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2011-02-15Merged revisions 307836 via svnmerge from tilghman1-1/+4
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r307836 | tilghman | 2011-02-15 01:01:37 -0600 (Tue, 15 Feb 2011) | 8 lines Need to retrieve the rows affected before using the associated variable. (closes issue #18795) Reported by: irroot Patches: 20110211__issue18795.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@307837 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-02Eliminate a file descriptor leak when using the FILE() dialplan function.tilghman1-0/+6
(closes issue #18731) Reported by: marioabajo git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@305844 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-02Replacing doc/* and asterisk.pdf with wiki linkslathama2-2/+2
Adding links to http(s)://wiki.asterisk.org git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@305838 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-30Add Function and Application Relationships to documentationlathama2-2/+32
Add and extend the see-also sections to the documentation for applications and functions in an effort to expand the online documentation of the wiki. Also check for and update any links to moved documentation in the doc folder. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@304908 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-14Add relationships to function documentation.lathama2-12/+12
Fix amatuer type mistake git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@301849 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-14Add relationships to function documentation.lathama2-0/+20
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@301844 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-16Merged revisions 298477 via svnmerge from tilghman1-0/+5
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r298477 | tilghman | 2010-12-16 02:54:23 -0600 (Thu, 16 Dec 2010) | 8 lines Eliminate duplicates from container. (closes issue #18091) Reported by: bunny Patches: 20101006__issue18091.diff.txt uploaded by tilghman (license 14) Tested by: bunny ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@298478 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-15Merged revisions 294988 via svnmerge from tilghman1-1/+4
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294988 | tilghman | 2010-11-15 01:42:39 -0600 (Mon, 15 Nov 2010) | 8 lines It is possible to crash Asterisk by feeding the curl engine invalid data. (closes issue #18161) Reported by: wdoekes Patches: 20101029__issue18161.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@294989 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-28Merged revisions 293158 via svnmerge from jpeeler1-3/+3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r293158 | jpeeler | 2010-10-28 11:09:40 -0500 (Thu, 28 Oct 2010) | 11 lines Fix infinite loop in FILTER(). Specifically when you're using characters above \x7f or invalid character escapes (e.g. \xgg). (closes issue #18060) Reported by: wdoekes Patches: issue18060_func_strings_filter_infinite_loop.patch uploaded by wdoekes (license 717) Tested by: wdoekes ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293159 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-30Solaris fixes.tilghman1-1/+6
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@289581 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-24Merged revisions 288712 via svnmerge from tilghman1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r288712 | tilghman | 2010-09-24 08:53:30 -0500 (Fri, 24 Sep 2010) | 5 lines Solaris won't printf a NULL. (closes issue #18041) Reported by: asgaroth ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@288713 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-20Addition of the FrameHook API (AKA AwesomeHooks)dvossel1-0/+365
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: http://svn.digium.com/svn/asterisk/branches/1.8@287647 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-10Merged revisions 286115 via svnmerge from twilson1-1/+19
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: http://svn.digium.com/svn/asterisk/branches/1.8@286189 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-09Merged revisions 285710 via svnmerge from bbryant3-4/+17
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r285710 | bbryant | 2010-09-09 14:50:13 -0400 (Thu, 09 Sep 2010) | 8 lines Fixes an issue with dialplan pattern matching where the specificity for pattern ranges and pattern special characters was inconsistent. (closes issue #16903) Reported by: Nick_Lewis Patches: pbx.c-specificity.patch uploaded by Nick Lewis (license 657) Tested by: Nick_Lewis ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@285711 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-03Sneak FIELDNUM() into 1.8. Returns a 1-based index into a list of a ↵tilghman1-0/+170
specified item. Matches up with FIELDQTY() and CUT(). (closes issue #17713) Reported by: gareth Patches: svn-279754.diff uploaded by gareth (license 208) Tested by: gareth, tilghman Review: https://reviewboard.asterisk.org/r/810/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@280809 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-21Remove built-in AES code and use optional_api insteadtwilson1-3/+6
Review: https://reviewboard.asterisk.org/r/793/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278538 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-20Add load priority order, such that preload becomes unnecessary in most casestilghman2-2/+10
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278132 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14Expand the caller ANI field to an ast_party_idrmudgett1-10/+47
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: http://svn.digium.com/svn/asterisk/trunk@276393 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14ast_callerid restructuringrmudgett5-961/+1370
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: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-13Oops, XML documentation fix.tilghman1-1/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276122 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-13It really cannot fail in the places below, but the stupid compiler doesn't ↵tilghman1-5/+17
know that. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276120 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-13Weird compiler error on Bamboo.tilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276118 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-13FILE() now supports line-mode and writing (altering) files.tilghman1-48/+1001
(closes issue #16461) Reported by: skyman Patches: 20100622__issue16461.diff.txt uploaded by tilghman (license 14) Tested by: tilghman Review: https://reviewboard.asterisk.org/r/737/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276114 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-09Kill some startup warnings and errors and make some messages more helpful in ↵tilghman2-2/+2
tracking down the source. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275105 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-08Add High Resolution Times to CDRs for Asterisksnuffy1-3/+37
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: http://svn.digium.com/svn/asterisk/trunk@269153 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-08Add SRTP support for Asterisktwilson1-0/+50
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: http://svn.digium.com/svn/asterisk/trunk@268894 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-03Handle OOM errors more gracefully.tilghman1-2/+22
(closes issue #17084) Reported by: falves11 Patches: issue17084_162_A.diff uploaded by falves11 (license 374) Tested by: falves11 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267669 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-30Needs to be wrapped in <para>tilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@266522 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-28Setup environment variables for the benefit of child processes and disallow ↵tilghman1-1/+2
changing them. (closes issue #14899) Reported by: jmls Patches: 20090916__issue14899.diff.txt uploaded by tilghman (license 14) Tested by: jmls git-svn-id: http://svn.digium.com/svn/asterisk/trunk@266385 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-17Enhancements to connected line and redirecting work.mmichelson3-2/+32
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: http://svn.digium.com/svn/asterisk/trunk@263541 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-07Double free crashtilghman1-0/+3
(closes issue #17245) Reported by: thedavidfactor Patches: 20100426__issue17245.diff.txt uploaded by tilghman (license 14) Tested by: murraytm git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261917 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-21Add small documentation update to func_callcompletion.c.mmichelson1-1/+4
This directs users to documents which can help explain the concepts and configuration options settable with the function. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258345 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-19Commit compromise I suggested on review 608.mmichelson1-0/+14
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: http://svn.digium.com/svn/asterisk/trunk@257851 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-13Address Russell's comments on func_srv from reviewboard.mmichelson1-7/+14
* 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: http://svn.digium.com/svn/asterisk/trunk@257025 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-09Fix some compiler errors that popped up after the CCSS merge.mmichelson1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@256529 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-09Merge Call completion support into trunk.mmichelson1-0/+114
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: http://svn.digium.com/svn/asterisk/trunk@256528 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-09func_srv and explicit specification of a remote IP for SIP.mmichelson1-0/+254
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: http://svn.digium.com/svn/asterisk/trunk@256485 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-03Consolidate ast_channel.cid.cid_rdnis into ast_channel.redirecting.from.number.rmudgett2-11/+5
SWP-1229 ABE-2161 * Ensure chan_local.c:local_call() will not leak cid.cid_dnid when copying. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@256104 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-20Fix memory corruption found by unit tests.russell1-1/+1
ast_str_reset() was being called on a potentially uninitialized pointer. Valgrind is my hero, once again. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@253579 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-10Hmmm, apparently needed to be fixed in trunk, too.tilghman1-1/+1
(closes issue #16900) Reported by: bluecrow76 Patches: asterisk-1.6.2.4-func_strings.diff uploaded by bluecrow76 (license 270) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@251682 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-10It's amazing what writing a test will find.tilghman1-1/+1
(issue #16900) Reported by: bluecrow76 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@251677 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-08Change needed to make Mac OS X 10.6 happytilghman1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@251262 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-05fixes xml error in func_pitchshiftdvossel1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@251087 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-05PITCH_SHIFT dialplan functiondvossel1-0/+503
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: http://svn.digium.com/svn/asterisk/trunk@251038 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-04Adjust XML for func_channel to indicate that rtpdest can take a "text" argument.mmichelson1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250730 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-24Remove unnecessary warning message, make a couple of formatting tweaksrussell1-3/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@248534 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-17Add support for GROUP_MATCH_COUNT regex matching on categoryjpeeler1-2/+3
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: http://svn.digium.com/svn/asterisk/trunk@247295 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-10Fussy compiler on another machine...tilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@246204 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-10Fix weird issue with unit tests on optimized build - turned out to be a ↵tilghman1-6/+7
signing issue. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@246200 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-10Enable warnings on atypical conditions for the FILTER function (suggested by ↵tilghman1-0/+8
mmichelson on the -dev list). git-svn-id: http://svn.digium.com/svn/asterisk/trunk@246022 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-04-23 18:57:34 +0000