asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	ba8e183938	Channel initialization failure causes crashes. __ast_channel_alloc_ap() has several points in the initialization of a new channel structure where it could fail. Since the channel structure is now an ao2 object, the destructor callback needs to be able to handle clean up when the structure setup is incomplete. Problems corrected: 1) Failing to setup the alertpipe would not unreference the structure but free it directly. Doing this to an ao2_object is very bad. 2) File descriptors need to be initialized to -1 before a construction failure could occur so the destructor will not close unopened descriptors. 3) The destructor needs to check that the string field has been initialized before using any string field values. Crashes expected. 4) The destructor should not notify devstate if the device name is empty. It is a waste of cycles and a couple ERROR log messages are generated. Review: https://reviewboard.asterisk.org/r/675/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265174 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	73e8c7572e	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/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264997 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	43991ce806	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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264828 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	815d7bfe44	Let ExtensionState resolve dynamic hints. (closes issue #16623) Reported by: tilghman Patches: 20100116__issue16623.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264779 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	dafb48fe09	Avoid crash in generic CC agent init if caller name or number is NULL. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264711 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	2aa0c11679	Correct 'all logger levels' patch to work properly. Nick Lewis pointed out that the patch as committed wouldn't actually include dynamic logger levels, which was missed by the other reviewers. Thanks! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264497 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	6bb45831eb	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264452 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	d7e9d07156	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) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264400 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	6eaf9b874f	Cast an unsigned int to a signed int when comparing it with 0. (AST-377) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264379 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	07df131a7f	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264204 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	e3c9e6ae86	Fix compilation problem with previous commit. (issue #16009) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264161 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	e77efbc12e	Add ability for logger channels to include all levels. Now that Asterisk modules can dynamically create and destroy logger levels on demand, it's useful to be able to configure a logger channel (console, file, whatever) to be able to accept log messages from all levels, even levels created dynamically. This patch adds support for this, by allowing the '*' level name to be used in logger.conf. Review: https://reviewboard.asterisk.org/r/663/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264160 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	a8a1961be7	Add ability to hangup all channels from the CLI. Added the keyword 'all' to the 'channel hangup request' CLI command so that you can request all channels to be hungup without having to restart Asterisk. (closes issue #16009) Reported by: moy Patches: hangup-all-rev-221688.patch uploaded by moy (license 222) Tested by: moy, russell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264117 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	f55aff74ed	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 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263950 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	10789ef88a	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263904 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	e3ac20a7f6	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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263640 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	b5d5cc565f	Enhancements to connected line and redirecting work. From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	fa5350f7d7	Missing newlines added to Set-Cookie line in manager.c Sean Bright pointed out that we lost a set of newline characters in commit 190349 on a line I had recently changed. Yay for code review on commits. (issue #17231, #10961) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263460 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	193d495a8a	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 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263457 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	c44da92360	Improve some very confusing structure names in astobj2.c As pointed out by 'akshayb' on #asterisk-dev, the code here called a list of bucket entries a 'bucket', and the entries within the bucket were called 'bucket_list'. This made the code very hard to understand without reading all of it... so I've renamed 'bucket_list' to 'bucket_entry' to clarify the purpose of the structure. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263208 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	420acb8f0a	Fix build on linux. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262896 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	8d6ee962c7	Add kqueue(2) implementation to Asterisk in various places. This will save a considerable amount of CPU on the BSDs, including Mac OS X, as it eliminates several places in the code that we previously used a busy loop. Additionally, this adds a res_timing interface, using kqueue timers. Review: https://reviewboard.asterisk.org/r/543/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262852 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	7d53dc86d6	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262800 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	12631bc3a0	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261496 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	b2f59bea24	New 'manager show settings' CLI command. 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: https://origsvn.digium.com/svn/asterisk/trunk@261180 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	6a0ea1d79e	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 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261095 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Eliel C. Sardanons	caa2eff30c	Avoid making AstData depend on libxml2 to compile. 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: https://origsvn.digium.com/svn/asterisk/trunk@260521 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	623ba816fa	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. 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: https://origsvn.digium.com/svn/asterisk/trunk@260292 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	d4358a46a9	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/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260050 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	6722251986	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/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259870 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	5fd23b2ed4	Shuffle some casts to make builds on bamboo happier. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259533 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	7108038175	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259439 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	57c8eea6fe	Change cc_ref and cc_unref from macros to inline functions. The hope is that Solaris won't be as whiny after this change. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259357 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	af6690ba7f	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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259105 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	317a12d950	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 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259023 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	99a7b2fed0	Fix previous commit. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258675 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	8c41f2db82	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258671 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	52a8ddba51	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258632 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	9e3f5fa6fb	Remove ABI differences that occured when compiling with DEBUG_THREADS. "Bad Things" would happen if Asterisk was compiled with DEBUG_THREADS, but a loaded module was not (or vice versa). This also immensely simplifies the lock code, since there are no longer 2 separate versions of them. Review: https://reviewboard.asterisk.org/r/508/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258557 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Eliel C. Sardanons	a753e8878b	Asterisk data retrieval API. This module implements an abstraction for retrieving and exporting asterisk data. Developed by: Brett Bryant <brettbryant@gmail.com> Eliel C. Sardanons (LU1ALY) <eliels@gmail.com> For the Google Summer of code 2009 Project. Documentation can be found in doxygen format and inside the header include/asterisk/data.h Review: https://reviewboard.asterisk.org/r/275/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258517 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Julian Lyndon-Smith	d85650e4aa	Added MixMonitorMute manager command Added a new manager command to mute/unmute MixMonitor audio on a channel. Added a new feature to audiohooks so that you can mute either read / write (or both) types of frames - this allows for MixMonitor to mute either side of the conversation without affecting the conversation itself. (closes issue #16740) Reported by: jmls Review: https://reviewboard.asterisk.org/r/487/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258190 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	c7cf47ce7b	Change log message to match severity. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257949 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	7965dd9509	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	9674766487	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257810 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	8ced3317ed	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/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257560 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	8b7a90a026	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257262 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	2724f89bba	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/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257146 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	69c252c290	Fix issue where recall would not happen when it should. Specifically, the situation would happen when multiple callers would request CC for a single generically-monitored device. If the monitored device became available but the caller did not answer the recall, then there was nothing that would poke the CC core to let it know that it should attempt to recall someone else instead. After careful consideration, I came to the conclusion that the only area of Asterisk that needed to be touched was the generic CC monitor. All other types of CC would require something outside of Asterisk to invoke a recall for a separate device. This was accomplished by changing the generic monitor destructor to poke other generic monitor instances if the device is currently available and the specific instance was currently not suspended. In order to not accidentally trigger recalls at bad times, the fit_for_recall flag was also added to the generic_monitor_instance_list struct. This gets set as soon as a monitored device becomes available. It gets cleared if a CCNR request triggers the creation of a new generic monitor instance. By doing this, we don't accidentally try to recall a device when the monitored device was being monitored for CCNR and never actually became available for recall in the first place. This error was discovered by Steve Pitts during in-house testing at Digium. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256985 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	d2e1f421fa	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) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256821 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	9afa6af881	Remove status_response callbacks where they are not needed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256661 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago

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2779 Commits (45895dc3200d4636efe0e1c2ead24cd7c2cb73dd)