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2010-06-25Implemement support for handling multiple documents when sending.Matthew Nicholson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272558 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-23Update configure when changing autconf m4 files...Terry Wilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272256 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-22Merged revisions 271689 via svnmerge from Matthew Nicholson
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/trunk@271690 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-18Merged revisions 271399 via svnmerge from Jeff Peeler
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/trunk@271483 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-17adds speex 16khz audio supportDavid Vossel
(closes issue #17501) Reported by: fabled Patches: asterisk-trunk-speex-wideband-v2.patch uploaded by fabled (license 448) Tested by: malcolmd, fabled, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271231 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-16addition of G.719 pass-through supportDavid Vossel
(closes issue #16293) Reported by: malcolmd Patches: g719.passthrough.patch.7 uploaded by malcolmd (license 924) format_g719.c uploaded by malcolmd (license 924) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270940 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-15Add distributed devicestate via the XMPP protocol.Tilghman Lesher
(closes issue #15757) Reported by: Marquis Patches: distributed_devstate-XMPP.txt uploaded by lmadsen (license 10) Tested by: Marquis, lmadsen, marcelloceschia Review: https://reviewboard.asterisk.org/r/351/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270519 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-13Reverting patch and reopening issue #16155, as patch breaksPaul Belanger
FreeBSD / OSX builds. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270151 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-11Use pkg-config to find gmime librariesPaul Belanger
This way the libraries can be found even if they are in non-standard locations. (closes issue #16155) Reported by: jcollie Patches: 0008-change-configure.ac-to-look-for-pkg-config-gmime-2.0.patch uploaded by jcollie (license 412) Tested by: jsmith, tilghman, pabelanger git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-09Resolve an invalid memory read on an event.Russell Bryant
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/trunk@269417 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-08Fix build on Mac OS X (and maybe FreeBSD, too)Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269119 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-08Fix some doxygen warnings.Leif Madsen
(closes issue #17336) Reported by: snuffy Patches: doxygen-fixes1.diff uploaded by snuffy (license 35) Tested by: russell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268969 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-08Add SRTP support for AsteriskTerry Wilson
After 5 years in mantis and over a year on reviewboard, SRTP support is finally being comitted. This includes generic CHANNEL dialplan functions that work for getting the status of whether a call has secure media or signaling as defined by the underlying channel technology and for setting whether or not a new channel being bridged to a calling channel should have secure signaling or media. See doc/tex/secure-calls.tex for examples. Original patch by mikma, updated for trunk and revised by me. (closes issue #5413) Reported by: mikma Tested by: twilson, notthematrix, hemanshurpatel Review: https://reviewboard.asterisk.org/r/191/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-07Seems strange (and the code backs up) that if the max and min of a statistic ↵Tilghman Lesher
is expressed as a double, the last value would not also need to be a double. (closes issue #15807) Reported by: klaus3000 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268773 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-04As signed linear audio data is accessed as 16-bit values, certain processors ↵Tilghman Lesher
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/trunk@267877 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-04As signed linear audio data is accessed as 16-bit values, certain processors ↵Tilghman Lesher
require the values to be aligned in memory. (closes issue #16912) Reported by: michaelevdokimov git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267862 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-04Merged revisions 267759 via svnmerge from Tilghman Lesher
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/trunk@267775 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-03Remove unnecessary code relating to PLC.Mark Michelson
The logic for handling generic PLC is now handled in ast_write in channel.c instead of in translation code. Review: https://reviewboard.asterisk.org/r/683/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267492 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-03Add ETSI Message Waiting Indication (MWI) support.Richard Mudgett
Add the ability to report waiting messages to ISDN endpoints (phones). Relevant specification: EN 300 650 and EN 300 745 Review: https://reviewboard.asterisk.org/r/599/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267399 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-02Add ETSI Malicious Call ID support.Richard Mudgett
Add the ability to report malicious callers as an AMI event in the call event class. Relevant specification: EN 300 180 Review: https://reviewboard.asterisk.org/r/576/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267350 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-02Add ETSI Call Waiting support.Richard Mudgett
Add the ability to announce a call to an endpoint when there are no B channels available. A call waiting call is a SETUP message with no B channel selected. Relevant specification: EN 300 056, EN 300 057, EN 300 058 For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the "no_media_path" option. * Returns "0" if there is a B channel associated with the call. * Returns "1" if no B channel is associated with the call. The call is either on hold or is a call waiting call. If you are going to allow incoming call waiting calls then you need to use CHANNEL(no_media_path) do determine if you must drop a call to accept the new call. Review: https://reviewboard.asterisk.org/r/568/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267261 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-02Generic Advice of Charge.Richard Mudgett
Asterisk Generic AOC Representation - Generic AOC encode/decode routines. (Generic AOC must be encoded to be passed on the wire in the AST_CONTROL_AOC frame) - AST_CONTROL_AOC frame type to represent generic encoded AOC data - Manager events for AOC-S, AOC-D, and AOC-E messages Asterisk App Support - app_dial AOC-S pass-through support on call setup - app_queue AOC-S pass-through support on call setup AOC Unit Tests - AOC Unit Tests for encode/decode routines - AOC Unit Test for manager event representation. SIP AOC Support - Pass-through of generic AOC-D and AOC-E messages to snom phones via the snom AOC specification. - Creation of chan_sip page3 flags for the addition of the new 'snom_aoc_enabled' sip.conf option. IAX AOC Support - Natively supports AOC pass-through through the use of the new AST_CONTROL_AOC frame type DAHDI AOC Support - ETSI PRI full AOC Pass-through support - 'aoc_enable' chan_dahdi.conf option for independently enabling pass-through of AOC-S, AOC-D, AOC-E. - 'aoce_delayhangup' option for retrieving AOC-E on disconnect. - DAHDI A() dial string option for requesting AOC services. example usage: ;requests AOC-S, AOC-D, and AOC-E on call setup exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e)) Review: https://reviewboard.asterisk.org/r/552/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267096 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-02Fix infinite loop when loading codec speexJeff Peeler
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/trunk@267065 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-02Add ETSI Advice Of Charge (AOC) event reporting.Richard Mudgett
This feature generates AMI events in the new aoc event class from the events passed up by libpri. Review: https://reviewboard.asterisk.org/r/537/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267008 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-02Add ETSI Explicit Call Transfer (ECT) support.Richard Mudgett
Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages to eliminate tromboned calls. Note: Asterisk already supported initiating the transfer of calls to eliminate tromboned calls to libpri so there was nothing to do for the asterisk portion. Review: https://reviewboard.asterisk.org/r/520/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266926 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-01Support setting locale per-mailbox (changes date/time languages for email, ↵Tilghman Lesher
pager messages). (closes issue #14333) Reported by: klaus3000 Patches: 20090515__issue14333.diff.txt uploaded by tilghman (license 14) app_voicemail.c-svn-trunk-rev211675-patch.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266828 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-26Ensure that libneon > 0.29.0 is installed for res_calendar_ewsTerry Wilson
This uses a modified version of pabelanger's patch that checks for NTLM support instead, which was added in 0.29.0 which is what is required for res_calendar_ews. (closes issue #17391) Reported by: loloski Patches: issue17391.patch.v2 uploaded by pabelanger (license 224) Tested by: twilson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265793 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-26Use configure to determine the prefixes and include directories properly.Tilghman Lesher
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/trunk@265747 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-21Merged revisions 265089 via svnmerge from Mark Michelson
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/trunk@265090 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-21Merged revisions 264999 via svnmerge from Mark Michelson
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/trunk@265000 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-21Merged revisions 264996 via svnmerge from Mark Michelson
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
2010-05-21Log spandsp's fax debug output to the FAX logger level.Mark Michelson
Review: https://reviewboard.asterisk.org/r/658 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264953 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-19Fix transcode_via_sln option with SIP calls and improve PLC usage.Mark Michelson
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
2010-05-19Merged revisions 264248 via svnmerge from Tilghman Lesher
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/trunk@264249 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17Cache sound tarfiles in a common directory, such that a clean reinstall does ↵Tilghman Lesher
not force a re-download of the tarballs. (closes issue #15370) Reported by: pprindeville Patches: asterisk-trunk-bugid15370.patch uploaded by pprindeville (license 347) Tested by: pprindeville, tilghman, seanbright git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263724 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17Enhancements to connected line and redirecting work.Mark Michelson
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
2010-05-13Add kqueue(2) implementation to Asterisk in various places.Tilghman Lesher
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
2010-05-12Convert to AST_CLI_YESNO and AST_CLI_ONOFFPaul Belanger
Clean up chan_sip.c to use new AST_CLI functions (closes issue #17287) Reported by: pabelanger Patches: issue17287.patch uploaded by pabelanger (license 224) Tested by: russell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-11Dialing an invalid extension causes incomplete hangup sequence.Richard Mudgett
Revision -r1489 of the libpri 1.4 branch corrected a deviation from Q.931 Section 5.3.2. However, this resulted in an unexpected behaviour change to the upper layer (Asterisk). This change uses pri_hangup_fix_enable() to follow Q.931 Section 5.3.2 call hangup better if the version of libpri supports it. (issue #17104) Reported by: shawkris Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262569 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-11Move cause 200 to cause 26, as specified in Q.850.Tilghman Lesher
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/trunk@262513 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-09Cleanup a bit more by getting rid of useless version defines. Also make ↵Tilghman Lesher
library detection use passed CFLAGS. (closes issue #17309) Reported by: stuarth git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262102 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-07Use the detected pthread building flags in every place, instead of ↵Tilghman Lesher
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/trunk@261913 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-05New 'manager show settings' CLI command.Paul Belanger
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
2010-05-02Avoid making AstData depend on libxml2 to compile.Eliel C. Sardanons
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
2010-04-29Merged revisions 260049 via svnmerge from David Vossel
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
2010-04-29Fix comment.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260007 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-27Update the Mantis Workflow document in doxygen.Leif Madsen
(closes issue #17175) Reported by: lmadsen Patches: Bug_Tracker_Workflow.v2.txt uploaded by pabelanger (license 224) Tested by: pabelanger, lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259438 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-26Update res_fax and res_fax_spandsp to be compatible with Fax For Asterisk 1.2.Matthew Nicholson
The fax session initilization code for T.38 faxes has been rewritten. T.38 session initialization was removed from generic_fax_exec, and split into two different code paths for receive and send. Also the 'z' option (to send a T.38 reinvite if we do not receive one) was added to sendfax. In the output of 'fax show sessions', the 'Type' column has been renamed to 'Tech' and replaced with a new 'Tech' column that will report 'G.711' or 'T.38'. Control of ECM defaults has been added to res_fax A 'fax show settings' CLI command has been added. Support of the new AST_T38_REQUEST_PARMS control method request to handle channels that have already received a T.38 reinvite before the FAX application is start has been added. Support for the 'fax show settings' command has been added to res_fax_spandsp and handling of the ECM flag has been slightly altered. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258896 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-22Remove ABI differences that occured when compiling with DEBUG_THREADS.Jason Parker
"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
2010-04-22Asterisk data retrieval API.Eliel C. Sardanons
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
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