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2007-12-05Rename "username" to "defaultuser" to match with "defaultip".oej1-5/+5
"Username" still works, but is deprecated. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91152 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-05Remove the cseqs from "sip show channel" and make more place for the call ID.oej1-5/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91151 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-05revert part of my changes from earlier today since this code is no longer ↵kpfleming1-4/+1
dependent on libpri.h git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91133 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-04Fix mwimonitornotify on reload ... again. This option was only read at startuprussell1-2/+2
so a reload would erase it and not reset it. (pointed out by tzafrir) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91069 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-04Don't error when we don't have libpri installed with libss7 support. Also, ↵mattf1-15/+9
print the debug message anyway if we can't find the right PRI git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91012 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-04Fix resetting mwimonitornotify on reload. I guess I only added this line in ↵russell1-0/+2
my head. (thanks to tzafrir for pointing it out) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91010 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-04Coding guidelines fixupstilghman1-805/+722
(Closes issue #11412) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90993 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-04Add manager action 'sipshowregistry'.qwell1-1/+56
Closes issue #11464, patch by eliel. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90991 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-04Add support for monitoring MWI on FXO lines.russell1-19/+121
This introduces two new options for zapata.conf: mwimonitor and mwimonitornotify. The mwimonitor option enables MWI monitoring. When the MWI state on a line changes, then the script specified by mwimonitornotify will be executed for custom handling of the state change, similar to the externnotify option of voicemail.conf. Also, when the MWI state on an FXO line changes, an internal Asterisk event is generated to indicate the new state of the associated mailbox. That may, any module that cares about MWI information will get notified and can handle it just as if app_voicemail had sent this notification. (BE-253, original patch from markster, with some minor modifications by me to add comments, documentation, and internal event support) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90949 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-04fix build of this module when libpri and/or libss7 are or are not presentkpfleming1-15/+22
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90880 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-04Merged revisions 90735 via svnmerge from mmichelson1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03 Dec 2007) | 22 lines A big one... This is the merge of the forward-loop branch. The main change here is that call-forwards can no longer loop. This is accomplished by creating a datastore on the calling channel which has a linked list of all devices dialed. If a forward happens, then the local channel which is created inherits the datastore. If, through this progression of forwards and datastore inheritance, a device is attempted to be dialed a second time, it will simply be skipped and a warning message will be printed to the CLI. After the dialing has been completed, the datastore is detached from the channel and destroyed. This change also introduces some side effects to the code which I shall enumerate here: 1. Datastore inheritance has been backported from trunk into 1.4 2. A large chunk of code has been removed from app_dial. This chunk is the section of code which handles the call forward case after the channel has been requested but before it has been called. This was removed because call-forwarding still works fine without it, it makes the code less error-prone should it need changing, and it made this set of changes much less painful to just have the forwarding handled in one place in each module. 3. Two new files, global_datastores.h and .c have been added. These are necessary since the datastore which is attached to the channel may be created and attached in either app_dial or app_queue, so they need a common place to find the datastore info. This approach was taken in case similar datastores are needed in the future, there will be a common place to add them. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90873 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-03Merged revisions 90639 via svnmerge from mmichelson1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90639 | mmichelson | 2007-12-03 14:59:51 -0600 (Mon, 03 Dec 2007) | 5 lines Changing some bad logic when calculating the interdigit timeout. (closes issue #11402, reported and patched by eferro) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90644 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-30Merged revisions 90348 via svnmerge from russell1-6/+6
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90348 | russell | 2007-11-30 13:26:04 -0600 (Fri, 30 Nov 2007) | 8 lines Change the behavior of ao2_link(). Previously, in inherited a reference. Now, it automatically increases the reference count to reflect the reference that is now held by the container. This was done to be more consistent with ao2_unlink(), which automatically releases the reference held by the container. It also makes it so it is no longer possible for a pointer to be invalid after ao2_link() returns. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90351 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-30Merged revisions 90269 via svnmerge from file1-3/+6
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90269 | file | 2007-11-30 10:43:15 -0400 (Fri, 30 Nov 2007) | 6 lines Fix locking issues under one legged replaces scenarios. (closes issue #11420) Reported by: irroot Patches: chan_sip_oneleg.patch uploaded by irroot (license 52) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90270 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-30Merged revisions 90231 via svnmerge from mmichelson1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90231 | mmichelson | 2007-11-29 18:16:04 -0600 (Thu, 29 Nov 2007) | 5 lines Clear the DTMF buffer if the call times out. (closes issue #11418, reported and patched by eferro) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90232 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27Merge changes from team/russell/iax2_frame_queuerussell1-68/+87
This patch is an optimization for chan_iax2. This module is now heavily multi-threaded. However, there is still a good number of globally shared resources that prevent things from happen asynchronously. One of those things was the global IAX frame queue. This queue was used to hold frames that have been deferred for transmitting by another thread, and frames that may need to get retransmitted. I changed the frame queue to be per-call, since almost all of the frame queue handling only cares about frames specific to a call number. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89887 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27Bring in a small change from team/russell/chan_refcountrussell1-23/+1
This replaces tab completion code with the use of a public function that does the same thing git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89835 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27More additions from the "moremanager" branch, this time for IAX2.oej1-9/+58
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89769 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27remove a duplicate manager eventrussell1-4/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89710 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27Manager events from the "moremanager" branchoej1-1/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89706 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27Starting to merge changes from the "moremanager" branch. Documentation willoej1-6/+33
follow. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89702 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27The following patch with updates for trunk. Works much better in trunk.oej1-8/+22
Also by accident fixed a bad typo by a previous committer, which actually made video calls not work fully... Merged revisions 89630 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89630 | oej | 2007-11-27 16:23:17 +0100 (Tis, 27 Nov 2007) | 12 lines If we get a codec offer using a well-known payload type, but using it for another codec that we don't know, Asterisk did not remove that codec from the list. With this patch, we remove the codec from audio and video rtp objects and deny it ever existed. Thanks to lasse for testing. (closes issue #11376) Reported by: lasse Patches: bug11376.txt uploaded by oej (license 306) Tested by: lasse ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89698 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26Rename "limitonpeer" to "counteronpeer" since the call-limit is deprecated.oej1-1/+1
Both still works in this version. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89613 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26Formatting, doxygenificationoej1-7/+10
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89611 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26Formatting changes, cleaning up some codeoej1-33/+45
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89609 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26Start using Doxygen groupings to group variables and defines.oej1-15/+39
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89607 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26Instead of printing out one codec in sip show channels print out all of the ↵file1-2/+3
native ones (this is for video). (closes issue #11366) Reported by: ovi git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89573 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-25Formatting changesoej1-93/+81
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89566 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-25Typo (someone needs to test compile before committing his changes)tilghman1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89560 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-25More doxygen changesoej1-3/+12
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89557 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-25Housekeepingoej1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89556 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-25Formatting, doxygen updatesoej1-30/+37
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89555 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-25- Deprecate "call-limit" in chan_sip. No other channel driver enforces ↵oej1-0/+16
call-limits and we now have the groupcount system to implement call-limits in the dialplan. You can use the "setvar" option in realtime/sip.conf to set limits per device. - Implement "callcounter" as a new option to enable the call counting we need to report device status to queue, manager and SIP subscriptions. The call counter setting is now enabled in the code by setting the device call-limit to 999. When we remove the call limit, we can simply enable this with a boolean setting. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89554 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-25Housekeeping...oej1-1/+1
- Fix typo in chan_sip - Remove changes to caller ID structure, moving it to branch (russellb) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89551 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-24remove a DEBUG_THREADS message that accesses private lock fields.rizzo1-3/+6
If needed, the code to extract this information should be implemented in some generic header or library and the function called here. (closed bug #11362) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89543 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-23put in the necessary hooks for video support in the console.rizzo1-2/+80
This is a NOP as far as the current code is concerned, but there is already support in ./configure and the Makefiles for the various libraries used by console_video.c (not yet in the tree) so addition is trivial. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89533 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-23set rtpmap video info according to what is read from SDP;rizzo1-2/+4
make the format explicit in a debug message; print the audio instead of aggregated peer capability in a debugging msg. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89532 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-21closes issue #11285, where an unload of a module that creates a dialplan ↵murf3-1/+19
context, causes a crash when you do a 'dialplan show' of that context. This is because the registrar string is defined in the module, and the stale pointer is traversed. The reporter offered a patch that would always strdup the registrar string, which is practical, but I preferred to destroy the created contexts in each module where one is created. That seemed more symmetric. There were only 6 place in asterisk where this is done: chan_sip, chan_iax2, chan_skinny, res_features, app_dial, and app_queue. The two apps destroyed the context, but left the contexts. All is fixed now and unloads should be dialplan friendly. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89513 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-21remove another set of redundant #include "asterisk/options.h"rizzo20-20/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89512 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-21Remove unneccessary explicit case for BRImattf1-3/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89510 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-21Take some debug code out :-)mattf1-1/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89509 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-21Add BRI support to chan_zapmattf1-25/+83
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89507 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-21fix a small gramatical error in a commentrussell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89488 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-21Fix some code that was supposed to ensure that a buffer was terminated, but wasrussell1-30/+7
writing to the wrong byte. Also, remove some non-thread safe test code. (closes issue #11317) Reported by: IgorG Patches: unistim-2.patch uploaded by IgorG (license 20) - additional changes by me git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89484 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-21get this to actually compile...kpfleming1-19/+24
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89481 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-21remove this file, it is not used anywhere.rizzo1-157/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89477 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-20move asterisk/paths.h outside asterisk.h and into those filesrizzo3-0/+3
who really need it. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89466 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-20Fix building of modules under cygwin.rizzo1-0/+4
After this commit we can actually load modules under windows, and we can start debugging more interesting problems related to the load order and functionality of modules. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89454 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-20Include the compatibility header file in ast_h323.cxx for compatibility reasons.file1-0/+1
(closes issue #11311) Reported by: falves11 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89447 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-20Fix sip show history.oej1-2/+2
Closes issue #11312 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89446 f38db490-d61c-443f-a65b-d21fe96a405b
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