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${ noResults }
526 Commits (f9bcb15db337a45d1582bf8e7d1ce33a9318adf6)
Author | SHA1 | Message | Date |
---|---|---|---|
|
4405813297 |
Add High Resolution Times to CDRs for Asterisk
People expressed an interest in having access to the exact length of calls to a finer degree than seconds. See the CHANGES and UPGRADE.txt for usage also updated the sample configs to note the change. Patch by snuffy. (closes issue #16559) Reported by: cianmaher Tested by: cianmaher, snuffy Review: https://reviewboard.asterisk.org/r/461/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269153 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
857814f435 |
Add SRTP support for Asterisk
After 5 years in mantis and over a year on reviewboard, SRTP support is finally being comitted. This includes generic CHANNEL dialplan functions that work for getting the status of whether a call has secure media or signaling as defined by the underlying channel technology and for setting whether or not a new channel being bridged to a calling channel should have secure signaling or media. See doc/tex/secure-calls.tex for examples. Original patch by mikma, updated for trunk and revised by me. (closes issue #5413) Reported by: mikma Tested by: twilson, notthematrix, hemanshurpatel Review: https://reviewboard.asterisk.org/r/191/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268894 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
da0138932e |
Handle OOM errors more gracefully.
(closes issue #17084) Reported by: falves11 Patches: issue17084_162_A.diff uploaded by falves11 (license 374) Tested by: falves11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267669 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
4eaea01cad |
Needs to be wrapped in <para>
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266522 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
2da88f1977 |
Setup environment variables for the benefit of child processes and disallow changing them.
(closes issue #14899) Reported by: jmls Patches: 20090916__issue14899.diff.txt uploaded by tilghman (license 14) Tested by: jmls git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266385 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
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 |
|
03e1608c29 |
Double free crash
(closes issue #17245) Reported by: thedavidfactor Patches: 20100426__issue17245.diff.txt uploaded by tilghman (license 14) Tested by: murraytm git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261917 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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693d1c44b1 |
Add small documentation update to func_callcompletion.c.
This directs users to documents which can help explain the concepts and configuration options settable with the function. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258345 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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6640f309a9 |
Commit compromise I suggested on review 608.
This allows for multiple SRV queries to be done from the dialplan for the same service on a single call while still allowing one to bypass the call to SRVQUERY if they so please. Taking action since no comments had been left for a while. This can easily be reverted if needed. External tests still pass. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257851 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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fb0a4e5bd0 |
Address Russell's comments on func_srv from reviewboard.
* Change copyright date * Place channel in autoservice when doing SRV lookup * Get rid of trailing whitespace * Change logic in load_module function git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257025 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
ae7b76a1b9 |
Fix some compiler errors that popped up after the CCSS merge.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256529 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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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 |
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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 |
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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 |
|
008930a3f2 |
Fix memory corruption found by unit tests.
ast_str_reset() was being called on a potentially uninitialized pointer. Valgrind is my hero, once again. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253579 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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afb6bac829 |
Hmmm, apparently needed to be fixed in trunk, too.
(closes issue #16900) Reported by: bluecrow76 Patches: asterisk-1.6.2.4-func_strings.diff uploaded by bluecrow76 (license 270) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251682 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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dd3176cc91 |
It's amazing what writing a test will find.
(issue #16900) Reported by: bluecrow76 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251677 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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e58fc610ae |
Change needed to make Mac OS X 10.6 happy
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251262 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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86a215c83e |
fixes xml error in func_pitchshift
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251087 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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f468595789 |
PITCH_SHIFT dialplan function
The PITCH_SHIFT function can be used on a channel to independently modify the pitch of both rx and tx audio streams. Now you can improve your conference calls by assigning a random pitch effect to everyone entering a meetme room, or just make your day more interesting by making your co-workers sound funny. These are just some of the numerious practical uses for this function. Enjoy! https://reviewboard.asterisk.org/r/526/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251038 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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7acfebf2b8 |
Adjust XML for func_channel to indicate that rtpdest can take a "text" argument.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250730 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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e3d176d0d3 |
Remove unnecessary warning message, make a couple of formatting tweaks
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@248534 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
27a4cda821 |
Add support for GROUP_MATCH_COUNT regex matching on category
Current support for regex matching was previously only available on the group. Also, error reporting for regex failures has been added. In addition to this feature enhancement a unit test has been written to check the regular expression logic to ensure the count operation is working as expected. (closes issue #16642) Reported by: kobaz Patches: groupmatch2.patch uploaded by kobaz (license 834) Review: https://reviewboard.asterisk.org/r/503/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247295 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
47f3850a1e |
Fussy compiler on another machine...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246204 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
00b5520a6f |
Fix weird issue with unit tests on optimized build - turned out to be a signing issue.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246200 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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eaea15aa02 |
Enable warnings on atypical conditions for the FILTER function (suggested by mmichelson on the -dev list).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246022 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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5b86e43b30 |
Merged revisions 245944 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r245944 | tilghman | 2010-02-10 07:37:13 -0600 (Wed, 10 Feb 2010) | 2 lines Include examples of FILTER usage in extension patterns where a "." may be a risk. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245945 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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bbed34f764 |
Various updates to the unit test API.
1) It occurred to me that the difference in usage between the error ast_str and the ast_test_update_status() usage has turned out to be a bit ambiguous in practice. In a lot of cases, the same message was being sent to both. In other cases, it was only sent to one or the other. My opinion now is that in every case, I think it makes sense to do both; we should output it to the CLI as well as save it off for logging purposes. This change results in most of the changes in this diff, since it required changes to all existing unit tests. It also allowed for some simplifications of unit test API implementation code. 2) Update ast_test_status_update() to include the file, function, and line number for the code providing the update. 3) There are some formatting tweaks here and there. Hopefully they aren't too distracting for code review purposes. Reviewboard's diff viewer seems to do a pretty good job of pointing out when something is a whitespace change. 4) I moved the md5_test and sha1_test into the test_utils module. It seemed like a better approach since these tests are so tiny. 5) I changed the number of nodes used in heap_test_2 from 1 million to 100 thousand. The only reason for this was to reduce the time it took for this test to run. 6) Remove an unused function prototype that was at the bottom of utils.h. 7) Simplify test_insert() using the LIST_INSERT_SORTALPHA() macro. The one minor difference in behavior is that it no longer checks for a test registered with the same name. 8) Expand the code in test_alloc() to provide specific error messages for each failure case, to clearly inform developers if they forget to set the name, summary, description, etc. 9) Tweak the output of the "test show registered" CLI command. I swapped the name and category to have the category first. It seemed more natural since that is the sort key. 10) Don't output the status ast_str in the "test show results" CLI command. This is going to tend to be pretty verbose, so just leave that for the detailed test logs (test generate results). Review: https://reviewboard.asterisk.org/r/493/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245864 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
5071b6debc |
Correct some off-by-one errors, especially when expressions don't contain expected spaces.
Also include the tests provided by the reporter, as regression tests. (closes issue #16667) Reported by: wdoekes Patches: astsvn-func_match-off-by-one.diff uploaded by wdoekes (license 717) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244331 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
34317fb0d3 |
Fix the ability to specify an OSP token for an outbound IAX2 call.
When this patch was originally submitted, the code allowed for the token to be set via a channel variable. I decided that a cleaner approach would be to integrate it into the CHANNEL() function. Unfortunately, that is not a suitable approach. It's not possible to get the value set on the channel soon enough using that method. So, go back to the simple channel variable method. (closes issue #16711) Reported by: homesick Patches: iax-svn.diff uploaded by homesick (license 91) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243482 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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7770192d7f |
Update func_aes to its pre-ast_str_substitution state.
This change makes the AES tests in test_substitution.c pass. We still need to work through what's going wrong in the ast_str version. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243118 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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44a9aab93a |
Merged revisions 241765 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r241765 | tilghman | 2010-01-20 23:53:17 -0600 (Wed, 20 Jan 2010) | 2 lines Guard against division by zero. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241766 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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f6b5cf960f |
Make HASHes inheritable across channel creation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241012 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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0a1b7d8965 |
Merged revisions 238230 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r238230 | tilghman | 2010-01-06 15:41:55 -0600 (Wed, 06 Jan 2010) | 4 lines Revise documentation on disposition values to the actual values used. (closes issue #16289) Reported by: wdoekes ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@238231 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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ec98fba3ad |
Merged revisions 232268 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r232268 | dvossel | 2009-12-02 09:41:36 -0600 (Wed, 02 Dec 2009) | 9 lines fixes segfault in func_groupcount closes issue #16337) Reported by: Parantido Patches: issue_16337.diff uploaded by dvossel (license 671) Tested by: Parantido, dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232269 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
4a0c4b0578 |
Fix a build error on FreeBSD.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232012 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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0bccc4fbe6 |
Add REPLACE & PASSTHRU functions, overhaul of func_strings, fix API docs for the ast_get_encoded_* functions.
* Add REPLACE function, which searches a given variable for a set of characters and replaces each with a given character. * Add PASSTHRU function, which passes a literal string back, like a NoOp for functions. Intent is to be able to specify a literal string to another function that takes a variable name as an argument. * Let the array manipulation functions work with dialplan functions, in addition to variables. This allows the array manipulation functions to modify ASTDB and ODBC backends, assuming the func_odbc configuration has both read and write functions. (closes issue #15223) Reported by: ajohnson Patches: 20091112__issue15223.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen, tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230994 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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a1037d3d7e |
Merged revisions 229669 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r229669 | dvossel | 2009-11-12 10:41:49 -0600 (Thu, 12 Nov 2009) | 6 lines fixes merging error, datastore was being freed in the wrong function. (closes issue #16219) Reported by: aragon ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229670 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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f44f8650cb |
Merged revisions 228378 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228378 | mnicholson | 2009-11-06 10:26:59 -0600 (Fri, 06 Nov 2009) | 8 lines Properly handle '=' while decoding base64 messages and null terminate strings returned from BASE64_DECODE. (closes issue #15271) Reported by: chappell Patches: base64_fix.patch uploaded by chappell (license 8) Tested by: kobaz ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228620 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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a729d9bb4b |
fixes memory leak in func_audiohookinherit.c
(closes issue #15394) Reported by: boroda Patches: bug15394_memoryleak_diff2.txt uploaded by dbrooks (license 790) Tested by: dbrooks, boroda git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228268 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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0d1a6d9303 |
Fix XML in func_cdr.c
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228233 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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d8e0c58437 |
Expand codec bitfield from 32 bits to 64 bits.
Reviewboard: https://reviewboard.asterisk.org/r/416/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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7ed425ec80 |
This patch adds a sequence field to CDRs that can be combined with the linkedid or uniqueid field to uniquely identify a CDR.
(closes issue #15180) Reported by: Nick_Lewis Patches: cdr-sequence10.diff uploaded by mnicholson (license 96) Tested by: mnicholson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227435 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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8021cf48d8 |
Adding some clarifications to func_speex doxygen docs.
The functions needed doesn't exist in Speex 1.05 which is what a lot of distros use. 1.2 seems to have been in beta status for years, and does include the sexy functions needed for func_speex to work. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227237 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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1174a61612 |
Add support for calling and called subaddress. Partial support for COLP subaddress.
The Telecom Specs in NZ suggests that SUB ADDRESS is always on, so doing "desk to desk" between offices each with an asterisk box over the ISDN should then be possible, without a whole load of DDI numbers required. (closes issue #15604) Reported by: alecdavis Patches: asterisk_subaddr_trunk.diff11.txt uploaded by alecdavis (license 585) Some minor modificatons were made. Tested by: alecdavis, rmudgett Review: https://reviewboard.asterisk.org/r/405/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225357 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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cdd1f9e296 |
Finish implementaton of astobj2 OBJ_MULTIPLE, and convert ast_channel_iterator to use it.
This patch finishes the implementation of OBJ_MULTIPLE in astobj2 (the case where multiple results need to be returned; OBJ_NODATA mode already was supported). In addition, it converts ast_channel_iterators (only the targeted versions, not the ones that iterate over all channels) to use this method. During this work, I removed the 'ao2_flags' arguments to the ast_channel_iterator constructor functions; there were no uses of that argument yet, there is only one possible flag to pass, and it made the iterators less 'opaque'. If at some point in the future someone really needs an ast_channel_iterator that does not lock the container, we can provide constructor(s) for that purpose. Review: https://reviewboard.asterisk.org/r/379/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225244 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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77031501a5 |
Merged revisions 224855 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224855 | tilghman | 2009-10-20 17:07:11 -0500 (Tue, 20 Oct 2009) | 5 lines Pay attention to the return value of the manipulate function. While this looks like an optimization, it prevents a crash from occurring when used with certain audiohook callbacks (diagnosed with SVN trunk, backported to 1.4 to keep the source consistent across versions). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224856 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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1c9fe00920 |
Recorded merge of revisions 222152 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05 Oct 2009) | 20 lines Fix ao2_iterator API to hold references to containers being iterated. See Mantis issue for details of what prompted this change. Additional notes: This patch changes the ao2_iterator API in two ways: F_AO2I_DONTLOCK has become an enum instead of a macro, with a name that fits our naming policy; also, it is now necessary to call ao2_iterator_destroy() on any iterator that has been created. Currently this only releases the reference to the container being iterated, but in the future this could also release other resources used by the iterator, if the iterator implementation changes to use additional resources. (closes issue #15987) Reported by: kpfleming Review: https://reviewboard.asterisk.org/r/383/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222176 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
63984d5c21 |
Merged revisions 221153,221157,221303 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221153 | mnick | 2009-09-30 10:37:39 -0500 (Wed, 30 Sep 2009) | 2 lines check bounds - prevents for buffer overflow ........ r221157 | mnick | 2009-09-30 10:41:46 -0500 (Wed, 30 Sep 2009) | 8 lines added a new dialplan function 'CSV_QUOTE' and changed the cdr_custom.sample.conf (closes issue #15471) Reported by: dkerr Patches: csv_quote_14.txt uploaded by mnick (license ) Tested by: mnick ........ r221303 | mnick | 2009-09-30 14:02:00 -0500 (Wed, 30 Sep 2009) | 2 lines changed the prototype definition of csv_quote ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221368 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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751f191dfa |
Allow locks to be inherited through a masquerade without causing starvation.
(closes issue #14859) Reported by: atis Patches: 20090821__issue14859.diff.txt uploaded by tilghman (license 14) 20090925__issue14859__1.6.1.diff.txt uploaded by tilghman (license 14) Tested by: atis, tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221044 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |