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2210 Commits (a39b2f5ed2be5df7d5d157374080cb92f6dfb5d8)
Author | SHA1 | Message | Date |
---|---|---|---|
|
ecabd15422 |
Addition of the FrameHook API (AKA AwesomeHooks)
So far all our tools for viewing and manipulating media streams within Asterisk have been entirely focused on audio. That made sense then, but is not scalable now. The FrameHook API lets us tap into and manipulate _ANY_ type of media or signaling passed on a channel present today or in the future. This tool is a step in the direction of expanding Asterisk's boundaries and will help generate some rather interesting applications in the future. In addition to the FrameHook API, a simple dialplan function exercising the api has been included as well. This function is called FRAME_TRACE(). FRAME_TRACE() allows for the internal ast_frames read and written to a channel to be output. Filters can be placed on this function to debug only certain types of frames. This function could be thought of as an internal way of doing ast_frame packet captures. Review: https://reviewboard.asterisk.org/r/925/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@287647 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
c9bfde6afd |
Add parking extension for non-default parking lots.
This is a new feature that allows for parking to custom parking lots to be accessed directly, rather than with channel variables or by changing the default parking lot. The extension is set with the parkext option just as the default parking lot is done. Also, the manager action has been updated to optionally allow a specified parking lot. (closes issue #14882) Reported by: vmikhnevych Patches: patch_14882.txt uploaded by mnick (license 874) modified by me Review: https://reviewboard.asterisk.org/r/884/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@286931 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
e72b55f3cf |
Merged revisions 286115 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r286115 | twilson | 2010-09-10 15:35:25 -0500 (Fri, 10 Sep 2010) | 23 lines Merged revisions 286059 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286059 | twilson | 2010-09-10 14:25:08 -0500 (Fri, 10 Sep 2010) | 16 lines Inherit CHANNEL() writes to both sides of a Local channel Having Local (/n) channels as queue members and setting the language in the extension with Set(CHANNEL(language)=fr) sets the language on the Local/...,2 channel. Hold time report playbacks happen on the Local/...,1 channel and therefor do not play in the specified language. This patch modifies func_channel_write to call the setoption callback and pass the CHANNEL() write info to the callback. chan_local uses this information to look up the other side of the channel and apply the same changes to it. (closes issue #17673) Reported by: Guggemand Review: https://reviewboard.asterisk.org/r/903/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@286189 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
6b87845350 |
Merged revisions 285961 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r285961 | tilghman | 2010-09-10 00:31:31 -0500 (Fri, 10 Sep 2010) | 6 lines Another fix for Mac OS X. While trying to fix this the "right" way, I wandered into dependency hell. Two hours later, I backed out, and just removed the offending code. ast_inline_api only goes one level deep and then it breaks. Ouch. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@285962 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
f7c710feed |
Merged revisions 285930 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r285930 | tilghman | 2010-09-09 20:16:32 -0500 (Thu, 09 Sep 2010) | 14 lines Merged revisions 285889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285889 | tilghman | 2010-09-09 19:13:45 -0500 (Thu, 09 Sep 2010) | 7 lines Fix Mac OS X build. This also fixes a rather grievous calculation error for the offset of ast_fdset, which was masked on Linux and FreeBSD, because these platforms check the first 256 FDs regardless of the bitmask setting (due to backwards compatibility). ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@285931 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
a766133be6 |
Merged revisions 285529 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r285529 | qwell | 2010-09-08 15:42:44 -0500 (Wed, 08 Sep 2010) | 1 line Follow coding guidelines in moh rescan fix. Also fix the documentation that got me in trouble. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@285530 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
f7f7eaa3a0 |
Fix build on FreeBSD 8.0, take 2.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@285336 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
2fc3cf65af |
Fix libsrtp -fPIC check for when non-standard prefix is used.
Thanks to loompek in #asterisk for reporting the issue and testing this patch. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@285161 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
222d4676d1 |
Add a C++ compatible version of AST_CLI_DEFINE().
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@285057 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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e7603921cf |
Support for calendar events priorities and categories
Review 880 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284849 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
7e3f95e00a |
When optional_api is non-optional, force dependent modules to be loaded.
(closes issue #17707) Reported by: ira Patches: 20100819__issue17707__asterisk1.8.diff.txt uploaded by tilghman (license 14) Tested by: tilghman Review: https://reviewboard.asterisk.org/r/876/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284610 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
6c61e312c6 |
Merged revisions 284593,284595 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r284593 | tilghman | 2010-09-01 17:59:50 -0500 (Wed, 01 Sep 2010) | 18 lines Merged revisions 284478 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r284478 | tilghman | 2010-09-01 13:49:11 -0500 (Wed, 01 Sep 2010) | 11 lines Ensure that all areas that previously used select(2) now use poll(2), with implementations that need poll(2) implemented with select(2) safe against 1024-bit overflows. This is a followup to the fix for the pthread timer in 1.6.2 and beyond, fixing a potential crash bug in all supported releases. (closes issue #17678) Reported by: russell Branch: https://origsvn.digium.com/svn/asterisk/team/tilghman/ast_select Review: https://reviewboard.asterisk.org/r/824/ ........ ................ r284595 | tilghman | 2010-09-01 22:57:43 -0500 (Wed, 01 Sep 2010) | 2 lines Failed to rerun bootstrap.sh after last commit ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284597 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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8a112de270 |
Fix SRTP for changing SSRC and multiple a=crypto SDP lines
Adding code to Asterisk that changed the SSRC during bridges and masquerades broke SRTP functionality. Also broken was handling the situation where an incoming INVITE had more than one crypto offer. This patch caches the SRTP policies the we use so that we can change the ssrc and inform libsrtp of the new streams. It also uses the first acceptable a=crypto line from the incoming INVITE. (closes issue #17563) Reported by: Alexcr Patches: srtp.diff uploaded by twilson (license 396) Tested by: twilson Review: https://reviewboard.asterisk.org/r/878/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284477 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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b246f8691e |
Make the AST_CEL_AMA enum match up with the AST_CDR_ ama flag values.
Really, having 2 enums for this is silly and error prone, demonstrated by the crash that I hit because there was an assumption in the code that the values in each matched up. However, this is a quick fix to get them to match up so it will work. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@283230 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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9fe871150e |
fixes truncated uint64_t value in put_unaligned_uint64_t() function
(issue #17804) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@282543 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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3c0616589e |
Fix our FRACKing issue with chan_iax2 a different way.
Review: https://reviewboard.asterisk.org/r/861/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@282366 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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48fb2c3276 |
res_stun_monitor for monitoring network changes behind a NAT device
Review: https://reviewboard.asterisk.org/r/854 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@282269 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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89872aac58 |
Separate call completion config parameter allocation and default initialization.
If you ever have a need to reset the call completion config parameters to defaults, now you can. And no Virginia, C++ idioms do not always work in C. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@282098 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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44bc8cd334 |
improved translation paths for wideband codecs
The problem I'm addressing is that Asterisk's current method of building the least cost translation paths between codecs does not take into account sample rate. For instance, it was possible for siren14 (a 32khz codec), to contain the a translation path to siren7 (a 16khz audio codec) that goes through slin at 8khz. In this case Asterisk takes a 32khz codec, down samples it to 8khz and then up samples it to 16khz which is terrible regardless if it is computationally less expensive. This patch now builds translation paths that give priority to maintaining the best possible sample rate before taking into consideration computational cost. This patch also adds cli commands to expose what translation paths are actually being used. Changes: 1. Translation paths will never contain a step that changes the sample rate unless absolutely necessary. 2. When choosing the best codec to make two channels compatible. Shared codecs with the highest sample rate are given priority. 3. A new cli command to show all translation paths available for a specific codec 'core show translation paths [codec name]' has been added. 4. 'core show translation' which displays the translation matrix now includes the new higher bit audio codecs in the table. 5. 'core show channel [channel name]' now displays the translation paths if translation is used. (closes issue #16841) Reported by: dvossel Review: https://reviewboard.asterisk.org/r/842/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@282047 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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5c1c1b35bd |
Fix parsing of IPv6 address literals in outboundproxy
(closes issue #17757) Reported by: oej Patches: 17757.diff uploaded by sperreault (license 252) sip.conf.diff uploaded by sperreault (license 252) Tested by: oej git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@281687 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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20582d0a6d |
Merged revisions 280983 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r280983 | tilghman | 2010-08-05 02:40:47 -0500 (Thu, 05 Aug 2010) | 15 lines Merged revisions 280982 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r280982 | tilghman | 2010-08-05 02:28:33 -0500 (Thu, 05 Aug 2010) | 8 lines Change context lock back to a mutex, because functionality depends upon the lock being recursive. (closes issue #17643) Reported by: zerohalo Patches: 20100726__issue17643.diff.txt uploaded by tilghman (license 14) Tested by: zerohalo ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@280984 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
0140af581e |
Revert r280019 for now - This was poorly executed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@280020 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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5fb5c90b89 |
Add ability to use system libedit and update bundled libedit.
The version of libedit that is bundled with asterisk is old and has some bugs. This patch updates the bundled version of libedit within asterisk, and also updates asterisk to use the system libedit instead if one is available (and pkg-config is available). This review integrates several patches from other users specifically kkm and tzafrir. (closes issue #15929) Reported by: kkm Patches: 015929-astcli-editrc-trunk.240324.diff uploaded by kkm (license 888) (issue #16858) Reported by: jw-asterisk (closes issue #17039) Reported by: tzafrir Patches: 0001-allow-using-system-copy-of-libedit.patch uploaded by tzafrir (license 46) Review: https://reviewboard.asterisk.org/r/807/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@280019 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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df89f8da6a |
Merged revisions 279946 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r279946 | dvossel | 2010-07-27 15:54:32 -0500 (Tue, 27 Jul 2010) | 24 lines Merged revisions 279945 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r279945 | dvossel | 2010-07-27 15:33:40 -0500 (Tue, 27 Jul 2010) | 19 lines remove empty audiohook write list on channel If a channel has an audiohook write list created on it, that list stays on the channel until the channel is destroyed. There is no reason to keep that list on the channel if it becomes empty. If it is empty that just means we are doing needless translating for every ast_read and ast_write. This patch removes the audiohook list from the channel once it is detected to be empty on either a read or write. If a audiohook is added back to the channel after this list is destroyed, the list just gets recreated as if it never existed to begin with. (closes issue #17630) Reported by: manvirr Review: https://reviewboard.asterisk.org/r/799/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@279949 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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a4f4f7443b |
Merged revisions 279657 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r279657 | qwell | 2010-07-26 17:59:52 -0500 (Mon, 26 Jul 2010) | 5 lines Really fix sounds Makefile (and make it readableish). There was a rather large syntax error that should have caused ALL versions of GNU make to fail. I don't know how it worked. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@279658 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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585d4c3440 |
Merged revisions 279561 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r279561 | tilghman | 2010-07-26 14:15:59 -0500 (Mon, 26 Jul 2010) | 2 lines Use a special Makefile for noobs who still have GNU Make 3.80. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@279562 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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bc3b185063 |
Allow for systems without locale support to be usable.
A recent change to SIP URI comparison code added a locale-specific string comparison to the mix, and certain systems do not support such functions. This fix allows for those systems to still use Asterisk 1.8 (closes issue #17697) Reported by: pprindeville Patches: asterisk-trunk-bugid17697.patch uploaded by pprindeville (license 347) Tested by: mmichelson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@279504 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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e0557f853e |
Check if ast_sockaddr is NULL then return.
(closes issue #17677) Reported by: outcast Patches: issue0017677.patch uploaded by pabelanger (license 224) Tested by: elguero git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@279280 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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3ab0041118 |
Merge the realtime failover branch
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278957 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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57a92a6a7c |
Allow IPv6 addresses for UDPTL streams.
Review: https://reviewboard.asterisk.org/r/795 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278908 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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1b2893cab4 |
Add the full current set of CDR drivers
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278579 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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d6e1c724e5 |
Remove built-in AES code and use optional_api instead
Review: https://reviewboard.asterisk.org/r/793/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278538 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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ef95349d1c |
Merged revisions 278167 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r278167 | tilghman | 2010-07-20 15:59:06 -0500 (Tue, 20 Jul 2010) | 4 lines Do not queue up DTMF frames while a call is on hold. (Fixes ABE-2110) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278272 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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b4e18d5660 |
Add load priority order, such that preload becomes unnecessary in most cases
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278132 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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6fa79e8f77 |
Make ACLs IPv6-capable.
ACLs can now be configured to match IPv6 networks. This is only relevant for ACLs in chan_sip for now since other channel drivers do not support IPv6 addressing. However, once those channel drivers are outfitted to support IPv6 addressing, the ACLs will already be ready for IPv6 support. https://reviewboard.asterisk.org/r/791 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277814 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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76d4bbacd1 |
Merged revisions 277738 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277738 | tilghman | 2010-07-17 11:59:11 -0500 (Sat, 17 Jul 2010) | 5 lines Remove uclibc cross-compile triplet, as uclibc has a working fork()... it's only uclinux that does not. (closes issue #17616) Reported by: pprindeville ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277775 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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a7c92fad28 |
Merged revisions 277568 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277568 | tilghman | 2010-07-16 16:54:29 -0500 (Fri, 16 Jul 2010) | 8 lines Since we split values at the semicolon, we should store values with a semicolon as an encoded value. (closes issue #17369) Reported by: gkservice Patches: 20100625__issue17369.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277773 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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e5c4c90064 |
Allow xmllint to be used for XML docs validation.
xmllint seems to be more commonly available since it comes with libxml2. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277703 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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fe9e0e672e |
Finally, a method that really fixes the assertions in chan_iax2.c related to cancelling lagid.
No, replacing usleep(1) with sched_yield() did not have an effect. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277484 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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753fa3b278 |
Define LLONG_MAX on systems that do not have it.
(closes issue #17644) Reported by: pprindeville git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276769 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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832d1296c6 |
Remove the old stub files, preferring the optional_api method.
(closes issue #17475) Reported by: tilghman Review: https://reviewboard.asterisk.org/r/695/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276490 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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cf7bbcc4c6 |
Expand the caller ANI field to an ast_party_id
Expand the ani field in ast_party_caller and ast_party_connected_line to an ast_party_id. This is an extension to the ast_callerid restructuring patch in review: https://reviewboard.asterisk.org/r/702/ Review: https://reviewboard.asterisk.org/r/744/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276393 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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ec37ffbdaf |
ast_callerid restructuring
The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
da8450323f |
Kill some startup warnings and errors and make some messages more helpful in tracking down the source.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275105 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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9aa4771a8d |
Merged revisions 275021 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275021 | russell | 2010-07-09 10:33:08 -0500 (Fri, 09 Jul 2010) | 4 lines Document that a leading and trailing slash is expected for test categories. Also, emit a warning if a test is registered without one of these. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275022 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
e10dcd165b |
Extend length limit on country name in indications.conf.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274907 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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cd4ebd336f |
Add IPv6 to Asterisk.
This adds a generic API for accommodating IPv6 and IPv4 addresses within Asterisk. While many files have been updated to make use of the API, chan_sip and the RTP code are the files which actually support IPv6 addresses at the time of this commit. The way has been paved for easier upgrading for other files in the near future, though. Big thanks go to Simon Perrault, Marc Blanchet, and Jean-Philippe Dionne for their hard work on this. (closes issue #17565) Reported by: russell Patches: asteriskv6-test-report.pdf uploaded by russell (license 2) Review: https://reviewboard.asterisk.org/r/743 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274783 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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a1b89a6a50 |
Implement AstData API data providers as part of the GSOC 2010 project,
midterm evaluation. Review: https://reviewboard.asterisk.org/r/757/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274727 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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f4d96da591 |
Merged revisions 273793 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273793 | tilghman | 2010-07-02 16:36:39 -0500 (Fri, 02 Jul 2010) | 9 lines Have the DEADLOCK_AVOIDANCE macro warn when an unlock fails, to help catch potentially large software bugs. (closes issue #17407) Reported by: pdf Patches: 20100527__issue17407.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/751/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273830 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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25156a326d |
Remove unnecessary if test in CV_DSTR()
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273198 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |