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https://origsvn.digium.com/svn/asterisk/trunk
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r179937 | mmichelson | 2009-03-03 14:59:16 -0600 (Tue, 03 Mar 2009) | 20 lines
Add documentation for timing modules used in Asterisk
This document specifies the timing modules available in Asterisk beginning
with Asterisk 1.6.1. The document goes into detail about the differences
between each and gives a general overview of what timing is used for in
Asterisk. There is also a section which can be used to help customize
your setup or to troubleshoot timing issues you may have.
I also added messages to the DAHDI timing test used in res_timing_dahdi.c
that points to this new documentation if people experience problems.
Big thanks to all who contributed comments on this.
(closes issue #14490)
Reported by: mmichelson
Patches:
timing.txt uploaded by mmichelson (license 60)
Review: http://reviewboard.digium.com/r/164/
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https://origsvn.digium.com/svn/asterisk/trunk
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r179164 | russell | 2009-02-27 15:47:18 -0600 (Fri, 27 Feb 2009) | 2 lines
Mark res_ais as experimental, as the binary event format is subject to change.
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https://origsvn.digium.com/svn/asterisk/trunk
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r179057 | qwell | 2009-02-27 13:04:57 -0600 (Fri, 27 Feb 2009) | 8 lines
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)
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r175952 | mvanbaak | 2009-02-16 01:26:59 +0100 (Mon, 16 Feb 2009) | 10 lines
Merged revisions 175921 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r175921 | mvanbaak | 2009-02-16 00:37:03 +0100 (Mon, 16 Feb 2009) | 3 lines
fix mis-spelling of the word registered.
Reported by De_Mon on #asterisk-dev.
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r170794 | rmudgett | 2009-01-23 17:10:34 -0600 (Fri, 23 Jan 2009) | 1 line
Fix asterisk.pdf generation if branch name has an underscore in it.
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https://origsvn.digium.com/svn/asterisk/trunk
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r170790 | russell | 2009-01-23 16:58:37 -0600 (Fri, 23 Jan 2009) | 2 lines
Don't blow up if a branch name has an underscore in it
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https://origsvn.digium.com/svn/asterisk/trunk
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r169793 | mvanbaak | 2009-01-21 23:04:16 +0100 (Wed, 21 Jan 2009) | 2 lines
remove duplicated sentence.
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https://origsvn.digium.com/svn/asterisk/trunk
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r162667 | mmichelson | 2008-12-10 10:39:10 -0600 (Wed, 10 Dec 2008) | 16 lines
Merged revisions 162659 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r162659 | mmichelson | 2008-12-10 10:10:25 -0600 (Wed, 10 Dec 2008) | 8 lines
Add missing documentation to misdn.txt
(closes issue #14052)
Reported by: festr
Patches:
misdn.txt.patch uploaded by festr (license 443)
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https://origsvn.digium.com/svn/asterisk/trunk
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r155803 | tilghman | 2008-11-10 14:49:59 -0600 (Mon, 10 Nov 2008) | 1 line
I got tired of saying this in every single bugnote referring to this file.
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https://origsvn.digium.com/svn/asterisk/trunk
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r149040 | lmadsen | 2008-10-14 15:35:54 -0400 (Tue, 14 Oct 2008) | 5 lines
Add missing documentation for SipShowRegistry action and RegistryEntry event.
(closes issue #13342)
Reported and patch by: Laureano
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https://origsvn.digium.com/svn/asterisk/trunk
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r147807 | murf | 2008-10-09 08:17:33 -0600 (Thu, 09 Oct 2008) | 15 lines
(closes issue #13557)
Reported by: nickpeirson
Patches:
pbx.c.patch uploaded by nickpeirson (license 579)
replace_bzero+bcopy.patch uploaded by nickpeirson (license 579)
Tested by: nickpeirson, murf
1. replaced all refs to bzero and bcopy to memset and memmove instead.
2. added a note to the CODING-GUIDELINES
3. add two macros to asterisk.h to prevent bzero, bcopy from creeping
back into the source
4. removed bzero from configure, configure.ac, autoconfig.h.in
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r143843 | mvanbaak | 2008-09-21 11:53:01 +0200 (Sun, 21 Sep 2008) | 3 lines
fix privacymanager example so it shows how to use the
PRIVACYMRGSTATUS variable
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https://origsvn.digium.com/svn/asterisk/trunk
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r143840 | mvanbaak | 2008-09-21 11:31:54 +0200 (Sun, 21 Sep 2008) | 3 lines
document the new context argument for privacymanager
so people can do pattern matching on the input
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r143837 | mvanbaak | 2008-09-21 11:27:08 +0200 (Sun, 21 Sep 2008) | 2 lines
fix privacy documentation. We no longer do priority jumping +101
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https://origsvn.digium.com/svn/asterisk/trunk
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r137987 | russell | 2008-08-14 17:43:15 -0500 (Thu, 14 Aug 2008) | 2 lines
Fix a bashism that causes an error when trying to build the pdf on ubuntu
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r137405 | seanbright | 2008-08-13 10:33:49 -0400 (Wed, 13 Aug 2008) | 1 line
Update docs to reflect the change to cdr_tds
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the user manual stuff in doc/tex/.
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memory allocator. See doc/hoard.txt for more details.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135405 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #13211)
Reported by: eliel
Patches:
fixtexerrors.patch uploaded by eliel (license 64)
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respectively. Also, take the opportunity to clean up the CLI prompt
generation code.
(closes issue #13175)
Reported by: eliel
Patches:
cliprompt.patch uploaded by eliel (license 64)
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(closes issue #13155)
Reported by: greenfieldtech
Patches:
app_voicemail.c.patch uploaded by greenfieldtech (license 369)
hebrew.ods uploaded by greenfieldtech (license 369)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@133904 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r132645 | oej | 2008-07-22 22:10:26 +0200 (Tis, 22 Jul 2008) | 9 lines
The most common question on the #asterisk iRC channel and on mailing lists
seems to be in regards to an error message when retransmit fails. This
is frequently misunderstood as a failure of Asterisk, not a failure of
the network to reach the other party.
This document tries to assist the Asterisk user in sorting out these
issues by explaining the logic and pointing at some possible
causes. Hopefully, we will get other questions now :-)
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To make sure nobody commits script-modified files we first make a backup
of asterisk.tex, run the script, generate the pdf and / or html,
and put the original asterisk.tex back.
This will guard us for the stuff that happened before that someone committed
a locally modified asterisk.tex, with changes done by this script.
(closes issue #13062)
Reported by: mvanbaak
Patches:
sed_without-i-v3.diff uploaded by mvanbaak (license 7)
Tested by: mvanbaak
Feedback from Corydon. Thanks for taking the time to go through this.
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This got lost in commit 97634
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in the documentation.
Patch provided by caio1982 (license 22)
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that we're
the reference implementation on now. :-)
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sizeof() and division.
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This section covers some locking fundamentals, as well as some information on
locking as it is used in Asterisk. It describes some of the ways that are used
and could be used to achieve deadlock avoidance. It also demonstrates the
unfortunate conclusion that with the use of recursive locks, none of the
constructs in use today are failsafe from deadlocks. Finally, it makes some
recommendations for new code being written. As proper locking strategies is a
complex subject, this section still has room for expansion and improvement.
This is a result of collaboration between Luigi Rizzo and myself on the
asterisk-dev mailing list.
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FreeTDS 0.60. Update the docs to reflect that we can now compile and run
against all modern releases of FreeTDS (0.60 through 0.82)
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wasn't working. :)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r124372 | jeffg | 2008-06-20 17:14:40 -0400 (Fri, 20 Jun 2008) | 1 line
Fix issues in digium-mib.txt and asterisk-mib.txt to placate smilint - bug 12905
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They want (char *)NULL as sentinel.
An example is OpenBSD (confirmed on 4.3) that ships with gcc 3.3.4
This commit introduces a contstant SENTINEL which is declared as:
#define SENTINEL ((char *)NULL)
All places I could test compile on my openbsd system are converted.
Update CODING-GUIDELINES to tell about this constant.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r123769 | tilghman | 2008-06-18 17:08:30 -0500 (Wed, 18 Jun 2008) | 8 lines
Add support for saying numbers in Hebrew.
(closes issue #11662)
Reported by: greenfieldtech
Patches:
say.c.patch-12042008 uploaded by greenfieldtech (license 369)
Hebrew-Sounds.ods uploaded by greenfieldtech
(with signficant changes to the spreadsheet by me)
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Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow.
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r121804 | jpeeler | 2008-06-11 11:11:09 -0500 (Wed, 11 Jun 2008) | 1 line
add instructions for logging gdb output via set logging on
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out from a branch
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This commit merges in the rest of the code needed to support distributed device
state. There are two main parts to this commit.
Core changes:
- The device state handling in the core has been updated to understand device
state across a cluster of Asterisk servers. Every time the state of a device
changes, it looks at all of the device states on each node, and determines the
aggregate device state. That resulting device state is what is provided to
modules in Asterisk that take actions based on the state of a device.
New module, res_ais:
- A module has been written to facilitate the communication of events between
nodes in a cluster of Asterisk servers. This module uses the SAForum AIS
(Service Availability Forum Application Interface Specification) CLM and EVT
services (Cluster Management and Event) to handle this task. This module
currently supports sharing Voicemail MWI (Message Waiting Indication) and
device state events between servers. It has been tested with openais, though
other implementations of the spec do exist.
For more information on testing distributed device state, see the following doc:
- doc/distributed_devstate.txt
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DUNDi uses a concept called the Entity ID for unique server identifiers. I have
pulled out the handling of EIDs and made it something available to all of Asterisk.
There is now a global Entity ID that can be used for other purposes as well, such
as code providing distributed device state, which is why I did this. The global
Entity ID is set automatically, just like it was done in DUNDi, but it can also be
set in asterisk.conf. DUNDi will now use this global EID unless one is specified
in dundi.conf.
The current EID for the system can be seen in the "core show settings" CLI command.
It is also available in the dialplan via the ENTITYID variable.
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r118052 | tilghman | 2008-05-23 07:59:16 -0500 (Fri, 23 May 2008) | 3 lines
Add information on using the Asterisk console, including tab command line
completion. (Closes issue #12681)
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(closes issue #12705)
Reported by: ctooley
Patches:
new_externalivr_argument_format-v2.diff uploaded by ctooley (license 136)
new_externalivr_documentation.diff uploaded by ctooley (license 136)
and a few additional fixes by me
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app_queue.
(closes issue #12591)
Reported by: CCHAsteria
Patches:
app_queue_RNA_event.diff uploaded by CCHAsteria (license 477)
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stop of monitoring (one-touch monitor).
Keep messages short, since the other party is waiting while one party hear the message...
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