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2010-05-07Fix build on Linuxtilghman1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261964 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-07Double free crashtilghman1-0/+3
(closes issue #17245) Reported by: thedavidfactor Patches: 20100426__issue17245.diff.txt uploaded by tilghman (license 14) Tested by: murraytm git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261917 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-07Use the detected pthread building flags in every place, instead of ↵tilghman3-7283/+29379
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 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261913 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-07Update UPGRADE-1.6.txt stating insecure=very has been removed.lmadsen1-0/+6
(closes issue #17282) Reported by: stuarth Tested by: stuarth git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261867 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-07Fix deadlock in sig_pri when hanging up.jpeeler1-10/+10
The pri_dchannel thread currently violates locking order by locking the private and then attempting to queue a frame, which needs to lock the channel. Queueing a frame is unneccesary though and is actually a regression since sig_pri. All the places that currently use ast_softhangup_nolock now will just set the softhangup value directly as before. (closes issue #17216) Reported by: lmsteffan Patches: bug17216.patch uploaded by jpeeler (license 325) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261866 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-06Some code optimizations.rmudgett1-80/+88
* Made more places use pri_queue_control() instead of pri_queue_frame() and a local frame variable. * Made pri_queue_frame() use sig_pri_lock_owner(). pri_queue_frame() no longer releases the libpri access lock unless it is required. * Made the pri_queue_frame() and pri_queue_control() parameter list similar to sig_pri_lock_owner(). git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261822 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-06Merged revisions 261735 via svnmerge from jpeeler1-1/+4
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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261736 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-06Merged revisions 261608 via svnmerge from qwell1-2/+7
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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261609 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-06Permit more lines within a SIP body to be parsed.tilghman1-1/+1
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 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261560 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-06Add test case for removing random elements from a heap.russell1-2/+99
I modified the original patch for trunk to use the unit test API. (issue #17277) Reported by: cappucinoking Patches: test_heap.diff uploaded by cappucinoking (license 1036) Tested by: cappucinoking, russell git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261500 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-06Fix handling of removing nodes from the middle of a heap.russell1-10/+14
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 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261496 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-06When failing to configure, don't destroy 'cfg' twicetzafrir1-1/+1
Fixes a crash when some config section had an incorrect channel config. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261451 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-05Avoid a crash on SS7 channels.rmudgett1-2/+8
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261405 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-05Restore previous asterisk.conf syntax, where the directories aren't ↵russell2-21/+21
commented out. This fixes some breakage in the test suite, that uses the contents of asterisk.conf to discover the install layout on the system. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261364 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-05fixes sip native transferdvossel1-2/+1
The Refer-To header field containing the Replaces header in the URI was not being decoded properly. This caused invalid parsing between the caller id field and the domain resulting in a failed transfer. (closes issue #17284) Reported by: dvossel git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261316 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-05Merged revisions 261274 via svnmerge from pabelanger1-3/+7
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/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261314 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-05Prevent unnecessary warnings when getting rtpsource or rtpdest.mmichelson1-2/+4
If a recognized media type was present, but the media type was not enabled for the channel, then a warning would be emitted. For instance, attempting to get CHANNEL(rtpsource,video) on a call with no video would cause a warning message to appear. With this change, the warning will only appear if the stream argument is not recognized as being a media type that can be specified. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261313 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-05'queue reset stats' erroneously clears wrapuptime configuration.pabelanger1-1/+1
Resets each member's lastcall to 0 now. (closes issue #17262) Reported by: rain Patches: wrapuptime_reset_fix.diff uploaded by rain (license 327) Tested by: rain git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261232 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-05New 'manager show settings' CLI command.pabelanger4-19/+71
See the CHANGES file for more details. (closes issue #16343) Reported by: pabelanger Patches: issue16343.patch.v5 uploaded by pabelanger (license 224) Tested by: pabelanger, tilghman, lmadsen Review: https://reviewboard.asterisk.org/r/630/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261180 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-05New static asterisk.conf.sample file.pabelanger2-67/+100
This simply moves the functionality from the Makefile (cleaning it up) into an external asterisk.conf.samples file. Also updates formatting (easier to read) and grammar changes to asterisk.conf.samples. (closes issue #17027) Reported by: pabelanger Patches: 0017027.asterisk.conf.v6.patch uploaded by pabelanger (license 224) Tested by: qwell, lmadsen, pabelanger, chappell Review: https://reviewboard.asterisk.org/r/616/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261124 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-04Merged revisions 261093-261094 via svnmerge from tilghman1-2/+7
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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261095 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-04Add new possible value to autopause option to allow members to be autopaused ↵mmichelson3-6/+61
in all queues. See the CHANGES file and queues.conf.sample for more details. (closes issue #17008) Reported by: jlpedrosa Patches: queues.autopause_en_review.diff uploaded by jlpedrosa (license 1002) Review: https://reviewboard.asterisk.org/r/581/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261051 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-04The inalarm flag is not passed up from the sig_analog and sig_pri submodules.rmudgett5-38/+95
The CLI "dahdi show channel" command was not correctly reporting the InAlarm status. The inalarm flag is now consistently passed between chan_dahdi and submodules. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261007 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-04Merged revisions 260923 via svnmerge from jpeeler1-7/+8
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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@260924 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-04Merged revisions 260801 via svnmerge from qwell1-1/+1
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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@260802 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-03Add new admin features to meetme: Roll call, eject all, mute all, record ↵jpeeler2-3/+160
in-conf This patch adds the following in-conference admin DTMF features: *81 - Roll call (or simply user count if INTROUSER isn't enabled) *82 - Eject all non-admins *83 - Mute/unmute all non-admins *84 - Start recording the conference on the fly FWIW, this code uses newly recorded prompts. (closes issue #16379) Reported by: rfinnie Patches: meetme-enhancements-232771-v1.patch uploaded by rfinnie (license 940) modified slightly by me git-svn-id: http://svn.digium.com/svn/asterisk/trunk@260757 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-03Merged revisions 260661-260662 via svnmerge from pabelanger1-1/+1
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. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@260663 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-03Merged revisions 260569 via svnmerge from lmadsen1-1/+1
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. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@260570 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-02Avoid making AstData depend on libxml2 to compile.eliel2-0/+4
We have some functions inside the AstData API to get the tree in XML form, but it is not required at the moment to compile asterisk and we can disable that part of the API if we don't have libxml2 support. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@260521 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-30Merged revisions 260434 via svnmerge from jpeeler3-0/+17
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) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@260437 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-30Separate the uses of NUM_DCHANS and MAX_CHANNELS into PRI, SS7, and MFCR2 users.rmudgett3-54/+56
Created SIG_PRI_MAX_CHANNELS, SIG_PRI_NUM_DCHANS SIG_SS7_MAX_CHANNELS, SIG_SS7_NUM_DCHANS SIG_MFCR2_MAX_CHANNELS Also fixed the declaration of pollers[] in mfcr2_monitor(). It was dimensioned to the number of bytes in struct dahdi_mfcr2.pvts[] and not to the same dimension of the struct dahdi_mfcr2.pvts[]. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@260435 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-30Merged revisions 260345 via svnmerge from mmichelson1-3/+14
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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@260346 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-30Fix logic reversal error when queue callers join the queue.mmichelson1-1/+1
When a specific position is specified for the queue, the idea was that the caller cannot be placed ahead of higher-priority callers. Unfortunately, the logic was reversed so that the caller could ONLY be placed ahead of higher priority callers. Discovered while writing a unit test. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@260344 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-30Don't allow file descriptors to go above 64k, when we're closing them in a ↵tilghman1-0/+32
fork(2). This saves time, when, even though the system allows the process limit to be that high, the practical limit is much lower. Also introduce an additional optimization, in the form of using the CLOEXEC flag to close descriptors at the right time. (closes issue #17223) Reported by: dbackeberg Patches: 20100423__issue17223.diff.txt uploaded by tilghman (license 14) Tested by: dbackeberg git-svn-id: http://svn.digium.com/svn/asterisk/trunk@260292 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-30Logic fixups for a sample FREENUM dialplan context.tilghman1-5/+9
(closes issue #17263) Reported by: pprindeville Patches: freenum-dialplan.patch#3 uploaded by pprindeville (license 347) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@260280 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-29Merged revisions 260195 via svnmerge from rmudgett2-23/+44
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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@260231 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-29Pattern match fail.tilghman1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@260148 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-29Merged revisions 260049 via svnmerge from dvossel2-14/+44
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/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@260050 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-29Fix comment.rmudgett1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@260007 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-28Don't override peer context with domain context.mmichelson2-3/+6
(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/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@259957 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-28Merged revisions 259858 via svnmerge from dvossel2-10/+16
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/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@259870 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-28Merged revisions 259852 via svnmerge from lmadsen1-92/+61
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. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@259853 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-28Merged revisions 259847 via svnmerge from qwell2-1/+27
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. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@259848 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-28Merged revisions 259833 via svnmerge from qwell1-1/+0
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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@259837 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-28Merged revisions 259748 via svnmerge from qwell3-48/+2
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). ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@259760 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-28Merged revisions 259664 via svnmerge from jpeeler1-2/+2
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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@259672 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-27Fix compile on systems without HAVE_NULLSAFE_PRINTF defined.qwell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@259617 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-27Be more explicit about field naming in a test.qwell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@259587 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-27Merged revisions 259531 via svnmerge from rmudgett1-6/+24
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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@259538 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-27Shuffle some casts to make builds on bamboo happier.mmichelson1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@259533 f38db490-d61c-443f-a65b-d21fe96a405b
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