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2008-01-10The fixes in this commit are mainly to allow compiling of trunk with ↵murf3-0/+3
--enable-dev-mode, mutex profiling, lock debugging, etc. Mainly, the version.c needs to be in the OBJS line; asterisk.h was chosen to have the prototypes for ast_get_version, ast_get_version_num; and the ASTERISK_FILE_VERSION macro needs to be used after including asterisk.h in a few files. I hope I did the right thing. If not, let me know. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97656 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-10Several manager changes:tilghman2-6/+6
1) Add the Dialplan class, for NewExten and VarSet events, which should cut down on the volume of traffic in the Call class. 2) Permit some commands to be run from multiple classes, such as allowing DBGet to be run from either the System or the Reporting class. 3) Heavily document each class in the sample config, as there were several that made no sense to be in the write= line, and two that made no sense to be in the read= line (since they controlled no permissions there). (Closes issue #10386) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97651 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-09remove a wrong 'const'rizzo1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97533 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-09Two changes:rizzo3-94/+120
- support scrolling of message window; - simplify the code for creating a message window, and try it using a second one in the top of the keypad (where we echo the dialed number). The 'skin' that supports these two windows will be committed separately. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97530 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-09Merged revisions 97489 via svnmerge from phsultan1-15/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97489 | phsultan | 2008-01-09 17:44:24 +0100 (Wed, 09 Jan 2008) | 7 lines Set the caller id within the gtalk_alloc function. As underlined in issue #10437 by Josh, we need to prevent a possible memory leak. We only set the name part of the caller id, the number part is not relevant when dealing with JIDs. Closes issue #11549. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97490 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-09Implement keyboard handling, and use it to enterrizzo4-67/+102
a number to dial in the 'message' area under the keypad. Now you can make calls using the keypad as a regular phone (or the keyboard for chars not present on the keypad) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97488 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-09Merged revisions 97448 via svnmerge from kpfleming1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97448 | kpfleming | 2008-01-09 09:43:19 -0600 (Wed, 09 Jan 2008) | 2 lines pass the right variable to get an error string... oops ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97449 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-09Merged revisions 97410 via svnmerge from kpfleming1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97410 | kpfleming | 2008-01-09 09:26:23 -0600 (Wed, 09 Jan 2008) | 2 lines add error number output to ioctl failure messages to help with debugging ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97421 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-09implement the "console startgui" and "console stopgui"rizzo2-21/+42
commands so you can start and stop the gui even outside of a call. This is convenient for testing, and also for using the keypad to pick up a call, and to dial a number (the latter not yet implemented, but should be close). git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97390 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-09make get_video_desc() return the active console ifrizzo1-2/+3
passed a null argument (channel). git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97389 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-08add copyright (most of this code was written by Marta Carbone),rizzo1-49/+35
remove some unused code, add/clarify some comments. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97303 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-08add support for textareas, used for various dialog windows on the gui.rizzo4-58/+402
The main code to implement the textarea is in console_board.c, and uses a simple png image with the font, blitting characters on the designated areas of the main screen. Additionally we provide some annotations in the image used as a skin to indicate which areas are used for text messages. (images will be committed separately). At the moment the dialog area is only used to display a running counter, just as a proof of concept. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97280 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-08Change reference to external library so it appears on the extref listingoej1-1/+1
http://www.asterisk.org/doxygen/trunk/extref.html git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97200 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-08One line documentation ftw!file1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97197 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-08Merged revisions 97195 via svnmerge from file1-11/+49
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97195 | file | 2008-01-08 16:48:20 -0400 (Tue, 08 Jan 2008) | 6 lines Fix various DTMF issues in chan_mgcp. (closes issue #11443) Reported by: eferro Patches: dtmf_control_hybrid-inband-mode.patch uploaded by eferro (license 337) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97196 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-08Move common code for setting T38 capabilities and fix a bug with fax ↵file1-19/+18
detection in the SIP RTP read callback. It's still sort of silly... but more on that later. (closes issue #11239) Reported by: dimas Patches: sipt38prop.patch uploaded by dimas (license 88) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97154 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-08Merged revisions 97077 via svnmerge from tilghman1-80/+143
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97077 | tilghman | 2008-01-08 12:02:13 -0600 (Tue, 08 Jan 2008) | 3 lines Apply multiple crash fixes, found in issue #11386, but not completely closing that issue. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97125 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-07add support for cropping the keypad image while displaying it.rizzo1-34/+65
This way it can contain additional elements (e.g. fonts, buttons, widgets) without having to use a zillion files to store them. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96988 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-07update comments to reflect reality (or at least planned behaviour).rizzo1-81/+84
minor code cleanups git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96836 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-07resolve a load-time problem avoiding a call to console_do_answer.rizzo1-2/+9
On passing, fix dialling from the keypad. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96776 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-05Now that the version.h file was getting properly regenerated every time the svnrussell1-2/+2
revision changed, every module that used the version was getting rebuilt after every svn update. This severly annoyed me pretty quickly, so I have improved the situation. Now, instead of generating version.h, main/version.c is generated. version.c includes the version information, as well as a couple of API calls for modules to retrieve the version. So now, only version.c will get rebuilt, and the main asterisk binary relinked, which is must faster than rebuilding http.c, manager.c, asterisk.c, relinking the asterisk binary, chan_sip.c, func_version.c, res_agi ... The only minor change in behavior here is that the version information reported by chan_sip, for example, is the version of the Asterisk core, and not necessarily the Asterisk version that the chan_sip module came from. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96717 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-05Add the URL to the home page for portaudio. Also add the location of therussell1-0/+5
svn repository to check out portaudio v19. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96692 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-05improve chan_usbradio to use indications just like chan_alsa/chan_oss do nowkpfleming1-249/+34
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96621 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-04Merged revisions 96525 via svnmerge from tilghman1-0/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r96525 | tilghman | 2008-01-04 13:27:25 -0600 (Fri, 04 Jan 2008) | 4 lines If you change the bindaddr in sip.conf to a non-bound address and reload, sip goes kablooie. Reported and patched by: one47 (Closes issue #11535) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96547 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-04[commit message]kpfleming1-5/+22
(closes issue #10393) Reported by: tzafrir Patches: chan_alarm_asterisk.diff uploaded by tzafrir (license 46) (modified by me and added configure script support) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96500 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-04Merged revisions 96449 via svnmerge from russell1-5/+5
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r96449 | russell | 2008-01-04 10:19:22 -0600 (Fri, 04 Jan 2008) | 7 lines Make use of the temporary channel pointer while the pvt is unlocked. (closes issue #11675) Reported by: flefoll Patches: chan_zap.c.patch-store-owner-before-unlock uploaded by flefoll (license 244) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96450 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-03Merged revisions 96394 via svnmerge from russell1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r96394 | russell | 2008-01-03 16:44:22 -0600 (Thu, 03 Jan 2008) | 3 lines Don't crash if the iax2 pvt structure has been destroyed before we get to this point (closes issue #11672, reported by snuffy, patched by me) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96395 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-03eliminiate sound_thread() and other stuff from chan_oss since Asterisk ↵kpfleming7-2427/+14
indications can handle it remove gentone and all the headers containing tones that are no longer needed git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96270 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-03coding guidelines cleanupkpfleming1-312/+51
remove background thread and all sound generation mechanisms, as the built-in indications can handle everything that is needed git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96245 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-03Merged revisions 96198 via svnmerge from crichter1-1/+8
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r96198 | crichter | 2008-01-03 13:08:40 +0100 (Do, 03 Jan 2008) | 1 line when overlapdial was used and no number was dialed, the call was dropped, now we just jump into the s extension, which makes a lot more sense. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96221 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-02Add support for generating a ringing sound on an incoming call. This is a bitrussell1-5/+8
of a hack. It just asks the core to generate the same tone that it would when you hear ringback when making an outbound call. But hey, it works, and you get the localized ring tone for the appropriate language set on the channel. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96079 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-02Note that this module doesn't actually play a ringing sound for an incoming callrussell1-0/+1
... oops git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96077 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-02Show the correct CLI command to answer the callrussell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96076 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-02actually parse and store echocan parameters from zapata.conf... this ↵kpfleming1-7/+59
*should* work <G> git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96073 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-02Add curly braces around a compound if statement so that trunk will build ↵mmichelson1-1/+2
properly git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96028 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-02another checkpoint... chan_zap can now use the new ZT_ECHOCAN_PARAMS ioctl ↵kpfleming1-20/+68
if it is present, but doesn't parse any supplied parameters yet (this implementation is not very memory efficient as the parameters and their values will be duplicated for each channel that has the same settings, but we can worry about that later once it is working) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96019 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-02Change instances of AST_NONSTANDARD_APP_ARGS(foo, bar, ',') to ↵mmichelson1-1/+1
AST_STANDARD_APP_ARGS(foo, bar) (closes issue #11668, reported and patched by mvanbaak) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95994 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-02Merged revisions 95946 via svnmerge from file1-1/+6
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r95946 | file | 2008-01-02 16:24:09 -0400 (Wed, 02 Jan 2008) | 4 lines Allocate a SIP refer structure when performing a transfer using BYE with Also so that the transfer information is properly stored. (AST-2008-001) (closes issue #11637) Reported by: greyvoip ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95947 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-02clean up hwgain CLI command and improve docs for swgain CLI commandkpfleming1-11/+9
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95939 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-02umm... this did not compile on x86-64, and could not possibly have worked on ↵kpfleming1-2/+4
any platform as it was passing string pointers to a function expecting ints git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95893 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-02fix some long-time breakage that kept chan_misdn from being embeddedkpfleming1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95841 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-02use the proper technique for including submodules so that embedding will workkpfleming1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95840 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-01call directly the cli command to implement hangup.rizzo1-2/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95673 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-01prevent a panic when destroying a channel with no incoming video.rizzo1-1/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95672 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-01remove a leftover sleep(1) used for debuggingrizzo1-2/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95671 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-31Update chan_console to natively use a 16 kHz sample rate. If it is talkingrussell1-20/+12
to an 8 kHz endpoint, then codec_resample will automatically be used to properly resample the audio before sending it to/from chan_console. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95527 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-31Merge the main set of changes from team/russell/chan_console.russell1-0/+1094
Add a new console channel driver, chan_console, which is a console channel driver that uses portaudio as a cross platform audio interface. It was written to provide a console channel driver that works with Mac CoreAudio, but it supports a number of other audio interfaces, as well, including OSS and ALSA. It could one day be the single console channel driver, but does not yet have as many features as chan_oss. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95412 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-29Move grabbers definitions to a separate file, vgrabbers.c, so it is easierrizzo5-300/+401
to add more entries. This required moving struct grab_desc to the common header, and adding an entry in the Makefile. On passing, cleanup some comments and file headers (some are still missing). git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95313 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-29virtualize the interface for video grabbers, which shouldrizzo2-147/+240
make it easier to add support for more grabbers (V4L2, firewire, and so on). git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95288 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-28Add a few entries up to 1408x1152 in the table of known video resolutions.rizzo1-0/+3
This makes it very convenient to enlarge images using the right-click on the video window. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95264 f38db490-d61c-443f-a65b-d21fe96a405b
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