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(issue #17220)
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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
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Added State and Direction variables for new MixMonitorMute AMI command.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r257426 | lmadsen | 2010-04-15 14:40:33 -0500 (Thu, 15 Apr 2010) | 13 lines
Update backtrace.txt documentation.
Update the backtrace.txt documentation so it conforms to the same layout as
other documents we've been working on recently. Additionally, add a bunch of
new information about gathering backtraces for crashes and deadlocks, along
with ways of verifying your file before uploading it. Create a couple of one
line commands for people to generate the files we need.
(closes issue #17190)
Reported by: lmadsen
Patches:
backtrace.txt.patch-2 uploaded by lmadsen (license 10)
Tested by: lmadsen, pabelanger
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r257342 | lmadsen | 2010-04-15 08:41:45 -0500 (Thu, 15 Apr 2010) | 1 line
Update address of the bug tracker.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r256900 | lmadsen | 2010-04-12 12:29:26 -0500 (Mon, 12 Apr 2010) | 15 lines
Add How-To document on collecting debugging info for issues.asterisk.org
Paul Belanger has been helping a lot with bug tracking recently and created
this document that we can now point to when additional debugging information
is required. This document will help those filing issues to know how to get
the information required when filing their issues. This will make things
easier on the developers.
Initial text and changes by pabelanger. Tweaks and editing by myself.
(closes issue #17159)
Reported by: pabelanger
Patches:
HOWTO_collect_debug_information.txt.patch uploaded by lmadsen (license 10)
Tested by: tzafrir, pabelanger, lmadsen
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In troff '-' is used for a hyphen. A minus is denoted by '\-' . This is
normally also used for a dash.
This patch converts all '-'-s that are minuses or dashes to '\-'.
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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
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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)
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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
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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
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(closes issue #16855)
Reported by: davidw
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See http://lists.digium.com/pipermail/asterisk-users/2010-January/243462.html
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Rewrote a large portion of the existing documentation
and added information about the TCP/IP socket interface
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Add a how-to set of documentation about building queues with Asterisk.
This documentation is based on Asterisk 1.6.2 but should work on most
versions with minor modifications.
(closes issue #16237)
Reported by: lmadsen
Patches:
Building Queues (FINAL).txt uploaded by lmadsen (license 10)
Tested by: pdhales, lmadsen, cmdrwalrus
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Also add an XXX comment that I'm baffled nobody has ever complained about. We
say "first message", and then we go into language-specific stuff where we
proceed to say..."first message".
(closes issue #15053)
Reported by: dinhtrung
Patches:
vietnamese.ods uploaded by dinhtrung (license 776)
app_voicemail.c.diff uploaded by dinhtrung (license 776)
(closes issue #15626)
Reported by: dinhtrung
Patches:
say.c.diff uploaded by dinhtrung (license 776)
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Update the IMAP build documentation to show how to build on 64-bit
platforms.
(issue #16433)
Reported by: shrift
Tested by: lmadsen
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configuration files.
This option can be used to enable #exec support in the asterisk.conf configuration file.
(closes issue #16260)
Reported by: atis
Patches:
exec_includes.patch uploaded by atis (license 242)
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with guidelines
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also include a better cmd define for S command
Review: https://reviewboard.asterisk.org/r/430/
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Remove non-functional socket implementation of ExternalIVR from documentation
(closes issue #16225)
Reported by: thedavidfactor
Patches:
externalivr.txt.20091111.1542.patch uploaded by thedavidfactor (license 903)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r229355 | diruggles | 2009-11-10 16:45:15 -0500 (Tue, 10 Nov 2009) | 9 lines
Fix ExternalIVR Documentation
Remove documentation for event that doesn't function
(closes issue #16220)
Reported by: thedavidfactor
Patches:
externalivr.txt.20091110.1622.patch uploaded by thedavidfactor (license 903)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r229191 | diruggles | 2009-11-10 12:23:59 -0500 (Tue, 10 Nov 2009) | 11 lines
Document ExternalIVR event tag collision
ExternalIVR uses the D tag for two different event types. This documents that
behavior and how to differentiate between the two cases. Also includes a minor
spelling fix and clarification
(closes issue #16211)
Reported by: thedavidfactor
Patches:
externalivr.txt.20091109.1507.patch uploaded by thedavidfactor (license 903)
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Reviewboard: https://reviewboard.asterisk.org/r/416/
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r225484 | lmadsen | 2009-10-22 16:51:52 -0500 (Thu, 22 Oct 2009) | 11 lines
Clean valgrind output by suppressing false errors.
Update valgrind.txt documentation and add valgrind.supp file in order to
allow those who are creating valgrind output to have less false errors in
the logfile.
(closes issue #16007)
Reported by: atis
Patches:
valgrind.txt.diff uploaded by atis (license 242)
asterisk2.supp uploaded by atis (license 242)
Tested by: atis, amorsen
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Nothing in utils/ is now built by default except for astcanary.
Review: https://reviewboard.asterisk.org/r/353/
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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/
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JABBER_RECEIVE (along with JabberSend) makes Asterisk interact with users over
XMPP to process calls.
SendText can be used instead of JabberSend in the context of XMPP based voice
channels (chan_gtalk and chan_jingle).
(closes issue #12569)
Reported by: eech55
Tested by: phsultan, asannucci, lmadsen, jtodd, maxgo
Review: https://reviewboard.asterisk.org/r/88/
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(closes issue #15946)
Reported by: jtodd
Patches:
leif-is-a-wuss.txt uploaded by jtodd (license 870)
Tested by: jparker, tilghman, jtodd, russellb, mmichelson, seanbright, kpfleming, and the rest of the usual suspects
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For new readers: The janitor list is a list of tasks we need help with in the Asterisk project. Taking up
one of these is often a good way to get into Asterisk development and getting a lot of developers in
the project to be grateful. It's stuff we could spend time on when the bug tracker is empty, when our
employers hasn't filled our task lists and our servers is running bugfree and happily without any issues.
If you want to start working on one of these small projects, feel free to ask for help in the #asterisk-dev
channel on IRC or asterisk-dev mailing list. We'll be more than happy to help you to start and reach
goal.
Thank you for your help.
</end of long commit message>
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r216263 | russell | 2009-09-04 05:48:00 -0500 (Fri, 04 Sep 2009) | 9 lines
Merged revisions 216262 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r216262 | russell | 2009-09-04 05:47:37 -0500 (Fri, 04 Sep 2009) | 2 lines
Add a plain text version of the IAX2 security document.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r216008 | russell | 2009-09-03 13:44:58 -0500 (Thu, 03 Sep 2009) | 9 lines
Merged revisions 216005 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r216005 | russell | 2009-09-03 13:42:24 -0500 (Thu, 03 Sep 2009) | 2 lines
Add IAX2 security document related to AST-2009-006.
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This patch adds a document describing the language prompt submission process,
licensing terms and other issues related to that process. In addition, it
modifies the sound file searching process to support language codes with
any number of suffices (not limited to just "xx" or "xx_YY"), so that prompts
can be named with gender, customer/company, etc. suffices as well.
(closes issue #15771)
Reported by: jtodd
Patches:
language-criteria.txt uploaded by jtodd
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the configured parkinglot in their response.
Prodded by snuff-work on #asterisk-dev IRC channel
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For more details:
http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r211583 | tilghman | 2009-08-10 14:48:48 -0500 (Mon, 10 Aug 2009) | 1 line
Conversion specifiers, not format specifiers
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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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.
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