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
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104419 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103725 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
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
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
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
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
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
- 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
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
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
- 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
- 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
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
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
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
* 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
entries in the queue log.
(issue #7561, reported and originally patched by fkasumovic, patch slightly
modified and updated to trunk by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62462 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Russell Bryant
Mark Michelson
Michiel van Baak
Jason Parker
Tilghman Lesher
Brett Bryant
Olle Johansson
Terry Wilson
Joshua Colp
Luigi Rizzo
Steve Murphy
Matthew Fredrickson
Kevin P. Fleming
Dwayne M. Hubbard