It was pretty sparsely documented.
This update fleshes out the pbx_lua module, to
add the switch statements to the extensions in the
extensions.lua file, as well as removing them when
the module is unloaded.
Many thanks to Matt Nicholson for his fine
contribution!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@144523 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r142865 | tilghman | 2008-09-12 15:37:18 -0500 (Fri, 12 Sep 2008) | 11 lines
Create rules for disallowing contacts at certain addresses, which may
improve the security of various installations. As this does not change
any default behavior, it is not classified as a direct security fix for
anything within Asterisk, but may help PBX admins better secure their
SIP servers.
(closes issue #11776)
Reported by: ibc
Patches:
20080829__bug11776.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76, blitzrage
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142866 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: erousseau
This was a reasonable enhancement request, which was
easy to implement. Since it's an enhancement, it
could only be applied to trunk.
Basically, for accounting where "initiated" seconds
are billed for, if the microseconds field on the end
time is greater than the microseconds field for the
answer time, add one second to the billsec field.
The implementation was requested by erousseau, and
I've implemented it as requested. I've updated the
CHANGES, the cdr.conf.sample, and the .h files
accordingly, to accept and set a flag for the
corresponding new option. cdr.c adds in the extra
second based on the usec fields if the option is
set. Tested, seems to be working fine.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@140057 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Made bearer2str() use allowed_bearers_array[]
* Made use the causes.h defines instead of hardcoded numbers.
* Made use Asterisk presentation indicator values if either of the
mISDN presentation or screen options are negative.
* Updated the misdn_set_opt application option descriptions.
* Renamed the awkward Caller ID presentation misdn_set_opt
application option value not_screened to restricted.
Deprecated the not_screened option value.
channels/misdn/isdn_lib.c
* Made use the causes.h defines instead of hardcoded numbers.
* Fixed some spelling errors and typos.
* Added all defined facility code strings to fac2str().
channels/misdn/isdn_lib.h
* Added doxygen comments to struct misdn_bchannel.
channels/misdn/isdn_lib_intern.h
* Added doxygen comments to struct misdn_stack.
channels/misdn_config.c
configs/misdn.conf.sample
* Updated the mISDN presentation and screen parameter descriptions.
doc/tex/misdn.tex
* Updated the misdn_set_opt application option descriptions.
* Fixed some spelling errors and typos.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@138738 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r137731 | russell | 2008-08-14 09:05:23 -0500 (Thu, 14 Aug 2008) | 4 lines
Comments in this config file were aligned only if your tab size was set to 8.
So, convert tabs to spaces so that things should be aligned regardless of what
tab size you use in your editor.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137732 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This set of changes enhances IAX2 encryption support by adding key rotation
to provide enhanced security. The key used for encryption is rotated right
after the call gets set up, and then again every few minutes. This was
discussed at the last AstriDevCon. For interoperability with older versions
of Asterisk, there is an option that disables key rotation.
(closes issue #13018)
Reported by: bbryant
Patches:
07072008__iax2_key_rotation.diff uploaded by bbryant (license 36)
Tested by: russell, bbryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
such as "Work" and "Family" would be subfolders of the
INBOX. This is an invalid assumption to make, but it could
be desirable to set up folders in this manner, so a new
option for voicemail.conf, "imapparentfolder" has been
added to allow for this.
(closes issue #13142)
Reported by: jaroth
Patches:
parentfolder.patch uploaded by jaroth (license 50)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
driver into a common place for multiple channel drivers.
(closes issue #13152)
Reported by: caio1982
Patches:
atxfer_complete_sound3.diff uploaded by caio1982 (license 22)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134401 65c4cc65-6c06-0410-ace0-fbb531ad65f3
fail to setup video RTP if the two endpoints will not support it. This assists
with call files and certain transfers to ensure that if two video phones are
ever connected, they will always share a video feed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r130039 | kpfleming | 2008-07-11 10:41:56 -0500 (Fri, 11 Jul 2008) | 4 lines
add support for a configuration parameter for 'inband audio during RELEASE', which is currently mandatory in libpri-1.4.4 but will become configurable in libpri-1.4.5 later today
(related to issue #13042)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130040 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Note: I don't think we can start properly without UDP port open, that needs to be tested.
- Removing "bindport" from configuration example, not needed to mention this any more
I suggest we deprecate "bindaddr" and "bindport" in trunk (for 1.6.1)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Adding IP address for TCP and/or TLS too if auto-domain is enabled and
binding to a different IP address
- Fixing documentation in sip.conf.sample
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128524 65c4cc65-6c06-0410-ace0-fbb531ad65f3
acceptdtmf and enddtmf. These allow for the DTMF pressed to be configurable
instead of being hardcoded to '#' and '*'.
(AST-86)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127558 65c4cc65-6c06-0410-ace0-fbb531ad65f3
for new dialogs, and the rest listed will be acceptable ways for that peer to contact us. This fixes a minor bug where, because SIP TCP/UDP run on
the same port, could cause a TCP peer to be saved in the ast_db. There will also be warnings when a transport is changed for an unexpected reason.
(issue #12799)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125891 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
and off for new installations. This includes the translation from pipes to commas
for pbx_realtime and the EXEC command for AGI, as well as the change to the Set
application not to support multiple variables at once.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120171 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r118646 | file | 2008-05-28 11:23:34 -0300 (Wed, 28 May 2008) | 4 lines
Add an option to use the source IP address of RTP as the destination IP address of UDPTL when a specific option is enabled. If the remote side is properly configured (ports forwarded) then UDPTL will flow.
(closes issue #10417)
Reported by: cstadlmann
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@118647 65c4cc65-6c06-0410-ace0-fbb531ad65f3
'unknown', and better document the use of each parameter.
(closes issue #12633)
Reported by: tzafrir
Patches:
pridialplan_unknown_2.diff uploaded by tzafrir (license 46)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117182 65c4cc65-6c06-0410-ace0-fbb531ad65f3
to announce-position, "limit" and "more," as well as a new option,
announce-position-limit. For more information on the use of these options,
see CHANGES or configs/queues.conf.sample.
(closes issue #10991)
Reported by: slavon
Patches:
app_q.diff uploaded by slavon (license 288)
Tested by: slavon, putnopvut
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114906 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: oej
Tested by: jpeeler
This patch implements multiple parking lots for parked calls. The default parkinglot is used by default, however setting the channel variable PARKINGLOT in the dialplan will allow use of any other configured parkinglot. See configs/features.conf.sample for more details on setting up another non-default parkinglot. Also, one can (currently) set the default parkinglot to use in the driver configuration file via the parkinglot option.
Patch initially written by oej, brought up to date and finalized by mvanbaak, and then stabilized and converted to astobj2 by me.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114487 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r113874 | tilghman | 2008-04-09 13:57:33 -0500 (Wed, 09 Apr 2008) | 4 lines
If the [csv] section does not exist in cdr.conf, then an unload/load sequence
is needed to correct the problem. Track whether the load succeeded with a
variable, so we can fix this with a simple reload event, instead.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113875 65c4cc65-6c06-0410-ace0-fbb531ad65f3
allow the list of periodic announcments specified to be played in a random
order instead of being played sequentially.
(closes issue #6681)
Reported by: alt_phil
Tested by: putnopvut
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109621 65c4cc65-6c06-0410-ace0-fbb531ad65f3
aastra-check-cfg is the same as the other check-cfg entries,
and aastra-xml is to load a pre-configured xml script.
(closes issue #12229)
Reported by: gowen72
Patches:
aastra.patch uploaded by gowen72 (license 432)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109111 65c4cc65-6c06-0410-ace0-fbb531ad65f3
(closes issue #9503)
Reported by: tzafrir
Patches:
fix_cleanups uploaded by tzafrir (license 46)
zapata_sections uploaded by tzafrir (license 46)
skipchannel_options uploaded by tzafrir (license 46)
conf_sample uploaded by tzafrir (license 46)
patches updated by me to better conform to coding guidelines and fix some problems
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108286 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r104119 | russell | 2008-02-25 18:25:29 -0600 (Mon, 25 Feb 2008) | 33 lines
Merge changes from team/russell/smdi-1.4
This commit brings in a significant set of changes to the SMDI support in Asterisk.
There were a number of bugs in the current implementation, most notably being that
it was very likely on busy systems to pop off the wrong message from the SMDI message
queue. So, this set of changes fixes the issues discovered as well as introducing
some new ways to use the SMDI support which are required to avoid the bugs with
grabbing the wrong message off of the queue.
This code introduces a new interface to SMDI, with two dialplan functions. First,
you get an SMDI message in the dialplan using SMDI_MSG_RETRIEVE() and then you access
details in the message using the SMDI_MSG() function. A side benefit of this is that
it now supports more than just chan_zap.
For example, with this implementation, you can have some FXO lines being terminated
on a SIP gateway, but the SMDI link in Asterisk.
Another issue with the current implementation is that it is quite common that the
station ID that comes in on the SMDI link is not necessarily the same as the Asterisk
voicemail box. There are now additional directives in the smdi.conf configuration
file which let you map SMDI station IDs to Asterisk voicemail boxes.
Yet another issue with the current SMDI support was related to MWI reporting over
the SMDI link. The current code could only report a MWI change when the change
was made by someone calling into voicemail. If the change was made by some other
entity (such as with IMAP storage, or with a web interface of some kind), then the
MWI change would never be sent. The SMDI module can now poll for MWI changes if
configured to do so.
This work was inspired by and primarily done for the University of Pennsylvania.
(also related to issue #9260)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
the saying of less-than for holdtime announcements since it can lead to awkward holdtime announcements. Using
'1' as a queue-round-seconds value is no longer valid.
(closes issue #9736)
Reported by: caio1982
Patches:
queue_announce5.diff uploaded by caio1982 (license 22)
Tested by: caio1982, putnopvut
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103687 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #11875)
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r101219 | qwell | 2008-01-30 09:34:37 -0600 (Wed, 30 Jan 2008) | 5 lines
Change default config to use descending channel order of groups, rather than ascending.
Fixes a potential source of confusion in glare-type situations.
Issue 11875, reported by JimVanM.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@101220 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r99341 | tilghman | 2008-01-21 12:11:07 -0600 (Mon, 21 Jan 2008) | 8 lines
Permit the user to specify number of seconds that a connection may remain idle,
which fixes a crash on reconnect with the MyODBC driver.
(closes issue #11798)
Reported by: Corydon76
Patches:
20080119__res_odbc__idlecheck.diff.txt uploaded by Corydon76 (license 14)
Tested by: mvanbaak
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99350 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Add support for multiple devices. All devices are configured in console.conf.
- Add "console list devices" CLI command to show configured devices. Also, changed
the old "list devices" to be "list available", which queries PortAudio for all
audio devices that are available for use.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99227 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
This set of changes introduces SIP session timers support (RFC 4028). In short,
this prevents stuck SIP sessions that were not properly torn down due to network
or endpoint failures during an established SIP session.
To quote some of the documentation supplied with the patch:
"The SIP Session-Timers is an extension of the SIP protocol that allows end-points and proxies to
refresh a session periodically. The sessions are kept alive by sending a RE-INVITE or UPDATE
request at a negotiated interval. If a session refresh fails then all the entities that support Session-
Timers clear their internal session state. In addition, UAs generate a BYE request in order to clear
the state in the proxies and the remote UA (this is done for the benefit of SIP entities in the path
that do not support Session-Timers)."
(closes issue #10665)
Reported by: rjain
Patches:
chan_sip.c.1.diff uploaded by rjain (license 226)
chan_sip.c.diff uploaded by rjain (license 226)
sip.conf.sample.diff uploaded by rjain (license 226)
proc_422_rsp_comment.diff uploaded by rjain (license 226)
chan_sip.c.cache.diff uploaded by rjain (license 226)
chan_sip.memalloc uploaded by rjain (license 226)
chan_sip.memalloc.bugfix uploaded by rjain (license 226)
Patches tracked in team/group/sip_session_timers, with some additional fixes
by russell and oej.
Tested by: jtodd, rjain, loloski
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98978 65c4cc65-6c06-0410-ace0-fbb531ad65f3
to set the qualify frequency.
(closes issue #11597)
Reported by: wilder
Patches:
qualifyfreq5.patch uploaded by wilder (license 362)
-- with some mods by me
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98027 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1) Add the Dialplan class, for NewExten and VarSet events, which should cut
down on the volume of traffic in the Call class.
2) Permit some commands to be run from multiple classes, such as allowing
DBGet to be run from either the System or the Reporting class.
3) Heavily document each class in the sample config, as there were several
that made no sense to be in the write= line, and two that made no sense to be
in the read= line (since they controlled no permissions there).
(Closes issue #10386)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97651 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
will monitor this second device's state for the member, even though it actually calls the first
interface. This ability has been added for statically defined queue members, realtime queue members,
and dynamic queue members added through the CLI, dialplan, or manager.
(closes issue #11603, reported by acidv)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97203 65c4cc65-6c06-0410-ace0-fbb531ad65f3
(this implementation is not very memory efficient as the parameters and their values will be duplicated for each channel that has the same settings, but we can worry about that later once it is working)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96019 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Add a new console channel driver, chan_console, which is a console channel
driver that uses portaudio as a cross platform audio interface. It was written
to provide a console channel driver that works with Mac CoreAudio, but it
supports a number of other audio interfaces, as well, including OSS and ALSA.
It could one day be the single console channel driver, but does not yet have
as many features as chan_oss.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95412 65c4cc65-6c06-0410-ace0-fbb531ad65f3
from 'keypad_entry' to 'region'. Fix the example file accordingly.
Also make some fixes in the code do reset entries on reload of the keypad.
The recently committed kpad2.jpg has the correct names.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94638 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
by adding an exclamation mark to the dial string.
This patch also exists for 1.4 in the fixtoheader-1.4 branch
and has been in production for quite some time.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93897 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
This introduces two new options for zapata.conf: mwimonitor and mwimonitornotify.
The mwimonitor option enables MWI monitoring. When the MWI state on a line changes,
then the script specified by mwimonitornotify will be executed for custom handling
of the state change, similar to the externnotify option of voicemail.conf.
Also, when the MWI state on an FXO line changes, an internal Asterisk event is
generated to indicate the new state of the associated mailbox. That may, any
module that cares about MWI information will get notified and can handle it
just as if app_voicemail had sent this notification.
(BE-253, original patch from markster, with some minor modifications by me to
add comments, documentation, and internal event support)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90949 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This works in much the same way as the automonitor, except that instead of using the monitor
app, it uses the mixmonitor app. By providing an 'x' or 'X' as a dial or queue option, a DTMF
sequence may be entered (as defined in features.conf) to start the one-touch mixmonitor.
This patch also introduces some new API calls to the audiohooks code for searching for an audiohook
by type and for searching for a running audiohook by type.
Big thanks to joetester for writing the initial patch, testing it and patiently waiting for it to
be committed.
(closes issue #10185, reported and patched by xmarksthespot)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90388 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1. When moh is started, we search first in memory to find the class. If we do not
find it in memory, we search realtime instead.
2. When moh is restarted (as in, it had been started on this particular channel, stopped,
and now we're starting it again), if using the "files" mode, then realtime will always
be rechecked. If you are using other modes, however, we will simply reattach to the external
running process which was playing moh earlier in the call. This is a necessary compromise so that
we don't end up with too many background processes.
3. musiconhold.conf has a general section now. It has one option: cachertclasses. If set to yes,
then moh classes found in realtime will be added to the in-memory list. This has the advantage
of not requiring database lookups each time moh is started, but it has the disadvantage of not
truly being realtime.
I have tested this for functionality, and it passes. I also tested this under valgrind and there
are no memory problems reported under typical use.
Special thanks to Sergee for implementing this feature and enduring my complaints on the bugtracker!
(closes issue #11196, reported and patched by sergee)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89946 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89622 | murf | 2007-11-26 23:24:02 -0700 (Mon, 26 Nov 2007) | 1 line
closes issue #11379; OK, this is an attempt to make both sides happy. To the cdr.conf file, I added the option 'unanswered', which defaults to 'no'. In this mode, you will see a cdr for a call, whether it was answered or not. The disposition will be NO ANSWER or ANSWERED, as appropriate. The src is as you'd expect, the destination channel will be one of the channels from the Dial() call, usually the last in the list if more than one chan was specified. With unanswered set to 'yes', you will still see this cdr entry in both cases. But in the case where the dial timed out, you will also see a cdr for each line attempted, marked NO ANSWER, with no destination channel name. The new option defaults to 'no', so you don't see the pesky extra cdr's by default, and you will not see the irritating 'not posted' messages.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89559 | tilghman | 2007-11-25 11:17:10 -0600 (Sun, 25 Nov 2007) | 14 lines
We previously attempted to use the ESCAPE clause to set the escape delimiter to
a backslash. Unfortunately, this does not universally work on all databases,
since on databases which natively use the backslash as a delimiter, the
backslash itself needs to be delimited, but on other databases that have no
delimiter, backslashing the backslash causes an error.
So the only solution that I can come up with is to create an option in res_odbc
that explicitly specifies whether or not backslash is a native delimiter. If
it is, we use it natively; if not, we use the ESCAPE clause to make it one.
Reported by: elguero
Patch by: tilghman
(Closes issue #11364)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89561 65c4cc65-6c06-0410-ace0-fbb531ad65f3
and we now have the groupcount system to implement call-limits in the dialplan. You
can use the "setvar" option in realtime/sip.conf to set limits per device.
- Implement "callcounter" as a new option to enable the call counting we need to
report device status to queue, manager and SIP subscriptions.
The call counter setting is now enabled in the code by setting the device call-limit
to 999. When we remove the call limit, we can simply enable this with a boolean
setting.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89554 65c4cc65-6c06-0410-ace0-fbb531ad65f3
way of handling DTMF in SIP. Totally undocumented, but implemented
in enough devices so we have to support it.
Code by sergee, small changes by oej.
Closes issue #11049
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89278 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89173 | crichter | 2007-11-12 12:26:48 +0100 (Mo, 12 Nov 2007) | 1 line
if we're NT and no number was dialed and overlapdial is set, we wait for the ISDN timeout instead of starting our own timer. added a comment for the misdn.conf.sample for the overlapdial config option.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89169 | crichter | 2007-11-12 10:45:36 +0100 (Mo, 12 Nov 2007) | 1 line
aded ntkeepcalls option, to avoid droÃpping calls when the L2 goes down on a PTP link. There are some pbx which do turn off the L1 for a very short while and restart it immediately. normally T310 should be started and after 10 seconds or so the calls should be dropped, this is a simple fix wihtout this timer.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89174 65c4cc65-6c06-0410-ace0-fbb531ad65f3
It was mistakenly deleted in 1.4 without ever being merged to trunk.
Reported by eliel on #asterisk-dev.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89132 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
phones to be rung in a specific order.
(closes issue #7279, reported and initially patched by diLLec, patch reworked by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@87154 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: ruffle
Patches:
app_voicemail.c.diff uploaded by ruffle (license 201)
10739-moveheard.diff uploaded by qwell (license 4)
Tested by: callguy, ruffle
Add an option to disable the automatic moving of "heard" messages to the Old folder.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82871 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: snar
Patches:
app-queue-cdr-trunk.patch uploaded by snar (license 245)
queues.conf.patch uploaded by snar (license 245)
Add an updatecdr option to queues.conf, so that if a "member name" is specified,
the cdr record will be updated with that, rather than the channel.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82800 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: junky
Patches:
register_trying.diff.txt uploaded by jcmoore
Disable sending 100 Trying on REGISTER attempts and make it an option. This has been signed off by oej.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82257 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Steve Murphy
Tilghman Lesher
Philippe Sultan
Richard Mudgett
Mark Michelson
Sean Bright
Russell Bryant
Kevin P. Fleming
Brett Bryant
Olle Johansson
Matthew Fredrickson
Jeff Peeler
Joshua Colp
Luigi Rizzo
Jason Parker
Claude Patry
Terry Wilson
Christian Richter
Dwayne M. Hubbard