Age | Commit message (Collapse) | Author | Files | Lines |
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Reported by: eliel
Patches:
res_features.c.patch uploaded by eliel (license 64)
res_agi.c.patch uploaded by seanbright (license 71)
res_musiconhold.c.patch uploaded by seanbright (license 71)
pbx.c.patch uploaded by moy (license 222)
logger.c.patch uploaded by moy (license 222)
frame.c.patch uploaded by moy (license 222)
manager.c.patch uploaded by moy (license 222)
http.c.patch uploaded by moy (license 222)
dnsmgr.c.patch uploaded by moy (license 222)
res_realtime.c.patch uploaded by eliel (license 64)
res_odbc.c.patch uploaded by seanbright (license 71)
res_jabber.c.patch uploaded by eliel (license 64)
chan_local.c.patch uploaded by eliel (license 64)
chan_agent.c.patch uploaded by eliel (license 64)
chan_alsa.c.patch uploaded by eliel (license 64)
chan_features.c.patch uploaded by eliel (license 64)
chan_sip.c.patch uploaded by eliel (license 64)
RollUp.1.patch (includes all of the above patches) uploaded by seanbright (license 71)
Convert many CLI commands to the NEW_CLI format.
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channel lock wrappers
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https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #10485)
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r79902 | qwell | 2007-08-17 12:44:22 -0500 (Fri, 17 Aug 2007) | 4 lines
Re-add the setting of callerid name and number.
Issue 10485, reported by and fix explained by paradise.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r79174 | file | 2007-08-13 11:18:04 -0300 (Mon, 13 Aug 2007) | 4 lines
(closes issue #10437)
Reported by: haklin
Don't set the callerid name and number a second time on a newly created channel. ast_channel_alloc itself already sets it and setting it twice would cause a memory leak.
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the ast_check_hangup() funciton. This function takes scheduled hangups into
account.
(closes issue #10230, patch by Juggie)
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applications
using old methods of parsing arguments to using the standard macros. However, the big
change is that the really old way of specifying application and arguments separated by
a comma will no longer work (e.g. NoOp,foo|bar). Instead, the way that has been
recommended since long before 1.0 will become the only method available (e.g. NoOp(foo,bar).
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universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
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run on peer. Had to upgrade ast_flag stuff to 64 bits to do this.
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phones.
If the call is answered by another phone, other phones won't display the call as "missed".
You can also add an option to the dial command so that you can have a "followme"
scenario and not count the calls as "missed" when you cancel the call.
Thanks to Ramon and Frank for feedback on this feature.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r73319 | file | 2007-07-05 10:27:40 -0300 (Thu, 05 Jul 2007) | 10 lines
Merged revisions 73318 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r73318 | file | 2007-07-05 10:26:02 -0300 (Thu, 05 Jul 2007) | 2 lines
Actually check to make sure a PBX was started on one of the Local channels instead of blindly assuming it was. (issue #10112 reported by makoto)
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(issue #9957, patches from mvanbaak, caio1982, critch, and dimas)
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guidelines changes
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r64193 | murf | 2007-05-14 07:58:42 -0600 (Mon, 14 May 2007) | 1 line
As per 9570, worrisome CDR warnings have been removed, that are either not helpful, or not relevant.
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NOT_INUSE or INUSE when Local channels are in use as opposed to just UNKNOWN.
It will still return INVALID if the extension doesn't exist at all.
(issue #8048, patch from tim_ringenbach)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r60989 | murf | 2007-04-09 12:32:07 -0600 (Mon, 09 Apr 2007) | 1 line
This is a big improvement over the current CDR fixes. It may still need refinement, but this won't have as many folks bothered.
This also adds the mods from 1.4/r.61136;
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r60847 | tilghman | 2007-04-08 21:42:48 -0500 (Sun, 08 Apr 2007) | 10 lines
Merged revisions 60846 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r60846 | tilghman | 2007-04-08 21:37:18 -0500 (Sun, 08 Apr 2007) | 2 lines
Bug 9505 - If the return value for local_queue_frame is set, then p->lock is no longer valid.
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r57318 | file | 2007-03-01 17:21:44 -0500 (Thu, 01 Mar 2007) | 10 lines
Merged revisions 57317 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r57317 | file | 2007-03-01 17:19:32 -0500 (Thu, 01 Mar 2007) | 2 lines
Don't even attempt to optimize things when a proxy channel is involved. It will just explode in weird and unexplaineable ways. (issue #9175 reported by clegall_proformatique)
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r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines
Merge the changes from the /team/group/vldtmf_fixup branch.
The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged. So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio. However,
since there was no audio coming in, the DTMF_END was never generated. This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf). If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback. This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597, maybe others...)
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r47751 | file | 2006-11-16 13:29:12 -0500 (Thu, 16 Nov 2006) | 10 lines
Merged revisions 47750 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r47750 | file | 2006-11-16 13:26:50 -0500 (Thu, 16 Nov 2006) | 2 lines
Because of the way chan_local is written we should be extra careful and make sure our callback functions have a tech_pvt. (issue #8275 reported by mflorell)
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r47712 | file | 2006-11-15 17:31:17 -0500 (Wed, 15 Nov 2006) | 10 lines
Merged revisions 47711 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r47711 | file | 2006-11-15 17:29:30 -0500 (Wed, 15 Nov 2006) | 2 lines
Make sure that the pvt structure exists before trying to do fixup on Local channels. (issue #7937 reported by mada123, fix by alamantia with mods by me)
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r47287 | file | 2006-11-07 15:14:58 -0500 (Tue, 07 Nov 2006) | 2 lines
This is not the commit you are looking for...
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r47284 | file | 2006-11-07 15:08:52 -0500 (Tue, 07 Nov 2006) | 2 lines
Make MOH work as it did before in chan_local, without this then it can go funky when transfers and MOH are involved. (issue #7671 reported by jmls)
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r47051 | tilghman | 2006-11-02 17:00:20 -0600 (Thu, 02 Nov 2006) | 2 lines
Reverse change of "show" to "list" and make several other commands more consistent with "category verb arguments"
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r43697 | file | 2006-09-26 16:19:33 -0400 (Tue, 26 Sep 2006) | 2 lines
Strip options off the argument passed for devicestate in chan_local. (issue #8034 reported by pcardozo)
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Russell Bryant and the RTP portion done by myself, Muffinlicious Joshua Colp. This has gone through so many discussions/revisions it's not funny but we finally have it!
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There are some situations in Asterisk where ast_frame and/or iax_frame
structures are rapidly allocatted and freed (at least 50 times per second
for one call).
This code significantly improves the performance of ast_frame_header_new(),
ast_frdup(), ast_frfree(), iax_frame_new(), and iax_frame_free() by keeping
a thread-local cache of these structures and using frames from the cache
whenever possible instead of calling malloc/free every time.
This commit also converts the ast_frame and iax_frame structures to use the
linked list macros.
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- restructured build tree and makefiles to eliminate recursion problems
- support for embedded modules
- support for static builds
- simpler cross-compilation support
- simpler module/loader interface (no exported symbols)
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a Local channel that's been improperly defined.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r33638 | kpfleming | 2006-06-12 11:03:29 -0500 (Mon, 12 Jun 2006) | 2 lines
only allow chan_local to masquerade the outbound channel onto its owner, instead of the other way around (this will ensure that group variables on the outbound channel as preserved)
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again :-)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r29464 | file | 2006-05-22 13:33:03 -0300 (Mon, 22 May 2006) | 2 lines
Preserve presentation bit when going through chan_local (issue #7002 reported by acunningham)
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->indicate()
update iax2_indicate to pass control frame payload to the connected channel
add an API call for sending an indication with payload, and use it for control frames with payload
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As partly documented in loader.c and include/asterisk/module.h,
modules are now expected to return all of their methods and flags
into a structure 'mod_data', and are normally loaded with RTLD_NOW
| RTLD_LOCAL, so symbols are resolved immediately and conflicts
should be less likely. Only in a small number of cases (res_*,
typically) modules are loaded RTLD_GLOBAL, so they can export
symbols.
The core of the change is only the two files loader.c and
include/asterisk/module.h, all the rest is simply adaptation of the
existing modules to the new API, a rather mechanical (but believe
me, time and finger-consuming!) process whose detail you can figure
out by svn diff'ing any single module.
Expect some minor compilation issue after this change, please
report it on mantis http://bugs.digium.com/view.php?id=6968
so we collect all the feedback in one place.
I am just sorry that this change missed SVN version number 20000!
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description() and key() return values
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rand() to threadsafe ast_random()
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CHANLOCALSTATUS on has gone away already.
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const-ify some more APIs
remove 'type' field from ast_channel, in favor of the one in the channel's tech structure
allow string field module users to specify the 'chunk size' for pool allocations
update chan_alsa to be compatible with recent const-ification patches
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