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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@100679 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@100422 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #5768)
Reported by: mguesdon
Patches:
res_config_ldap-v0.7.tar.gz uploaded by mguesdon (license 121)
res_ldap.conf.sample uploaded by suretec (license 70)
asterisk-v3.1.4.ldif uploaded by suretec (license 70)
asterisk-v3.1.4.schema uploaded by suretec (license 70)
Tested by: oej, mguesdon, suretec, cthorner
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99696 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99647 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99482 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This set of changes introduces TCP and TLS support for chan_sip. There are various
new options in configs/sip.conf.sample that are used to enable these features. Also,
there is a document, doc/siptls.txt that describes some things in more detail.
This code was implemented by Brett Bryant and James Golovich. It was reviewed
by Joshua Colp and myself. A number of other people participated in the testing
of this code, but since it was done outside of the bug tracker, I do not have their
names. If you were one of them, thanks a lot for the help!
(closes issue #4903, but with completely different code that what exists there.)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99085 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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the HTTP request for the config came in on and the SERVER_PORT to the
bindport setting in sip.conf. I've left in the ability to override these
options, because I can't always guess how someone might decide to do something
weird with what is available to them--although needing to is pretty unlikely.
Documentation was updated to reflect preference for not setting serveraddr,
serveriface, or serverport. Tested on Linux and OS X.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #11763)
Reported by: IgorG
Patches:
docupd.v1.diff uploaded by IgorG (license 20)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98695 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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run "make asterisk.pdf" when not all of the right packages are installed.
(closes issue #10763)
Reported by: Corydon76
Patches:
20070919__bug10763.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98454 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98152 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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
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go a long way towards preventing unexplainable hangs experienced by people. In the
case of MWI hangs, this also will mean that the SIP port isn't blocked anymore.
(closes issue #11665, reported by yehavi)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96934 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Closes issue #11645, patch by Laureano.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95070 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94903 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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adjust
the QUEUE_MAX_PENALTY and the newly introduced QUEUE_MIN_PENALTY during a call depending
on the amount of time passed. The purpose is to allow the call to open up to more (or maybe
just different) members without the caller's losing his place in the queue. See
configs/queuerules.conf.sample for an example of how to set up queue rules and configs/queues.conf.sample
for how to associate a rule with a queue.
Along with the functional changes, new CLI and manager commands exist to show the rules defined and
there is an additional CLI command to reload the queue rules.
Future enhancements that may be made: support for realtime queue rules and support for dynamically adding
a rule through the manager or CLI. Also a manager command to reload the queue rules (I'll probably write
this myself very soon).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94370 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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instead of pointing
to configuration file with -C
Reported by: sobomax
Patches:
asterisk.c.diff.trunk uploaded by sobomax (license 359)
doc changes by committer
(closes issue #11598)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93854 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93557 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Refer to the proper documentation
- Implement separate signalling/media QoS/CoS in many channels using RTP
- Improve warnings and verbose messages
- Deprecate some old settings
Minor modifications by me, a big effort from IgorG.
Thanks!
Reported by: IgorG
Patches:
qoscleanup-89394-4-trunk.patch uploaded by IgorG (license 20)
Tested by: IgorG
(closes issue #11145)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93163 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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manager 1.1. Update documentation too.
(closes issue #11495)
Reported by: caio1982
Patches:
extra_vm_manager_info1.diff uploaded by caio1982 (license 22)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92140 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92084 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Add number of total calls and current calls to SNMP.
Closes issue #10057, patch by jcmoore.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91779 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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version 1.1 - hopefully a more consistent manager interface.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91438 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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consistency's sake
(closes issue #11381, reported and patched by jon)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89621 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89526 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89450 | murf | 2007-11-20 08:22:08 -0700 (Tue, 20 Nov 2007) | 1 line
closes issue #11324; break statements missing in switch cases.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89416 | file | 2007-11-19 11:24:12 -0400 (Mon, 19 Nov 2007) | 4 lines
Clarify documentation a bit, include that a frame has to pass through the core in order for the Local channel optimization to happen.
(closes issue #11246)
Reported by: jon
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89417 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89103 | tilghman | 2007-11-07 22:55:19 -0600 (Wed, 07 Nov 2007) | 2 lines
Typo
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89104 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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greetingsfolder options
in voicemail.conf
(closes issue #11133, reported by selsky, patched by blitzrage)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89075 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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much identical to the S() and L() options to Dial(). They let you set
timeouts for the conference, as well as have warning sounds played to
let the caller know how much time is left, and when it is running out.
(closes issue #8030)
Reported by: areski
Patches:
meetme_timeout_timelimit_v2.patch uploaded by areski (license 29)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89069 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This introduces a new channel driver, chan_unistim, that supports the Unistim
VoIP protocol for Nortel phones. The following models have been confirmed
to work: i2002, i2004 and i2050.
(closes issue #8864)
Reported by: c_hans
Patches:
chan_unistim.patch uploaded by c (license 304)
ustm_no_conf.diff uploaded by junky (license 177)
Tested by: c_hans, dbowerman, math, junky, loloski
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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SVN-group-chan_unistim-r88326M-/trunk
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88327 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r88116 | tilghman | 2007-11-01 12:17:56 -0500 (Thu, 01 Nov 2007) | 2 lines
Add some notes on using valgrind
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88117 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Remove old unused defines for old style.
Closes issue 10860, patch by IgorG.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85764 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85683 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Many uses of the astlisting environment around verbatim text to ensure that
it gets properly formatted and doesn't run off the page.
- Update some things that have been deprecated.
- Add escaping as needed
- and more ...
(closes issue #10978)
Reported by: IgorG
Patches:
texdoc-85542-1.patch uploaded by IgorG (license 20)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85547 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Here it is.
(closes issue #10962)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85539 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85526 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r85523 | tilghman | 2007-10-12 13:30:55 -0500 (Fri, 12 Oct 2007) | 2 lines
Change Digium address
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85524 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- 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
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in the Dial command. The 'j' option _must_ be used in conjunction with the 'n'
option.
This feature will allow you to use the existing jitterbuffer implementation to
put a jitterbuffer on incoming SIP calls connecting to Asterisk applications by
putting a local channel in the middle.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85097 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@84580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@83834 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@83574 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@83499 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@83278 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #10738, rbraun_proformatique)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82679 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r82376 | mmichelson | 2007-09-14 09:42:29 -0500 (Fri, 14 Sep 2007) | 5 lines
Fixing a typo in the coding guidelines
(closes issue #10717, reported and patched by leedm777)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82377 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #10711, rbraun_proformatique)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82321 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Reported by: nic_bellamy
Patches:
2006-10-03_svn_44249_voicemail_lockmode_v3.patch uploaded by nic_bellamy (license 213)
Add support for configurable file locking methods. The default is "lockfile",
which is the old behavior. There is an additional option, "flock", which is
intended for use in situations where the lockfile method will not work, such as
with SMB/CIFS mounts.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@81233 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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