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2010-05-06Update ChangeLogv1.4.32-rc1lmadsen1-0/+24
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.32-rc1@261554 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-06Use autotagged externalslmadsen0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.32-rc1@261544 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-06Importing files for 1.4.32-rc1 release.lmadsen3-0/+28516
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.32-rc1@261543 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-06Creating tag for the release of asterisk-1.4.32-rc1lmadsen0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.32-rc1@261542 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-05Registration fix for SIP realtime.pabelanger1-3/+7
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/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@261274 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-04Add a tiny corner case to the previous committilghman1-1/+5
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@261094 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-04Protect against overflow, when calculating how long to wait for a frame.tilghman1-1/+1
(closes issue #17128) Reported by: under Patches: d.diff uploaded by under (license 914) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@261093 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-04Voicemail transfer to operator should occur immediately, not after main menu.jpeeler1-7/+8
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 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@260923 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-04Fix FILTER() examples to work in 1.4tringenbach1-4/+4
Review: https://reviewboard.asterisk.org/r/644/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@260887 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-04Fix fallout from removing from configure script. Pointed out by philipp64 ↵qwell1-1/+1
on #asterisk-dev git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@260801 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-03Should have removed /usr/lib/ part. Thanks Qwell.pabelanger1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@260662 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-03non-root make install PREFIX=/tmp fails.pabelanger1-1/+1
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 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@260661 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-03Minor typo pointed out by pabelanger on IRC.lmadsen1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@260569 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-30Ensure channel state is not incorrectly set in the case of a very early answer.jpeeler1-1/+1
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) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@260434 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-30Fix potential crash from race condition due to accessing channel data ↵mmichelson1-1/+13
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 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@260345 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-29DTMF CallerID detection problems.rmudgett1-12/+22
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 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@260195 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-29Fixes crash in audiohook_write_listdvossel2-14/+43
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/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@260049 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-28resolves deadlocks in chan_localdvossel2-10/+16
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/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@259858 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-28Update config.guess.lmadsen1-92/+61
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. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@259852 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-28Add AC_CONFIG_AUX_DIR to configure script, so systems without install can ↵qwell2-1/+27
use install-sh from our source dir. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@259847 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-28Missed this when removing $IDqwell1-1/+0
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@259833 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-28Remove usage of `id` since it isn't useful and was causing breakge.qwell3-48/+2
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). git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@259748 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-28Do not play goodbye prompt after timeout of message review.jpeeler1-2/+2
ABE-2124 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@259664 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-27DAHDI "WARNING" message is confusing and vaguermudgett1-3/+14
"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 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@259531 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-27Update sounds files.lmadsen1-2/+2
* 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 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@259526 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-27Add gar to the check for AR for those silly OSes (Solaris) that don't have ar.qwell2-59/+64
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@259441 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-27Support the silly OSes that don't have ar and strip.qwell2-19456/+6389
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. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@259352 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-27hidecalleridname parameter in chan_dahdi.confrmudgett2-0/+5
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 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@259270 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-26Let compilation succeed warning-free when DONT_OPTIMIZE is turned off.mmichelson1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@259104 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-26Prevent Newchannel manager events for dummy channels.mmichelson1-1/+7
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 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@259018 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-25When StopMonitor is called, ensure that it will not be restarted by a ↵tilghman1-0/+2
channel event. (closes issue #16590) Reported by: kkm Patches: resmonitor-16590-trunk.239289.diff uploaded by kkm (license 888) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@258775 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-22Fix broken CDR behavior.mnicholson3-7/+8
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 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@258670 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-21Fix looping forever when no input received in certain voicemail menu scenarios.jpeeler1-2/+10
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 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@258432 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-20Play correct prompt when voicemail store failure occurs after attempted forward.jpeeler1-29/+55
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 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@258029 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-19make app_voicemail compile with IMAP_STORAGEjpeeler1-1/+9
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@257856 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-16Make the mixmonitor thread process audio frames fasterdhubbard1-5/+6
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/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@257686 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-15Allow application options with arguments to contain parentheses, through a ↵tilghman2-1/+33
variety of escaping techniques. Fixes SWP-1194 (ABE-2143). Review: https://reviewboard.asterisk.org/r/604/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@257544 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-15Don't recreate peer, when responding to a repeated deregistration attempt.tilghman1-0/+15
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 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@257467 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-15Update backtrace.txt documentation.lmadsen1-27/+105
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 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@257426 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-15Update address of the bug tracker.lmadsen1-1/+3
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@257342 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-14When forwarding a message, ensure that prepending works correctly.tilghman1-42/+25
This is a regression in 1.4, only. (closes issue #17103) Reported by: mglazer Patches: 20100408__issue17103.diff.txt uploaded by tilghman (license 14) Tested by: tilghman git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@257266 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-13Add an option to restore past broken behavor of the Events manager actionmnicholson2-1/+29
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/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@257070 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-12Add How-To document on collecting debugging info for issues.asterisk.orglmadsen1-0/+89
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 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@256900 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-06DAHDI/PRI call to pri_channel_bridge() not protected by PRI lock.rmudgett1-7/+8
SWP-1231 ABE-2163 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@256225 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-02Resolve a deadlock that occurs due to a pointless call to ast_bridged_channel()russell1-24/+8
(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 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@256014 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-02Remove extremely verbose debug message.russell1-4/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@256009 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-31Ensure line terminators in email are consistent.tilghman1-12/+29
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/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@255591 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-31Add documentation clarifying when 't' and 'T' can be used.lmadsen2-3/+6
(closes issue #17021) Reported by: kovzol Tested by: lmadsen, kovzol, davidw, ebroad git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@255503 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-30Don't kill Asterisk if the H323 listener does not start.russell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@255409 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-30Don't make Asterisk not start if pbx_dundi fails to initialize.russell1-5/+5
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@255322 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-29 01:44:47 +0000