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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r107826 | qwell | 2008-03-11 18:37:05 -0500 (Tue, 11 Mar 2008) | 7 lines
Update documentation for pgsql ODBC voicemail.
(closes issue #12186)
Reported by: jsmith
Patches:
vm_pgsql_doc_update.patch uploaded by jsmith (license 15)
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an attended transfer over AMI
(closes issue #10585)
Reported by: ornati
Patches:
atxfer-trunk-r90428.diff uploaded by ornati (license 210)
(with modifications from me)
Tested by: putnopvut
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106236 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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configuration issues
(closes issue #12151)
Reported by: caio1982
Patches:
DB_metric3.diff uploaded by caio1982 (license 22)
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that if we keep this in the tree, it will be much easier to keep up to date.
The page on asterisk.org just links to this on svn.digium.com/view
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104418 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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2) Revise the argument parsing section, showing use of the standard macros.
3) Fix a typo.
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valid ciphers provided in both files. .. First commit since July, woot
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privilege to call out to a subshell.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r103722 | mmichelson | 2008-02-15 11:26:37 -0600 (Fri, 15 Feb 2008) | 8 lines
Final round of changes for configure script logic for IMAP
Now if a directory is specified, then we will search that directory for
a source installation of the IMAP toolkit. If none is found, then we will
use that directory as the basis for detecting a package installation of
the IMAP c-client. If that check fails, then configure will fail.
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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
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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.)
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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.
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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
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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.
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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)
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Closes issue #11645, patch by Laureano.
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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).
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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)
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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)
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Add number of total calls and current calls to SNMP.
Closes issue #10057, patch by jcmoore.
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version 1.1 - hopefully a more consistent manager interface.
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consistency's sake
(closes issue #11381, reported and patched by jon)
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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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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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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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greetingsfolder options
in voicemail.conf
(closes issue #11133, reported by selsky, patched by blitzrage)
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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
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SVN-group-chan_unistim-r88326M-/trunk
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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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