asterisk

Commit Graph

Author	SHA1	Message	Date
Tilghman Lesher	3c3d054ef2	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/trunk@267775 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	a68f5b96bc	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. Review: https://reviewboard.asterisk.org/r/683/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267492 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	1c67f208a7	Add ETSI Message Waiting Indication (MWI) support. 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	15 years ago
Richard Mudgett	0760f4e70a	Add ETSI Malicious Call ID support. 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	15 years ago
Richard Mudgett	afcbc93dae	Add ETSI Call Waiting support. 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	15 years ago
Richard Mudgett	afd4454c44	Generic Advice of Charge. 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	15 years ago
Jeff Peeler	f9467341cd	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/trunk@267065 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	28264c52b9	Add ETSI Advice Of Charge (AOC) event reporting. 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	15 years ago
Richard Mudgett	48dd4d1249	Add ETSI Explicit Call Transfer (ECT) support. 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	15 years ago
Tilghman Lesher	b0357dcc3e	Support setting locale per-mailbox (changes date/time languages for email, 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	15 years ago
Terry Wilson	f1503b9e1d	Ensure that libneon > 0.29.0 is installed for res_calendar_ews 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	15 years ago
Tilghman Lesher	a7498ae02e	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/trunk@265747 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	1225ee831c	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/trunk@265090 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	f30952c577	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/trunk@265000 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 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	15 years ago
Mark Michelson	0a63e3fa10	Log spandsp's fax debug output to the FAX logger level. Review: https://reviewboard.asterisk.org/r/658 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264953 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 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	15 years ago
Tilghman Lesher	b5a629624a	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/trunk@264249 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	4c034c1f72	Cache sound tarfiles in a common directory, such that a clean reinstall does 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	15 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	15 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	15 years ago
Paul Belanger	4b1d9f85a7	Convert to AST_CLI_YESNO and AST_CLI_ONOFF 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	15 years ago
Richard Mudgett	9534f72cb0	Dialing an invalid extension causes incomplete hangup sequence. 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	15 years ago
Tilghman Lesher	2c10997e99	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/trunk@262513 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	92a8650677	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/trunk@262102 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	13f15cae67	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/trunk@261913 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 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	15 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	15 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	15 years ago
Richard Mudgett	6adcbd6be5	Fix comment. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260007 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	595245c0e0	Update the Mantis Workflow document in doxygen. (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	15 years ago
Matthew Nicholson	13f523731a	Update res_fax and res_fax_spandsp to be compatible with Fax For Asterisk 1.2. 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	15 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	15 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	15 years ago
Leif Madsen	8b11ae2e4f	Add ability to generate ASCII documentation from the TeX files. These changes add the ability to run 'make asterisk.txt' just like the existing 'make asterisk.pdf' commands to generate a text document from the TeX files we have in the doc/tex/ directory. I've also updated a few of the .tex files because they weren't properly escaping certain characters so they would show up as Unicode characters (like [U+021C]). Made changes to the configure scripts so it would detect the catdvi program which is required to convert the .dvi file generated by latex. I've also added a few lines to the build_tools/prep_tarball script so that the text documentation gets generated and added to future tarballs of Asterisk releases. (closes issue #17220) Reported by: lmadsen Patches: asterisk.txt.patch uploaded by lmadsen (license 10) asterisk.txt.patch-v4 uploaded by pabelanger (license 224) Tested by: lmadsen, pabelanger git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258351 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 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	15 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	15 years ago
Richard Mudgett	537edff10f	Remove PRI CCSS BUGBUG message and update configure script. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256569 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	e24661fd18	Merge Call completion support into trunk. From Reviewboard: CCSS stands for Call Completion Supplementary Services. An admittedly out-of-date overview of the architecture can be found in the file doc/CCSS_architecture.pdf in the CCSS branch. Off the top of my head, the big differences between what is implemented and what is in the document are as follows: 1. We did not end up modifying the Hangup application at all. 2. The document states that a single call completion monitor may be used across multiple calls to the same device. This proved to not be such a good idea when implementing protocol-specific monitors, and so we ended up using one monitor per-device per-call. 3. There are some configuration options which were conceived after the document was written. These are documented in the ccss.conf.sample that is on this review request. For some basic understanding of terminology used throughout this code, see the ccss.tex document that is on this review. This implements CCBS and CCNR in several flavors. First up is a "generic" implementation, which can work over any channel technology provided that the channel technology can accurately report device state. Call completion is requested using the dialplan application CallCompletionRequest and can be canceled using CallCompletionCancel. Device state subscriptions are used in order to monitor the state of called parties. Next, there is a SIP-specific implementation of call completion. This method uses the methods outlined in draft-ietf-bliss-call-completion-06 to implement call completion using SIP signaling. There are a few things to note here: * The agent/monitor terminology used throughout Asterisk sometimes is the reverse of what is defined in the referenced draft. * Implementation of the draft required support for SIP PUBLISH. I attempted to write this in a generic-enough fashion such that if someone were to want to write PUBLISH support for other event packages, such as dialog-state or presence, most of the effort would be in writing callbacks specific to the event package. * A subportion of supporting PUBLISH reception was that we had to implement a PIDF parser. The PIDF support added is a bit minimal. I first wrote a validation routine to ensure that the PIDF document is formatted properly. The rest of the PIDF reading is done in-line in the call-completion-specific PUBLISH-handling code. In other words, while there is PIDF support here, it is not in any state where it could easily be applied to other event packages as is. Finally, there are a variety of ISDN-related call completion protocols supported. These were written by Richard Mudgett, and as such I can't really say much about their implementation. There are notes in the CHANGES file that indicate the ISDN protocols over which call completion is supported. Review: https://reviewboard.asterisk.org/r/523 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256528 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	6cad0f1602	func_srv and explicit specification of a remote IP for SIP. From Review Board: There are two interrelated changes here. First, there is the introduction of func_srv. This adds two new read-only dialplan functions, SRVQUERY and SRVRESULT. They work very similarly to the ENUMQUERY and ENUMRESULT functions, except that this allows one to query SRV records instead. In order to facilitate this work, I added a couple of new API calls to srv.h. ast_srv_get_record_count tells the number of records returned by an SRV lookup. This number is calculated at the time of the SRV lookup. ast_srv_get_nth_record allows one to get a numbered SRV record. Second, there is the modification to chan_sip that allows one to specify a hostname or IP address (along with a port) to send an outgoing INVITE to when dialing a SIP peer. This goes hand-in-hand with func_srv. You can query SRV records and then use the host and port from the results to dial via a specific host instead of what is configured in sip.conf. Review: https://reviewboard.asterisk.org/r/608 SWP-1200 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256485 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	bcbafc800e	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/trunk@256370 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	a5a0a5f867	Consolidate ast_channel.cid.cid_rdnis into ast_channel.redirecting.from.number. SWP-1229 ABE-2161 * Ensure chan_local.c:local_call() will not leak cid.cid_dnid when copying. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256104 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	17a72827c8	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/trunk@255796 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Kevin P. Fleming	32938f653f	Remove no-longer-used (and unsafe) field in ast_channel for linked lists. The ast_channel structure had a field used for linking a channel into a linked list, but now that ast_channel structures are ao2 objects, this is no longer needed, and could be harmful as ao2 objects really shouldn't ever be placed into linked lists (since those lists don't assist with reference count management on the objects). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254637 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	a1e426cd2a	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/trunk@254553 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Kevin P. Fleming	42577406fd	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/trunk@254450 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Kevin P. Fleming	ae6008ef3a	Change per-file debug and verbose levels to be per-module, the way users expect them to work. 'core set debug' and 'core set verbose' can optionally change the level for a specific filename; however, this is actually for a specific source file name, not the module that source file is included in. With examples like chan_sip, chan_iax2, chan_misdn and others consisting of multiple source files, this will not lead to the behavior that users expect. If they want to set the debug level for chan_sip, they want it set for all of chan_sip, and not to have to also set it for reqresp_parser and other files that comprise the chan_sip module. This patch changes this functionality to be module-name based instead of file-name based. To make this work, some Makefile modifications were required to ensure that the AST_MODULE definition is present in each object file produced for each module as well. Review: https://reviewboard.asterisk.org/r/574/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253917 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	374e4e9d72	Fix bamboo compile error by calculating an integer with the same size as a pointer. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252980 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	385a40226a	Fix test_time on Mac OS X (and other platforms without inotify) Reviewboard: https://reviewboard.asterisk.org/r/554/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252846 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	68d1ded8dd	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/trunk@252089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago

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