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(closes issue #17392)
Reported by: dps
Patches:
20100525__issue17392.diff.txt uploaded by tilghman (license 14)
Tested by: dps
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@265894 f38db490-d61c-443f-a65b-d21fe96a405b
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This commit adds support for calendaring with Exchange Server 2007+ via
Exchange Web Services. Full write support and for querying attendees. Many
thanks to Jan Kaláb for the feature.
(closes issue #17022)
Reported by: pitel
Patches:
res_calendar_ews.c uploaded by pitel (license 1008)
Tested by: pitel, twilson
Review: https://reviewboard.asterisk.org/r/557/
Review: https://reviewboard.asterisk.org/r/668/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@265317 f38db490-d61c-443f-a65b-d21fe96a405b
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directmediapermit/directmediadeny support to restrict which peers can do
directmedia based on ip address. In some networks not all phones are fully
routed, i.e. not all phones can ping each other. This patch adds a way to
restrict directmedia for certain peers between certain networks.
(closes issue #16645)
Reported by: raarts
Patches:
directmediapermit.patch uploaded by raarts (license 937)
Tested by: raarts
Review: https://reviewboard.asterisk.org/r/467/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@264626 f38db490-d61c-443f-a65b-d21fe96a405b
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Now that Asterisk modules can dynamically create and destroy logger levels
on demand, it's useful to be able to configure a logger channel (console,
file, whatever) to be able to accept log messages from *all* levels, even
levels created dynamically. This patch adds support for this, by allowing
the '*' level name to be used in logger.conf.
Review: https://reviewboard.asterisk.org/r/663/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@264160 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17359)
Reported by: alecdavis
Patches:
bug17359.diff.txt uploaded by alecdavis (license 585)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@264031 f38db490-d61c-443f-a65b-d21fe96a405b
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From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@263541 f38db490-d61c-443f-a65b-d21fe96a405b
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commented out.
This fixes some breakage in the test suite, that uses the contents of asterisk.conf
to discover the install layout on the system.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261364 f38db490-d61c-443f-a65b-d21fe96a405b
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This simply moves the functionality from the Makefile (cleaning it up) into an external
asterisk.conf.samples file. Also updates formatting (easier to read) and grammar
changes to asterisk.conf.samples.
(closes issue #17027)
Reported by: pabelanger
Patches:
0017027.asterisk.conf.v6.patch uploaded by pabelanger (license 224)
Tested by: qwell, lmadsen, pabelanger, chappell
Review: https://reviewboard.asterisk.org/r/616/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261124 f38db490-d61c-443f-a65b-d21fe96a405b
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in all queues.
See the CHANGES file and queues.conf.sample for more details.
(closes issue #17008)
Reported by: jlpedrosa
Patches:
queues.autopause_en_review.diff uploaded by jlpedrosa (license 1002)
Review: https://reviewboard.asterisk.org/r/581/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261051 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17263)
Reported by: pprindeville
Patches:
freenum-dialplan.patch#3 uploaded by pprindeville (license 347)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@260280 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r259270 | rmudgett | 2010-04-27 13:14:54 -0500 (Tue, 27 Apr 2010) | 14 lines
hidecalleridname parameter in chan_dahdi.conf
Issue #7321 implements a new chan_dahdi configuration option. However, a
change mentioned in the issue was never implemented. This is the change
that will allow the feature to work.
I added a note to chan_dahdi.conf.sample about the feature.
(closes issue #17143)
Reported by: djensen99
Patches:
diff.txt uploaded by djensen99 (license NA) (One line change)
Tested by: djensen99
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Minor tweaks and documentation added by me.
(closes issue #17058)
Reported by: pprindeville
Patches:
freenum.patch#5 uploaded by pprindeville (license 347)
Tested by: lmadsen
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258147 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@257768 f38db490-d61c-443f-a65b-d21fe96a405b
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caused an issue.
Application arguments within the feature map could possibly contain a comma,
which conflicts with the syntax of the features.conf configuration file. This
patch allows the argument to be wrapped in parentheses or quoted, to allow the
application arguments to be interpreted as a single configuration parameter.
(closes issue #16646)
Reported by: pinga-fogo
Patches:
20100414__issue16646.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
Review: https://reviewboard.asterisk.org/r/547/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@257262 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r257070 | mnicholson | 2010-04-13 11:46:30 -0500 (Tue, 13 Apr 2010) | 9 lines
Add an option to restore past broken behavor of the Events manager action
Before r238915, certain values for the EventMask parameter of the Events action would result in no response being returned. This patch adds an option to restore that broken behavior. Also while fixing this bug I discovered that passing an empty EventMasks parameter would also result in no response being returned, this has been fixed as well while being preserved when the broken behavior is requested.
(closes issue #17023)
Reported by: nblasgen
Review: https://reviewboard.asterisk.org/r/602/
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From Reviewboard:
CCSS stands for Call Completion Supplementary Services. An admittedly out-of-date
overview of the architecture can be found in the file doc/CCSS_architecture.pdf
in the CCSS branch. Off the top of my head, the big differences between what is
implemented and what is in the document are as follows:
1. We did not end up modifying the Hangup application at all.
2. The document states that a single call completion monitor may be used across
multiple calls to the same device. This proved to not be such a good idea
when implementing protocol-specific monitors, and so we ended up using one
monitor per-device per-call.
3. There are some configuration options which were conceived after the document
was written. These are documented in the ccss.conf.sample that is on this
review request.
For some basic understanding of terminology used throughout this code, see the
ccss.tex document that is on this review.
This implements CCBS and CCNR in several flavors.
First up is a "generic" implementation, which can work over any channel technology
provided that the channel technology can accurately report device state. Call
completion is requested using the dialplan application CallCompletionRequest and can
be canceled using CallCompletionCancel. Device state subscriptions are used in order
to monitor the state of called parties.
Next, there is a SIP-specific implementation of call completion. This method uses the
methods outlined in draft-ietf-bliss-call-completion-06 to implement call completion
using SIP signaling. There are a few things to note here:
* The agent/monitor terminology used throughout Asterisk sometimes is the reverse of
what is defined in the referenced draft.
* Implementation of the draft required support for SIP PUBLISH. I attempted to write
this in a generic-enough fashion such that if someone were to want to write PUBLISH
support for other event packages, such as dialog-state or presence, most of the effort
would be in writing callbacks specific to the event package.
* A subportion of supporting PUBLISH reception was that we had to implement a PIDF
parser. The PIDF support added is a bit minimal. I first wrote a validation
routine to ensure that the PIDF document is formatted properly. The rest of the
PIDF reading is done in-line in the call-completion-specific PUBLISH-handling
code. In other words, while there is PIDF support here, it is not in any state
where it could easily be applied to other event packages as is.
Finally, there are a variety of ISDN-related call completion protocols supported. These
were written by Richard Mudgett, and as such I can't really say much about their
implementation. There are notes in the CHANGES file that indicate the ISDN protocols
over which call completion is supported.
Review: https://reviewboard.asterisk.org/r/523
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@256528 f38db490-d61c-443f-a65b-d21fe96a405b
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sip.conf
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@255751 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17021)
Reported by: kovzol
Tested by: lmadsen, kovzol, davidw, ebroad
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@255504 f38db490-d61c-443f-a65b-d21fe96a405b
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This documentation associated wth tlsbindaddr is still useful so lets
synchronize it between trunk and 1.6.x branches.
(issue #17054)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@255066 f38db490-d61c-443f-a65b-d21fe96a405b
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Update some confusing documentation for the tlsbindaddr
option in sip.conf.sample. Point at a link instead which
has better documentation.
(closes issue #17054)
Reported by: klaus3000
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@255021 f38db490-d61c-443f-a65b-d21fe96a405b
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application is executing on a channel.
This patch addresses an issue found during working with end-users
using res_fax. If an incoming call is answered in the dialplan, or
jumps to the 'fax' extension due to reception of a CNG tone (with
faxdetect enabled), and then the remote endpoint sends a T.38
re-INVITE, it is possible for the channel's T.38 state to be
'T38_STATE_NEGOTIATING' when the application starts up. Unfortunately,
even if the application wants to use T.38, it can't respond to the
peer's negotiation request, because the AST_CONTROL_T38_PARAMETERS
control frame that chan_sip sent originally has been lost, and the
application needs the content of that frame to be able to formulate a
reply.
This patch adds a new 'request' type to AST_CONTROL_T38_PARAMETERS,
AST_T38_REQUEST_PARMS. If the application sends this request, chan_sip
will re-send the original control frame (with
AST_T38_REQUEST_NEGOTIATE as the request type), and the application
can respond as normal. If this occurs within the five second timeout
in chan_sip, the automatic cancellation of the peer reinvite will be
stopped, and the application will 'own' the negotiation process from
that point onwards.
This also improves the code path in chan_sip to allow sip_indicate(),
when called for AST_CONTROL_T38_PARAMETERS, to be able to return a
non-zero response, which should have been in place before since the
control frame *can* fail to be processed properly. It also modifies
ast_indicate() to return whatever result the channel driver returned
for this control frame, rather than converting all non-zero results
into '-1'. Finally, the new request type intentionally returns a
positive value, so that an application that sends
AST_T38_REQUEST_PARMS can know for certain whether the channel driver
accepted it and will be replying with a control frame of its own, or
whether it was ignored (if the sip_indicate()/ast_indicate() path had
properly supported failure responses before, this would not be
necessary).
This patch also modifies res_fax to take advantage of the new request.
In addition, this patch makes sip_t38_abort() actually lock the
private structure before doing its work... bad programmer, no donut.
This patch also enhances chan_sip's 'faxdetect' support to allow
triggering on T.38 re-INVITEs received as well as CNG tone detection.
Review: https://reviewboard.asterisk.org/r/556/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@254450 f38db490-d61c-443f-a65b-d21fe96a405b
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Previously only configurable globally. A unit test has also been written to
provide protection against parse failures for supported mailbox options.
(closes issue #16864)
Reported by: kobaz
Patches:
voicemail2.patch uploaded by kobaz (license 834)
Review: https://reviewboard.asterisk.org/r/555/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@254321 f38db490-d61c-443f-a65b-d21fe96a405b
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mmichelson's feedback.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@253712 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r253018 | lmadsen | 2010-03-16 19:26:19 -0500 (Tue, 16 Mar 2010) | 6 lines
Add french snipset to say.conf.
Add the french snipset to say.conf.
(Closes issue #15799)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r252761 | lmadsen | 2010-03-16 13:46:20 -0500 (Tue, 16 Mar 2010) | 7 lines
Additional extensions.ael global variable fixes.
Fixing up a couple more overlapping global variable namespaces shared with
extensions.conf.sample. Also noticed a few of the lines that were commented
out didn't have the closing semi-colon so I added that as well.
(issue #17035)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r252533 | lmadsen | 2010-03-15 15:48:56 -0500 (Mon, 15 Mar 2010) | 7 lines
Update extensions.ael file to not overlap extensions.conf.
Updated the extensions.ael file so the global variables don't overlap
those that we have in extensions.conf (sample files). This way unexpected
things won't happed hopefully if both pbx_ael and res_config are loaded.
(closes issue #17035)
Reported by: pprindeville
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added response to roundtrip delay requests from opposite side
added roundtrip delay request sending to opposite side after answer,
added options for sending request (interval between request and
count of unreplied requests before forced call hangup)
(closes issue #16976)
Reported by: vmikhelson
Patches:
rtdr-1.6.0-2.patch uploaded by may213 (license 454)
Tested by: vmikhelson, may213
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@252277 f38db490-d61c-443f-a65b-d21fe96a405b
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This change basically reverts the change reviewed in
https://reviewboard.asterisk.org/r/374/ and instead limits the
updating of the RTP synchronization source to only those times when we
detect that the other side of the conversation has changed the ssrc.
The problem is that SRCUPDATE control frames are sent many times where
we don't want a new ssrc, including whenever Asterisk has to send DTMF
in a normal bridge. This is also not the first time that this mistake
has been made. The initial implementation of the ast_rtp_new_source
function also changed the ssrc--and then it was removed because of
this same issue. Then, we put it back in again to fix a different
issue. This patch attempts to only change the ssrc when we see that
the other side of the conversation has changed the ssrc.
It also renames some functions to make their purpose more clear.
Review: https://reviewboard.asterisk.org/r/540/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@252089 f38db490-d61c-443f-a65b-d21fe96a405b
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New config parameter "reportalarms" added in chan_dahdi.conf which supports the
following possible values:
"channels": report each channel alarms (current behavior, default for backward compatibility)
"spans": report an "SpanAlarm" event when the span of any configured channel is alarmed
"all": report channel and span alarms (aggregated behavior)
"none": do not report any alarms
(closes issue #16709)
Reported by: nahuelgreco
Patches:
chan_dahdi.c.reportalarms.patch uploaded by nahuelgreco (license 162)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250392 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r250043 | lmadsen | 2010-03-02 15:51:35 -0500 (Tue, 02 Mar 2010) | 7 lines
Update documentation to clarify purpose of unanswered option.
(closes issue #16267)
Reported by: elsto
Patches:
cdr.conf.sample.patch.txt uploaded by lmadsen (license 10)
Tested by: davidw, elsto
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When configuring the adaptive jitterbuffer, the target_extra
value not only could not be set from the configuration, but was
not even being set to its proper default. This value is required
in order for the adaptive jitterbuffer to work correctly. To resolve
this a config option has been added to expose this value to the conf
files, and a default value is provided when no config specific value
is present.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249893 f38db490-d61c-443f-a65b-d21fe96a405b
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This feature allows for parkinglots to be created dynamically within
the dialplan. Thanks to all who were involved with getting this patch
written and tested!
(closes issue #15135)
Reported by: IgorG
Patches:
features.dynamic_park.v3.diff uploaded by IgorG (license 20)
2009090400_dynamicpark.diff.txt uploaded by mvanbaak (license 7)
dynamic_parkinglot.diff uploaded by dvossel (license 671)
Tested by: eliel, IgorG, acunningham, mvanbaak, zktech
Review: https://reviewboard.asterisk.org/r/352/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@247248 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r245944 | tilghman | 2010-02-10 07:37:13 -0600 (Wed, 10 Feb 2010) | 2 lines
Include examples of FILTER usage in extension patterns where a "." may be a risk.
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First off, these options weren't actually doing anything.
By the time the options were parsed, the peer and dialog
containers had already been allocated with their default
values.
Second, hash table size is something that doesn't really
make sense to change in a config file. If a user is that
interested in changing the hashtable size, he can modify
the source itself.
I have removed the parsing of the hash_peer, hash_user,
and hash_dialog options. I have removed the hash_user_size
variable altogether since it is not used at all. I also
changed hash_peer_size and hash_dialog_size to be constant,
and have changed the symbols to be in all caps as constants
typically are. I have also removed the entire section in
sip.conf.sample regarding configurable hashtable sizes.
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functionality and provide commonly useful features.
(closes issue #16090)
Reported by: pprindeville
Patches:
extensions.conf-bugid16090.patch#3 uploaded by pprindeville (license 347)
Tested by: tzafrir, pprindeville, lmadsen
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@239834 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16411)
Reported by: stanusr
Patches:
__20091210-sip.conf.sample-documentation.txt uploaded by lmadsen (license 10)
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(typically, UTC)
(closes issue #16401)
Reported by: lordmortis
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2. Added service type ported number query.
3. Formated code.
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The option is global and currently the acceptable values as noted in the sample
config are accept or deny.
(closes issue #15228)
Reported by: lp0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@235342 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r235181 | jsmith | 2009-12-15 15:07:55 -0600 (Tue, 15 Dec 2009) | 4 lines
Add a line showing that we can use CIDR notation.
patch by jsmith, after discussion with jtodd
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@235298 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #15132)
Reported by: floletarmo
Patches:
voicemail_changes.patch uploaded by floletarmo (license 784)
(with some additional changes by me)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@234820 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r233279 | dvossel | 2009-12-04 15:54:01 -0600 (Fri, 04 Dec 2009) | 7 lines
clarify requirecalltoken option in iax.sample.conf
(closes issue #16223)
Reported by: bklang
Patches:
clarify-iax-requirecalltoken.patch uploaded by bklang (license 919)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@233280 f38db490-d61c-443f-a65b-d21fe96a405b
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members when enabled
(closes issue #14559)
Reported by: fiddur
Patches:
trunk-199584-1.diff uploaded by fiddur (license 678)
Tested by: fiddur, dvossel
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@233196 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16263)
Reported by: andrew
Patches:
pagerdate.patch uploaded by andrew (license 240)
(with a slight modification by me)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@232916 f38db490-d61c-443f-a65b-d21fe96a405b
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