asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	44cc294c89	Added documentation for CONNECTEDLINE and REDIRECTING functions. (closes issue #17808) Reported by: jtodd Review: https://reviewboard.asterisk.org/r/875/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284698 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	e34a92e2d0	Merged revisions 282469 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r282469 \| lmadsen \| 2010-08-16 13:00:09 -0500 (Mon, 16 Aug 2010) \| 7 lines Add information about creating sounds files using the sounds tools publically available so that others can create their own sounds prompts using the same tools we use to generate sounds releases. This allows people creating their own prompts to sound consistent with the prompts available from the open source project. SWP-595 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@282470 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
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
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
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
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
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
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	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
Sean Bright	3f12b282b6	Use actual tables instead of ASCII art ones. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163951 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	8fb13c6803	Grammar hax from Qwell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137301 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	f6bda9def2	Note that developer documentation belongs in doxygen, and not integrated with the user manual stuff in doc/tex/. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137299 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	8faa81be79	restore ASTERISKVERSION marker to asterisk.tex. This got lost in commit 97634 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130479 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	3570ad103d	Added a new module, res_phoneprov, which allows auto-provisioning of phones based on configuration templates that use Asterisk dialplan function and variable substitution. It should be possible to create phone profiles and templates that work for the majority of phones provisioned over http. It is currently only intended to provision a single user account per phone. An example profile and set of templates for Polycom phones is provided. NOTE: Polycom firmware is not included, but should be placed in AST_DATA_DIR/phoneprov/configs to match up with the included templates. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97634 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	8b8a42e61c	Many doc directory improvements, including: - Added development section (backtrace.tex) - Correct filesystem path formating - Replace all "\|" argument separator to "," - Endless count of spaces at the end of line - Using astlisting to make listings do not take so much place - Take back ASTRISKVERSION on first page - Make localchannel.tex readable by inserting extra end of lines (closes issue #10962) Reported by: IgorG Patches: texdoc-85177-1.patch uploaded by IgorG (license 20) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85519 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	51e7035dfe	Merge a big batch of documentation fixes for escaping, marking URLs, places where verbatim text went off the end of the page on the PDF, and various other improvements (closes issue #10307, IgorG) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77284 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	95bee48cde	Fix escaping and some of the formattting (closes issue #10285 ) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76874 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	d2a85e5f64	After some discussion on the asterisk-dev list, we determined that this approach for extracting application, function, manager, and agi documentation is the wrong one to take. The most severe problem is that the output depends on which modules are loaded as well as compile time options, which both determine which parts are available. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@72986 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	df1a340774	* Move LaTeX docs into a tex/ subdirectory of the doc/ dir * Add a Makefile in doc/tex/ for generating PDF and HTML * Add a README.txt file to doc/tex/ to document which tools are used and what web sites to visit for getting them. * Update build_tools/prep_tarball to put the proper Asterisk version string in the automatically generated PDF for release tarballs git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@72982 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago

25 Commits (c271acacdab9165f3cfae52c53d7670d0e73f07f)