asterisk

Commit Graph

Author	SHA1	Message	Date
Tilghman Lesher	5da0ea7b8e	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 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280983 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	3043c8f34e	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/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@279946 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	19ecaa8b4d	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@279657 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	0bf59fb242	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) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@278273 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	7e4192b3ae	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 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@277776 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	5177d85be1	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. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@275023 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	c25cc0c6cf	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/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@273831 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	6ed96a542a	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@272531 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	444956b09f	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. ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@271691 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	bef7474554	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 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@271484 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	2fd6cbc156	Merged revisions 269417 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r269417 \| russell \| 2010-06-09 16:11:43 -0500 (Wed, 09 Jun 2010) \| 6 lines Resolve an invalid memory read on an event. Valgrind pointed out that attempting to get an IE value from an event that has no IEs produces an invalid memory read past the end of the event. Thanks to mmichelson for pointing the problem out to me and then testing the fix. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@269418 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	1a4abf2895	Fix some chan_oss build system problems (these changes were already in trunk). This is what caused a bunch of tests to fail on 1.6.2. They expected a console channel driver, but chan_oss was failing to load. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@268815 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	f5297544fc	Merged revisions 267862,267877 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r267862 \| tilghman \| 2010-06-03 21:58:55 -0500 (Thu, 03 Jun 2010) \| 5 lines As signed linear audio data is accessed as 16-bit values, certain processors require the values to be aligned in memory. (closes issue #16912) Reported by: michaelevdokimov ........ r267877 \| tilghman \| 2010-06-03 22:20:47 -0500 (Thu, 03 Jun 2010) \| 8 lines As signed linear audio data is accessed as 16-bit values, certain processors require the values to be aligned in memory. (closes issue #16912) Reported by: michaelevdokimov Patches: asterisk.patch uploaded by michaelevdokimov (license 997) Tested by: michaelevdokimov ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@267883 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	5de8e8a0b0	Merged revisions 267775 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r267775 \| tilghman \| 2010-06-03 20:20:17 -0500 (Thu, 03 Jun 2010) \| 14 lines Merged revisions 267759 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r267759 \| tilghman \| 2010-06-03 20:16:26 -0500 (Thu, 03 Jun 2010) \| 7 lines Make the default install path appear to be /usr on Linux, instead of /usr/local. Also, reorganize the options, so that they're more alphabetical. (closes issue #17013) Reported by: klaus3000 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@267787 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	6da8627ed4	Add fields back to ast_translator for ABI reasons. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@267527 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	efe0f4231e	Merged revisions 267492 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r267492 \| mmichelson \| 2010-06-03 12:09:11 -0500 (Thu, 03 Jun 2010) \| 6 lines Remove unnecessary code relating to PLC. The logic for handling generic PLC is now handled in ast_write in channel.c instead of in translation code. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@267507 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	3c047b9a31	Merged revisions 267065 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r267065 \| jpeeler \| 2010-06-02 12:29:35 -0500 (Wed, 02 Jun 2010) \| 12 lines Fix infinite loop when loading codec speex This changes the sample slinear frame data to contain non-zero data so that translation calculations for speex works when preprocessing and VAD is turned on. The encoder expects samples to be returned, but when attempted with the mentioned two options and silent sample frames everything was discarded. (closes issue #17240) Reported by: seandarcy Review: https://reviewboard.asterisk.org/r/682/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@267073 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	a199f18aa6	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 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@265748 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	a2b47e4eda	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 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@265521 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	d3ae642446	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 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@265091 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	f83214b612	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. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@265001 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	9a124ea569	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/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@264998 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	af727635c5	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/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@264453 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	f134c6effe	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 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@264250 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	2a3a1fe3cf	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 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@262516 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	15e31273f5	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 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@262105 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	c3cee12c21	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 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@261916 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	55993d7b13	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/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@260884 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	4b4569810a	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/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@260051 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	e862c28407	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/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@257597 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	d680d86948	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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@256373 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	52715df18f	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) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@255816 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	3aed281c75	Fix DEBUG_THREADS issue with out-of-tree modules. Take 2, without ABI breakage this time. Review: https://reviewboard.asterisk.org/r/588/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@254770 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	1c36715ade	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 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@254556 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	dbf0243679	Revert API change in release branches This re-renames ast_rtp_update_source to ast_rtp_new_source git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@253158 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	6bfb1fca95	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/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@252137 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	07a05534e0	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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@249895 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	b8fd3f54b2	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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@249407 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	a03d04345a	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... ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@247337 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	872c5f5412	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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@246989 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	e2e66d446c	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/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@246199 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	e8162c4245	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 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@244445 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	c2746e94a7	Merged revisions 242521 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r242521 \| tilghman \| 2010-01-24 00:40:31 -0600 (Sun, 24 Jan 2010) \| 15 lines Merged revisions 242520 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r242520 \| tilghman \| 2010-01-24 00:33:01 -0600 (Sun, 24 Jan 2010) \| 8 lines Only rebuild bison and flex source files on demand, if bison and flex are detected by the configure script. Changed after discussion on the -dev list about possible unnecessary build failures, due to checkouts/untars causing these special source files to possibly be newer than their resulting C files. This should additionally ensure that nobody need learn about extra Makefile arguments to ensure the proper files get rebuilt when changes are made to these special source files. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@242522 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	46c684eed9	Merged revisions 238635 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r238635 \| dvossel \| 2010-01-08 13:39:30 -0600 (Fri, 08 Jan 2010) \| 22 lines fixes AUDIOHOOK_INHERIT regression During the process of removing an audiohook from one channel and attaching it to another the audiohook's status is updated to DONE and then back to whatever it was previously. Typically updating the status after setting it to DONE is not a good idea because DONE can trigger unrecoverable audiohook destruction events... because of this a conditional check was added to audiohook_update_status to explicitly prevent the audiohook from ever changing after being set to DONE. It was this check that prevented audiohook inherit from work properly though. Now ast_audiohook_move_by_source is treated as a special exception, as the audiohook must be returned to its previous status after attaching it to the new channel. This is only a safe operation because the audiohook's lock is held the entire time, otherwise this could cause trouble. (closes issue #16522) Reported by: corruptor ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@238637 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	5bc3fe1baa	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 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@238499 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	21ddcd4709	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 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@237409 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	d05977c54f	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. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@236635 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	4f7c2bdf62	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) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@235665 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	98ecc431c9	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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@231696 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	6f122d97ae	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/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@231689 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago

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