanspy.c, /,
- include/asterisk/audiohook.h: Merged revisions 197543 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r197543 | mmichelson | 2009-05-28 09:58:06 -0500
- (Thu, 28 May 2009) | 27 lines Merged revisions 197537 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r197537 | mmichelson | 2009-05-28 09:49:13 -0500 (Thu, 28 May
- 2009) | 21 lines Add flags to chanspy audiohook so that audio
- stays in sync. There are two flags being added to the chanspy
- audiohook here. One is the pre-existing
- AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set, we ensure that
- the read and write slinfactories on the audiohook do not skew
- beyond a certain tolerance. In addition, there is a new audiohook
- flag added here, AST_AUDIOHOOK_SMALL_QUEUE. With this flag set,
- we do not allow for a slinfactory to build up a substantial
- amount of audio before flushing it. For this particular issue,
- this means that the person spying on the call will hear the
- conversations in real time with very little delay in the audio.
- (closes issue #13745) Reported by: geoffs Patches: 13745.patch
- uploaded by mmichelson (license 60) Tested by: snblitz ........
- ................
-
-2009-05-28 14:56 +0000 [r197471-197542] Joshua Colp <jcolp@digium.com>
-
- * /, main/utils.c: Merged revisions 197538 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r197538 |
- file | 2009-05-28 11:51:43 -0300 (Thu, 28 May 2009) | 5 lines Fix
- a bug in stringfields where it did not actually free the pools of
- memory. (closes issue #15074) Reported by: pj ........
-
- * /, channels/chan_sip.c: Merged revisions 197467 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r197467 | file | 2009-05-28 10:47:45 -0300 (Thu, 28 May 2009) |
- 15 lines Merged revisions 197466 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r197466 | file | 2009-05-28 10:44:58 -0300 (Thu, 28 May 2009) | 8
- lines Fix a bug where the flag indicating the presence of rport
- would get overwritten by the nat setting. The presence of rport
- is now stored as a separate flag. Once the dialog is setup and
- authenticated (or it passes through unauthenticated) the proper
- nat flag is set. (closes issue #13823) Reported by: dimas
- ........ ................
-
-2009-05-28 11:40 +0000 [r197441] Gavin Henry <ghenry@suretecsystems.com>
-
- * contrib/scripts/asterisk.ldap-schema,
- contrib/scripts/asterisk.ldif, doc/ldap.txt,
- configs/res_ldap.conf.sample: issue #15155 and issue #15156 from
- trunk
-
-2009-05-27 23:49 +0000 [r197375] Tilghman Lesher <tlesher@digium.com>
-
- * /, main/xml.c: Merged revisions 197374 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r197374 |
- tilghman | 2009-05-27 18:48:15 -0500 (Wed, 27 May 2009) | 2 lines
- Revert commit 192032. This define is needed on Mac OS X. ........
-
-2009-05-27 22:23 +0000 [r197336] Kevin P. Fleming <kpfleming@digium.com>
-
- * include/asterisk/agi.h, /: Merged revisions 197335 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r197335 | kpfleming | 2009-05-27 17:21:53 -0500 (Wed, 27 May
- 2009) | 3 lines Ensure that this header includes xmldoc.h, since
- it depends on it. ........
-
-2009-05-27 20:11 +0000 [r197263] Sean Bright <sean.bright@gmail.com>
-
- * Makefile, /: Merged revisions 197260 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r197260 |
- seanbright | 2009-05-27 16:08:16 -0400 (Wed, 27 May 2009) | 6
- lines Use bash explicitly when calling build_tools/mkpkgconfig
- from the Makefile. Since we use bashisms in
- build_tools/mkpkgconfig, we should call on bash explicitly when
- running from the Makefile, otherwise we get errors during a 'make
- install.' (closes issue #15209) Reported by: seandarcy ........
-
-2009-05-27 19:30 +0000 [r197247] Tilghman Lesher <tlesher@digium.com>
-
- * /, funcs/func_cut.c: Recorded merge of revisions 197209 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r197209 | tilghman | 2009-05-27 14:20:56 -0500
- (Wed, 27 May 2009) | 12 lines Recorded merge of revisions 197194
- via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r197194 | tilghman | 2009-05-27 14:09:42 -0500 (Wed, 27 May 2009)
- | 5 lines Use a different determinator on whether to print the
- delimiter, since leading fields may be blank. (closes issue
- #15208) Reported by: ramonpeek Patch by me, though inspired in
- part by a patch from ramonpeek ........ ................
-
-2009-05-27 17:28 +0000 [r197176] Jeff Peeler <jpeeler@digium.com>
-
- * main/channel.c, include/asterisk/channel.h: Fix broken attended
- transfers The bridge was terminating immediately after the
- attended transfer was completed. The problem was because upon
- reentering ast_channel_bridge nexteventts was checked to see if
- it was set and if so could possibly return AST_BRIDGE_COMPLETE.
- (closes issue #15183) Reported by: andrebarbosa Tested by:
- andrebarbosa, tootai, loloski
-
-2009-05-27 16:12 +0000 [r196950-197092] Sean Bright <sean.bright@gmail.com>
-
- * configs/smdi.conf.sample, configs/extensions.conf.sample,
- configs/sla.conf.sample, configs/chan_dahdi.conf.sample, /,
- configs/vpb.conf.sample: Merged revisions 197089 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r197089 | seanbright | 2009-05-27 12:07:57 -0400 (Wed, 27 May
- 2009) | 6 lines Fix references to /etc/dahdi/system.conf and
- /etc/asterisk/chan_dahdi.conf in the sample configuration files.
- (closes issue #15207) Reported by: seandarcy ........
-
- * /, channels/chan_alsa.c: Merged revisions 196988 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r196988 | seanbright | 2009-05-27 09:02:54 -0400 (Wed, 27 May
- 2009) | 9 lines Display an error message when chan_alsa fails to
- load due to a missing or inaccessible configuration file. Before
- this change, when chan_alsa failed to load due to a missing or
- inaccessible configuration file, no message would be displayed.
- With this change, when chan_alsa fails to load due to a missing
- or inaccessible configuration file, a message will be displayed.
- (closes issue #14760) Reported by: Nick_Lewis Patches:
- chan_alsa.c-confload.patch uploaded by Nick (license 657)
- ........
-
- * main/xmldoc.c, /: Merged revisions 196948 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r196948 |
- seanbright | 2009-05-26 18:43:21 -0400 (Tue, 26 May 2009) | 8
- lines Reset the terminal to the correct fg/bg after XML
- documenation is rendered. (closes issue #15200) Reported by:
- ajohnson Patches: 05262009_xmldoc.patch uploaded by seanbright
- (license 71) Tested by: ajohnson ........
-
- * main/manager.c, /: Merged revisions 196945 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r196945 |
- seanbright | 2009-05-26 18:38:05 -0400 (Tue, 26 May 2009) | 13
- lines Add ActionID to CoreShowChannel event. There is
- inconsistency in how we handle manager responses that are lists
- of items and, unfortunately, third parties have come to rely on
- ActionID being on every event within those lists instead of just
- keeping track of the ActionID for the current response. This
- change makes CoreShowChannels include the ActionID with each
- CoreShowChannel event generated as a result of it being called.
- (closes issue #15001) Reported by: sum Patches:
- patchactionid2.patch uploaded by sum (license 766) ........
-
-2009-05-26 22:44 +0000 [r196870-196949] Russell Bryant <russell@digium.com>
-
- * /, autoconf/ast_check_osptk.m4 (added), configure,
- include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
- 196946 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r196946 |
- russell | 2009-05-26 17:40:34 -0500 (Tue, 26 May 2009) | 8 lines
- Update configure script to check for OSP toolkit 3.5.0. (closes
- issue #14988) Reported by: tzafrir Patches: configure.ac.diff
- uploaded by homesick (license 91) new_ast_check_osptk.m4 uploaded
- by homesick (license 91) ........
-
- * /, res/res_convert.c: Merged revisions 196843 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ................
- r196843 | russell | 2009-05-26 13:20:57 -0500 (Tue, 26 May 2009)
- | 16 lines Merged revisions 196826 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r196826 | russell | 2009-05-26 13:14:36 -0500 (Tue, 26 May 2009)
- | 9 lines Resolve a file handle leak. The frames here should have
- always been freed. However, out of luck, there was never any
- memory leaked. However, after file streams became reference
- counted, this code would leak the file stream for the file being
- read. (closes issue #15181) Reported by: jkroon ........
- ................
-
-2009-05-26 16:39 +0000 [r196793] Sean Bright <sean.bright@gmail.com>
-
- * apps/app_queue.c, /: Merged revisions 196792 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r196792 |
- seanbright | 2009-05-26 12:38:54 -0400 (Tue, 26 May 2009) | 2
- lines Add a missing unref for queues in handle_statechange.
- ........
-
-2009-05-26 13:47 +0000 [r196661-196724] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: Merged revisions 196721 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r196721 |
- file | 2009-05-26 10:43:13 -0300 (Tue, 26 May 2009) | 7 lines Fix
- a bug where the sip unregister CLI command did not completely
- unregister the peer. (closes issue #15118) Reported by: alecdavis
- Patches: chan_sip_unregister.diff2.txt uploaded by alecdavis
- (license 585) ........
-
- * contrib/scripts/safe_asterisk, /: Merged revisions 196658 via
- svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r196658 | file | 2009-05-26 10:06:50 -0300 (Tue,
- 26 May 2009) | 14 lines Merged revisions 196657 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r196657 | file | 2009-05-26 10:06:09 -0300 (Tue, 26 May 2009) | 7
- lines Remove some bash specific stuff from safe_asterisk. (closes
- issue #10812) Reported by: paravoid Patches:
- safe_asterisk_bashism.diff uploaded by tzafrir (license 46)
- ........ ................
-
-2009-05-23 05:29 +0000 [r196487] Moises Silva <moises.silva@gmail.com>
-
- * channels/chan_dahdi.c, /: Merged revisions 196456 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk ........
- r196456 | moy | 2009-05-22 23:27:47 -0500 (Fri, 22 May 2009) | 1
- line set MFCR2_CATEGORY just when starting the pbx ........
-
-2009-05-22 21:59 +0000 [r196452] David Vossel <dvossel@digium.com>
-
- * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions
- 196416 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r196416 |
- dvossel | 2009-05-22 16:09:45 -0500 (Fri, 22 May 2009) | 19 lines
- SIP set outbound transport type from Registration In sip.conf the
- transport option allows for the configuration of what transport
- types (udp, tcp, and tls) a peer will accept, but only the first
- type listed was used for outbound connections. This patch changes
- this. Now the default transport type is only used until the peer
- registers. When registration takes place the transport type is
- parsed out of the Contact header. If the Contact header's
- transport type is equal to one that the peer supports, the peer's
- default transport type for outbound connections is set to match
- the Contact header's type. If the Contact header's transport type
- is not present, then the peer's default transport type is set to
- match the one the peer registered with. When a peer unregisters
- or the registration expires, the default transport type for that
- peer is reset. (closes issue #12282) Reported by: rjain Patches:
- reg_patch_1.diff uploaded by dvossel (license 671) Tested by:
- dvossel (closes issue #14727) Reported by: pj Patches:
- reg_patch_3.diff uploaded by dvossel (license 671) Tested by: pj,
- dvossel Review: https://reviewboard.asterisk.org/r/249/ ........
-
-2009-05-22 19:48 +0000 [r196378] Eliel C. Sardanons <eliels@gmail.com>
-
- * /, apps/app_minivm.c: Merged revisions 196377 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/trunk ........ r196377 |
- eliel | 2009-05-22 15:38:33 -0400 (Fri, 22 May 2009) | 11 lines
- Unregister every registered application by MiniVM. The MinivmMWI
- application was not being unregistered on unload and we were not
- able to load again the module or reload it. (closes issue #15174)
- Reported by: junky Patches: unregister_minivm_mwi.diff uploaded
- by junky (license 177) ........
-
-2009-05-22 13:59 +0000 [r196120] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_misdn.c, /: Merged revisions 196117 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r196117 | file | 2009-05-22 10:56:47 -0300 (Fri,
- 22 May 2009) | 12 lines Merged revisions 196116 via svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r196116 | file | 2009-05-22 10:54:17 -0300 (Fri, 22 May 2009) | 5
- lines Fix a bug where using immediate with mISDN caused a cause
- code of 16 to get sent back instead of 1 if the 's' extension did
- not exist. (closes issue #12286) Reported by: lmamane ........
- ................
-
-2009-05-21 19:15 +0000 [r196000] David Vossel <dvossel@digium.com>
-
- * channels/chan_iax2.c, /: Merged revisions 195995 via svnmerge
- from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r195995 | dvossel | 2009-05-21 14:11:49 -0500
- (Thu, 21 May 2009) | 20 lines Merged revisions 195991 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r195991 | dvossel | 2009-05-21 14:04:56 -0500 (Thu, 21 May 2009)
- | 14 lines Sign problem calculating timestamp for iax frame leads
- to no audio on the receiving peer. There are rare cases in which
- a frame's delivery timestamp is slightly less than the iax2_pvt's
- offset. This causes the pvt's timestamp to be a small negative
- number, but since the timestamp value is unsigned it looks like a
- huge positive number. This patch checks for this negative case
- and sets the ms to zero. A similar check is already done right
- below this one in the 'else' statement. (closes issue #15032)
- Reported by: guillecabeza Patches: chan_iax2.c.patch_timestamp
- uploaded by guillecabeza (license 380) Tested by: guillecabeza
- (closes issue #14216) Reported by: Andrey Sofronov ........
- ................
-
-2009-05-21 15:57 +0000 [r195883] Matthew Nicholson <mnicholson@digium.com>
-
- * main/cdr.c, /, include/asterisk/cdr.h: Merged revisions 195882
- via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
- ................ r195882 | mnicholson | 2009-05-21 10:33:55 -0500
- (Thu, 21 May 2009) | 20 lines Merged revisions 195881 via
- svnmerge from
- https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
- r195881 | mnicholson | 2009-05-21 10:25:50 -0500 (Thu, 21 May
- 2009) | 13 lines This commit prevents cdr records with
- AST_CDR_FLAG_ANSLOCKED and AST_CDR_FLAG_LOCKED from being updated
- in certain cases. This is accomplished by adding two functions to
- update the answer time and disposition of calls that checks for
- the proper l