110 Commits (fc21c6f9e91af13c7379671a5bbe14ee4832fd9d)

Author	SHA1	Message	Date
Olle Johansson	e129b31fc6	Add ability to configure the Max-Forwards header in the dialplan, as well as in ... sip.conf configuration for the channel and for devices. The Max-Forwards header is used to prevent loops in a SIP network. Each intermediary, like SIP proxys and SBCs, decrement this counter and detects when it reaches zero, at which point the SIP request is nicely killed in a SIP-friendly way. Review: https://reviewboard.asterisk.org/r/778/ Thanks to dvossel for the review and good advice. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276951 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tim Ringenbach	e19a6c248f	Fix documentation for pgsql cel and cdr, and slightly improve pgsql_cel. ... Change the documented pgsql schema to use "timestamp" instead of "time", as the latter is only a time without a date. Added some missing columns for cel's pgsql schema, and corrected spelling on some others. Updated cel's uniqueid size to be the same as the cdr. Added id column to cel's pgsql schema and updated code to allow unknown columns to get their default value instead of forcing 0 or empty string. Added microseconds to the timestamp cel logs to pgsql. Review: https://reviewboard.asterisk.org/r/734 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276349 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	910cd8be1d	Use the underscore package so that underscores do not need to be escaped. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272684 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	5d2bbe86ad	Update formatting for channelvariables.tex ... (closes issue #17511) Reported by: klaus3000 Patches: channelvariables.tex-patch.txt uploaded by klaus3000 (license 65) Tested by: pabelanger git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270801 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	326c783685	Add documentation explaining PLC in Asterisk. ... Review: https://reviewboard.asterisk.org/r/688/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269749 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	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
Terry Wilson	3c9a8ebadb	Add the FullyBooted AMI event ... It is possible to connect to the manager interface before all Asterisk modules are loaded. To ensure that an application does not send AMI actions that might require a module that has not yet loaded, the application can listen for the FullyBooted manager event. It will be sent upon connection if all modules have been loaded, or as soon as loading is complete. The event: Event: FullyBooted Privilege: system,all Status: Fully Booted Review: https://reviewboard.asterisk.org/r/639/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265320 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	8096f0fecc	Add MEETMEBOOKID from r256019. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258515 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	5258e5e683	Missed this when reverting the bad version change in asterisk.tex. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258387 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	8ea4ecd58a	Fix change in asterisk.tex that got merged in after testing. ... (issue #17220) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258383 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	8b11ae2e4f	Add ability to generate ASCII documentation from the TeX files. ... These changes add the ability to run 'make asterisk.txt' just like the existing 'make asterisk.pdf' commands to generate a text document from the TeX files we have in the doc/tex/ directory. I've also updated a few of the .tex files because they weren't properly escaping certain characters so they would show up as Unicode characters (like [U+021C]). Made changes to the configure scripts so it would detect the catdvi program which is required to convert the .dvi file generated by latex. I've also added a few lines to the build_tools/prep_tarball script so that the text documentation gets generated and added to future tarballs of Asterisk releases. (closes issue #17220) Reported by: lmadsen Patches: asterisk.txt.patch uploaded by lmadsen (license 10) asterisk.txt.patch-v4 uploaded by pabelanger (license 224) Tested by: lmadsen, pabelanger git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258351 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Julian Lyndon-Smith	81fd235286	Added NEW ACTIONS entry for new MixMonitorMute AMI command. ... Added State and Direction variables for new MixMonitorMute AMI command. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258228 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	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
Leif Madsen	3a4baef6d9	Fix for localchannel.tex to allow PDFs to be generated again. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256161 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	b6e783d97c	Update to new Local channel documentation. ... Add same changes as commit to 1.4, but convert to TeX. (issue #16963) Reported by: kobaz Patches: localchannel-2.txt uploaded by kobaz (license 834) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253256 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	06ff72a78e	Update existing Local channel documentation. ... A complete re-write of the Local channel documentation has been performed, with the existing information from localchannel.txt and localchannel.tex merged in. (closes issue #16637) Reported by: kobaz Patches: localchannel.tex uploaded by lmadsen (license 10) localchannel.txt uploaded by lmadsen (license 10) Tested by: lmadsen, jsmith, mmichelson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250609 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	491ea82b6d	Update IMAP documentation. ... Update the IMAP documentation to make it clear that storing voicemails in the same folder as a large number of emails could potentially cause significant slow downs when writing or retrieving voicemails. (issue #16704) Reported by: TimeHider Tested by: lmadsen, TimeHider git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250051 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	aec5c6f840	Update documentation to not imply we support overriding options. ... (closes issue #16855) Reported by: davidw git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250037 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
TransNexus OSP Development	ce80ff55c8	Updated channel variable list of osplookup application. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239625 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	7ab15342ac	Update IMAP build documentation. ... Update the IMAP build documentation to show how to build on 64-bit platforms. (issue #16433) Reported by: shrift Tested by: lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234631 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	7c4fd85293	Fix the localchannel.tex file. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228499 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	b9c370da86	Merged revisions 226531 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226531 | file | 2009-10-29 15:11:26 -0300 (Thu, 29 Oct 2009) | 6 lines Add an option to enabling passing music on hold start and stop requests through instead of acting on them in chan_local. (closes issue #14709) Reported by: dimas ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226532 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	6b6fee6c95	Merged revisions 226377 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226377 | lmadsen | 2009-10-28 14:48:29 -0500 (Wed, 28 Oct 2009) | 7 lines Update CALLINGSUBADDR channel variable documentation. (closes issue #15734) Reported by: alecdavis Patches: channelvariables.tex.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226378 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	87ff40d3f3	Add 'mohsuggest' configuration option to 'sip show peer' CLI command and ... SIPShowPeer AMI action. (closes issue #15990) Reported by: _brent_ Patches: sip_peer_info_mohsuggest-r3.patch uploaded by brent (license 388) Review: https://reviewboard.asterisk.org/r/381/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225245 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	08df1b85bf	Add forgotten documentation for new channel variables added in 214309. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214355 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	0e70c71c25	Convert this branch to Opsound music-on-hold. ... For more details: http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@212922 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	b8f1c9c4c3	Merged revisions 210563 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r210563 | lmadsen | 2009-08-05 13:46:21 -0500 (Wed, 05 Aug 2009) | 11 lines Update imapstorage.txt documentation. Updated the imapstorage.txt documentation to reflect that issues with c-client versions older than 2007 seem to cause crashing issues that are not seen with more recent versions. Documentation has been updated to reflect this. (closes issue #14496) Reported by: vbcrlfuser Patches: __20090727-imap-documentation-patch.txt uploaded by lmadsen (license 10) Tested by: lmadsen, mmichelson, dbrooks ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@210564 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	e9d22f802e	Rename 'canreinvite' option to 'directmedia', with backwards compatibility. ... It is clear from multiple mailing list, forum, wiki and other sorts of posts that users don't really understand the effects that the 'canreinvite' config option actually has, and that in some cases they think that setting it to 'no' will actually cause various other features (T.38, MOH, etc.) to not work properly, when in fact this is not the case. This patch changes the proper name of the option to what it should have been from the beginning ('directmedia'), but preserves backwards compatibility for existing configurations. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@210190 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Bradley Latus	9eb46f3286	Update documentation in relation to UnixODBC ... (closes issue #15516) Reported by: snuffy Patches: bug_odbc_tex_update_v2.diff uploaded by snuffy (license 35) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209959 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	aa379bb741	Document the "flag" field in the voicemessages table. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207224 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	4cf8a968fd	Add an API for reporting security events, and a security event logging module. ... This commit introduces the security events API. This API is to be used by Asterisk components to report events that have security implications. A simple example is when a connection is made but fails authentication. These events can be used by external tools manipulate firewall rules or something similar after detecting unusual activity based on security events. Inside of Asterisk, the events go through the ast_event API. This means that they have a binary encoding, and it is easy to write code to subscribe to these events and do something with them. One module is provided that is a subscriber to these events - res_security_log. This module turns security events into a parseable text format and sends them to the "security" logger level. Using logger.conf, these log entries may be sent to a file, or to syslog. One service, AMI, has been fully updated for reporting security events. AMI was chosen as it was a fairly straight forward service to convert. The next target will be chan_sip. That will be more complicated and will be done as its own project as the next phase of security events work. For more information on the security events framework, see the documentation generated from doc/tex/. "make asterisk.pdf" Review: https://reviewboard.asterisk.org/r/273/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206021 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	c511a26749	Move Asterisk-addons modules into the main Asterisk source tree. ... Someone asked yesterday, "is there a good reason why we can't just put these modules in Asterisk?". After a brief discussion, as long as the modules are clearly set aside in their own directory and not enabled by default, it is perfectly fine. For more information about why a module goes in addons, see README-addons.txt. chan_ooh323 does not currently compile as it is behind some trunk API updates. However, it will not build by default, so it should be okay for now. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204413 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	0264eef115	Merge the new Channel Event Logging (CEL) subsystem. ... CEL is the new system for logging channel events. This was inspired after facing many problems trying to represent what is possible to happen to a call in Asterisk using CDR records. For more information on CEL, see the built in HTML or PDF documentation generated from the files in doc/tex/. Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard work developing this code. Also, thanks to Matt Nicholson (mnicholson) and Sean Bright (seanbright) for their assistance in the final push to get this code ready for Asterisk trunk. Review: https://reviewboard.asterisk.org/r/239/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203638 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	5059530d62	Change some section names in the CDR tex documentation. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202841 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	233f1bc8da	Remove some trailing whitespace before making content changes. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202840 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	1fe9c73c19	Clean up section hierarchy for the CDR chapter. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202804 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	dcc651d99b	Add missing closure of verbatim environment. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200762 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	298d745fb4	Add the ability to execute connected line interception macros. ... When connected line updates are received or generated in the middle of an application call, it is now possible to execute a macro to manipulate the connected line data. This way, phone numbers may be manipulated to be more presentable to users, names may be changed for...whatever reason, or whatever else needs to be done may be. Review: https://reviewboard.asterisk.org/r/256 AST-165 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198727 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	ce004fbf1f	Add some TeX docs for calendaring. ... I still need to set up tests to make sure my examples are completely correct, but I ran out of time tonight and felt that they at least would give an idea as to how to use calendaring. I will try to test the examples and do some cleanup on the docs tomorrow night. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197926 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	f51bb019bb	Update references to bugs.digium.com and reviewboard.digium.com to the new URLs. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197824 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	4bf441ca39	UniqueID column has a maximum size of 150 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190904 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	1d941ad821	Allow for a position to be specified when entering a queue. ... This would allow for one to add a caller to a specific place in the queue instead of just placing the caller in the back every time. To help facilitate some interesting manipulations, a new channel variable called QUEUEPOSITION has been added. When a caller is removed from a queue, his position in that queue is stored in the QUEUEPOSITION variable. One such strategy an administrator can employ is to allow for the removal of a caller from one queue followed by the insertion of the same caller into a separate queue in the same position. Review: http://reviewboard.digium.com/r/189 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190626 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	4eaa651a8a	Add support for changing the outbound codec on a SIP call using ... a dialplan variable. This adds a dialplan variable (SIP_CODEC_OUTBOUND) which controls the codec offered for an outgoing SIP call. This is much like the SIP_CODEC dialplan variable and has the same restrictions. The codec set must be one that is configured for the call. (closes issue #13243) Reported by: samdell3 Patches: 13243.diff uploaded by file (license 11) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186624 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	796ec24ed5	Update documentation for DIALEDTIME and ANSWEREDTIME variables. ... (closes issue #14566) Reported by: klaus3000 Patches: ANSWEREDTIME-1.4-patch.txt uploaded by klaus3000 (license 65) ANSWEREDTIME-trunk-patch.txt uploaded by klaus3000 (license 65) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179057 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	7ed9ece337	Fix asterisk.pdf generation if branch name has an underscore in it. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170794 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	ac7bd2ddd9	Don't blow up if a branch name has an underscore in it ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170790 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	2c6a7907ff	remove duplicated sentence. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169793 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	45bc07c439	Use tables instead of ASCII art. Also change a bit of minor formatting. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164565 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	c0fc8edbbd	Use a \picture instead of ASCII art. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164137 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	3b96ae826e	This shouldn't have gotten commited. We might want to generate this into a separate file instead of the version controlled one. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163952 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago