Asterisk now has protocol independent support for processing text messages
outside of a call. Messages are routed through the Asterisk dialplan.
SIP MESSAGE and XMPP are currently supported. There are options in sip.conf
and jabber.conf that enable these features.
There is a new application, MessageSend(). There are two new functions,
MESSAGE() and MESSAGE_DATA(). Documentation will be available on
the project wiki, wiki.asterisk.org.
Thanks to Terry Wilson for the assistance with development and to David Vossel
for helping with some additional testing.
Review: https://reviewboard.asterisk.org/r/1042/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321546 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r321330 | rmudgett | 2011-05-27 16:31:25 -0500 (Fri, 27 May 2011) | 8 lines
The app_privacy args have undocumented "options" position, interferes with "context" position.
* Add documention for unused "options" position to match existing code.
The trunk(v1.10) version will remove the unused options position.
(closes issue #19273)
Reported by: mdavenport
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r320823 | rmudgett | 2011-05-25 12:06:38 -0500 (Wed, 25 May 2011) | 18 lines
The AMI Newstate event contains different information between v1.4 and v1.8.
The addition of connected line support in v1.8 changes the behavior of the
channel caller ID somewhat. The channel caller ID value no longer time
shares with the connected line ID on outgoing call legs. The timing of
some AMI events/responses output the connected line ID as caller ID.
These party ID's are now separate.
* The ConnectedLineNum and ConnectedLineName headers were added to many
AMI events/responses if the CallerIDNum/CallerIDName headers were also
present.
(closes issue #18252)
Reported by: gje
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1227/
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r320650 | rmudgett | 2011-05-23 12:53:44 -0500 (Mon, 23 May 2011) | 16 lines
Add ConnectedLineNum/Name headers to output of AMI action Status.
* Add ConnectedLineNum and ConnectedLineName headers to the output of the
AMI action Status. This makes it easier to find out who the channel is
connected to without having to lookup BridgedChannel or when they are
connected to an application (e.g.: VoiceMail) which has no bridged
channel.
* Bridged channels with no CallerID had "" instead of "<unknown>" output,
that might be a bug as "<unknown>" was what older versions used.
(closes issue #18158)
Reported by: gareth
Patches:
svn-292308.diff uploaded by gareth (license 208)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320651 65c4cc65-6c06-0410-ace0-fbb531ad65f3
state of the channel reverts to unknown this should be rejected.
this is important for negotiating T.38 gateway see #13405
This patch adds a option T38_REJECTED that behaves as T38_DISABLED except it reports state rejected.
Trivial Change to res_fax to honnor UNAVAILABLE and REJECTED states.
(closes issue #18889)
Reported by: irroot
Tested by: irroot, darkbasic, mnicholson
Review: https://reviewboard.asterisk.org/r/1115
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319087 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This code will actually detect any dialplan jump from any application that
calls ast_explicit_goto(). This change is only being done in trunk as it may
change the way some dialplans execute.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317721 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The original implementation of this did the merging of all contexts with the
same name in the realtime layer, but that implementation severely breaks
drivers which use the same context name (e.g. iax.conf, type={peer,user}).
Therefore, the implementation needs to do the merging for particular entries
only, based upon what contexts would allow that in the channel driver itself.
This implementation is for chan_sip only, but others could be added in the
future.
(closes issue #17957)
Reported by: marcelloceschia
Patches:
chan-sip_parsing-general_branch162.patch uploaded by marcelloceschia (license 1079)
Tested by: tilghman
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316428 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If the user invokes 'dialplan add extension' into a non-existing context, the context will be created
and a message informing the user of the context being created will be issued in cli.
(closes issue #17431)
Reported by: leearcher
Patches:
context_auto_create.diff uploaded by kobaz (license 834)
Tested by: leearcher, kobaz, jrose
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312678 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In chan_dahdi.conf, the user can now use length 4 patterns in addition to the usual length 2 patterns. The s
ntax remains the same and the method used to track the pattern history will only change when using the length
4 patterns.
(closes issue SWP-3250)
Code:
jrose
rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Adding the setvar option with variable substitution on the value allows things
like setting the outbound caller id name to the summary of a calendar event,
etc. Values could be chained together as they are appended in order to do some
scripting if necessary.
Review: https://reviewboard.asterisk.org/r/1134/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309640 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
From the submitter:
I've added a new manager action to list only the active conferences on an
Asterisk system. It shows the same data displayed when you run a 'meetme list'
on the Asterisk CLI.
(closes issue #17905)
Reported by: rcasas
Patches:
app_meetme.c.patch uploaded by rcasas (license 641)
Review: https://reviewboard.asterisk.org/r/874/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307359 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Pass a MCID request to the bridged channel so the bridged channel can send
it to the network.
The ability to send the MCID request on an ISDN span is enabled with the
new chan_dahdi.conf mcid_send option.
JIRA SWP-2845
JIRA ABE-2736
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306755 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The display ie handling can be controlled independently in the send and
receive directions with the following options:
* Block display text data.
* Use display text in SETUP/CONNECT messages for name.
* Use display text for COLP name updates (FACILITY/NOTIFY as appropriate).
* Pass arbitrary display text during a call. Sent in INFORMATION
messages. Received from any message that the display text was not used as
a name.
If the display options are not set then the options default to legacy
behavior.
The arbitrary display text is exchanged between bridged channels using the
AST_FRAME_TEXT frame type.
To send display text from the dialplan use the SendText() application when
the arbitrary display text option is enabled.
JIRA SWP-2688
JIRA ABE-2693
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306396 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Discussion on #asterisk on 2011-01-19:
(02:07:03 PM) boch: i wonder how to cycle all entries in a tree
(02:07:11 PM) leifmadsen: use While()
(02:07:17 PM) leifmadsen: you need to know the tree structure already though
(02:07:36 PM) boch: what you mean?
(02:09:02 PM) leifmadsen: you need to know the structure prior to looping, because you can't just return the structure from the dialplan
(02:09:43 PM) leifmadsen: the only way I can think of doing that is via something like writing the output of: asterisk -rx "database show" to a file, then looping through that to know the structure of the database and check everything
(02:09:59 PM) leifmadsen: but at that point you're better off just using either a relational database or an external script
(02:10:13 PM) boch: for example i need to know all entries in the tree
(02:10:15 PM) boch: got it
(02:10:20 PM) leifmadsen: exactly
(02:10:22 PM) leifmadsen: that's the problem
(02:10:22 PM) boch: thank you
(02:13:09 PM) mateu: yeah, i'm surprised there isn't something from the dialplan like 'database show family' so one can get all keys in a family to loop over.
(02:15:35 PM) leifmadsen: database shows everything
(02:16:22 PM) mateu: i mean something from the dial plan that mimics 'database show <family>'
(02:16:41 PM) leifmadsen: guess no one has found that important enough to program :)
(02:16:52 PM) leifmadsen: at that point you should probably just use a relational database...
(02:17:10 PM) mateu: i dunno
(02:17:16 PM) mateu: seems pretty basic to me.
(02:17:16 PM) leifmadsen: me either
(02:17:19 PM) leifmadsen: sure does
(02:17:24 PM) leifmadsen: no one has programmed it though
(02:17:28 PM) ***leifmadsen shrugs
(02:17:43 PM) mateu: ok, well at least we know how it currently stands. thanks leifmadsen
(02:28:52 PM) Corydon76-home: leifmadsen: something like HASHKEYS() ?
(02:30:11 PM) leifmadsen: Corydon76-home: ummm, I was thinking more like DUNDI_QUERY() and DUNDI_RESULT()
(02:30:31 PM) leifmadsen: although HASHKEYS() might work
(02:30:58 PM) leifmadsen: actually ya, looking at it, similar to HASHKEYS()
(02:31:01 PM) leifmadsen: DBKEYS() I guess?
(02:31:45 PM) Corydon76-home: So with no argument, retrieves families, with an argument, retrieves keys of that family?
(02:34:02 PM) leifmadsen: ya
(02:34:16 PM) leifmadsen: how would you iterate through layers of them?
(02:34:30 PM) leifmadsen: i.e. family/key/key/key ?
(02:34:43 PM) Corydon76-home: Essentially, yes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303198 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Added the moh_signaling option to specify what to do when the channel's
bridged peer puts the ISDN channel on and off of hold.
Implemented as a FSM to control libpri ISDN signaling when the bridged
peer places the channel on and off of hold with the AST_CONTROL_HOLD and
AST_CONTROL_UNHOLD control frames.
JIRA SWP-2687
JIRA ABE-2691
Review: https://reviewboard.asterisk.org/r/1063/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300212 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Added option v(mailbox@[context]) which tells MeetMe where to look for a users greet file. If one does not exist it clears the v option and defers to the functionality of i/I as/if set by the MeetMe() command.
Review: https://reviewboard.asterisk.org/r/1009/
(closes issue #18297)
Reported by: parisioa
Patches:
meetme_final_patch_v.diff uploaded by parisioa (license 1153)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296249 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r287647 | dvossel | 2010-09-20 17:09:16 -0500 (Mon, 20 Sep 2010) | 21 lines
Addition of the FrameHook API (AKA AwesomeHooks)
So far all our tools for viewing and manipulating media streams
within Asterisk have been entirely focused on audio. That made
sense then, but is not scalable now. The FrameHook API lets us
tap into and manipulate _ANY_ type of media or signaling passed
on a channel present today or in the future. This tool is a step
in the direction of expanding Asterisk's boundaries and will help
generate some rather interesting applications in the future.
In addition to the FrameHook API, a simple dialplan function
exercising the api has been included as well. This function
is called FRAME_TRACE(). FRAME_TRACE() allows for the internal
ast_frames read and written to a channel to be output. Filters
can be placed on this function to debug only certain types of frames.
This function could be thought of as an internal way of doing
ast_frame packet captures.
Review: https://reviewboard.asterisk.org/r/925/
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r286931 | jpeeler | 2010-09-15 14:22:15 -0500 (Wed, 15 Sep 2010) | 16 lines
Add parking extension for non-default parking lots.
This is a new feature that allows for parking to custom parking lots to be
accessed directly, rather than with channel variables or by changing the
default parking lot. The extension is set with the parkext option just as the
default parking lot is done. Also, the manager action has been updated to
optionally allow a specified parking lot.
(closes issue #14882)
Reported by: vmikhnevych
Patches:
patch_14882.txt uploaded by mnick (license 874)
modified by me
Review: https://reviewboard.asterisk.org/r/884/
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r284950 | dvossel | 2010-09-03 12:29:02 -0500 (Fri, 03 Sep 2010) | 14 lines
authenticate OPTIONS requests just like we would an INVITE
OPTIONS requests should be treated the same as an INVITE
This includes authentication. This patch adds the ability for
incoming out of dialog OPTION requests to be authenticated
before providing a response indicating whether an extension
is available or not. The authentication routine works the
exact same way as it does for incoming INVITEs. This means
that if a peer has 'insecure=invite' in their peer definition,
the same will be true for the processing of the OPTIONS request.
Review: https://reviewboard.asterisk.org/r/881/
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r282047 | dvossel | 2010-08-12 15:15:41 -0500 (Thu, 12 Aug 2010) | 35 lines
improved translation paths for wideband codecs
The problem I'm addressing is that Asterisk's current
method of building the least cost translation paths
between codecs does not take into account sample rate.
For instance, it was possible for siren14 (a 32khz codec),
to contain the a translation path to siren7 (a 16khz
audio codec) that goes through slin at 8khz. In this
case Asterisk takes a 32khz codec, down samples it to
8khz and then up samples it to 16khz which is terrible
regardless if it is computationally less expensive. This
patch now builds translation paths that give priority to
maintaining the best possible sample rate before taking
into consideration computational cost. This patch also
adds cli commands to expose what translation paths are
actually being used.
Changes:
1. Translation paths will never contain a step that changes
the sample rate unless absolutely necessary.
2. When choosing the best codec to make two channels compatible.
Shared codecs with the highest sample rate are given priority.
3. A new cli command to show all translation paths available
for a specific codec 'core show translation paths [codec name]'
has been added.
4. 'core show translation' which displays the translation
matrix now includes the new higher bit audio codecs in the table.
5. 'core show channel [channel name]' now displays the
translation paths if translation is used.
(closes issue #16841)
Reported by: dvossel
Review: https://reviewboard.asterisk.org/r/842/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282048 65c4cc65-6c06-0410-ace0-fbb531ad65f3
sip.conf configuration for the channel and for devices.
The Max-Forwards header is used to prevent loops in a SIP network. Each intermediary,
like SIP proxys and SBCs, decrement this counter and detects when it reaches zero,
at which point the SIP request is nicely killed in a SIP-friendly way.
Review: https://reviewboard.asterisk.org/r/778/
Thanks to dvossel for the review and good advice.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds a generic API for accommodating IPv6 and IPv4 addresses
within Asterisk. While many files have been updated to make use of the
API, chan_sip and the RTP code are the files which actually support
IPv6 addresses at the time of this commit. The way has been paved for
easier upgrading for other files in the near future, though.
Big thanks go to Simon Perrault, Marc Blanchet, and Jean-Philippe Dionne
for their hard work on this.
(closes issue #17565)
Reported by: russell
Patches:
asteriskv6-test-report.pdf uploaded by russell (license 2)
Review: https://reviewboard.asterisk.org/r/743
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274783 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch as documented in the sample config allows one to optionally apply
white, black, or both types of filtering to manager events. The new
'eventfilter' option is set per user.
(closes issue #14861)
Reported by: fnordian
Patches:
eventfilter3.patch uploaded by fnordian (license 110),
modified by me
Review: https://reviewboard.asterisk.org/r/673/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271868 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* chan_dahdi supports dialing configuring and dialing by device file name.
DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
* A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
False by default. If set, chan_dahdi will ignore failed 'channel' entries.
Handy for the above name-based syntax as it does not depend on
initialization order.
* have my_pri_make_cc_dialstring() only manupulate dial-strings of group
(gGrR) dialing, which make it lsightly more complicated.
https://reviewboard.asterisk.org/r/535/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269238 65c4cc65-6c06-0410-ace0-fbb531ad65f3
People expressed an interest in having access to the exact length of calls to a finer degree than seconds. See the CHANGES and UPGRADE.txt for usage also updated the sample configs to note the change.
Patch by snuffy.
(closes issue #16559)
Reported by: cianmaher
Tested by: cianmaher, snuffy
Review: https://reviewboard.asterisk.org/r/461/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269153 65c4cc65-6c06-0410-ace0-fbb531ad65f3
After 5 years in mantis and over a year on reviewboard, SRTP support is finally
being comitted. This includes generic CHANNEL dialplan functions that work for
getting the status of whether a call has secure media or signaling as defined
by the underlying channel technology and for setting whether or not a new
channel being bridged to a calling channel should have secure signaling or
media. See doc/tex/secure-calls.tex for examples.
Original patch by mikma, updated for trunk and revised by me.
(closes issue #5413)
Reported by: mikma
Tested by: twilson, notthematrix, hemanshurpatel
Review: https://reviewboard.asterisk.org/r/191/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Updated the UPGRADE.txt and CHANGES file stating that CDR records will not be explicity
written unless cdr.conf exists and is configured.
(closes issue #17373)
Reported by: wdoekes
Tested by: pabelanger
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267624 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Add the ability to announce a call to an endpoint when there are no B
channels available. A call waiting call is a SETUP message with no B
channel selected.
Relevant specification: EN 300 056, EN 300 057, EN 300 058
For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
"no_media_path" option.
* Returns "0" if there is a B channel associated with the call.
* Returns "1" if no B channel is associated with the call. The call is
either on hold or is a call waiting call.
If you are going to allow incoming call waiting calls then you need to use
CHANNEL(no_media_path) do determine if you must drop a call to accept the
new call.
Review: https://reviewboard.asterisk.org/r/568/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267261 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Added ability to send and receive ETSI Explicit Call Transfer (ECT)
messages to eliminate tromboned calls.
Note: Asterisk already supported initiating the transfer of calls to
eliminate tromboned calls to libpri so there was nothing to do for the
asterisk portion.
Review: https://reviewboard.asterisk.org/r/520/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266926 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Queries from the PBX core come in 3's. Caching avoids the additional
performance penalty from those two additional queries hitting the database.
(closes issue #16521)
Reported by: tilghman
Patches:
20091229__issue16521.diff.txt uploaded by tilghman (license 14)
Tested by: Hubguru, tilghman
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266238 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@264626 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Added the keyword 'all' to the 'channel hangup request' CLI command
so that you can request all channels to be hungup without having to
restart Asterisk.
(closes issue #16009)
Reported by: moy
Patches:
hangup-all-rev-221688.patch uploaded by moy (license 222)
Tested by: moy, russell
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264117 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r263769 | jpeeler | 2010-05-18 13:54:58 -0500 (Tue, 18 May 2010) | 10 lines
Modify directory name reading to be interrupted with operator or pound escape.
In the case of accidentally entering the wrong first three letters for the
reading, users could be very frustrated if the name listing is very long. This
allows interrupting the reading by pressing 0 or #. 0 will attempt to execute
a configured operator (o) extension and # will exit and proceed in the
dialplan.
ABE-2200
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263807 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds the following in-conference admin DTMF features:
*81 - Roll call (or simply user count if INTROUSER isn't enabled)
*82 - Eject all non-admins
*83 - Mute/unmute all non-admins
*84 - Start recording the conference on the fly
FWIW, this code uses newly recorded prompts.
(closes issue #16379)
Reported by: rfinnie
Patches:
meetme-enhancements-232771-v1.patch uploaded by rfinnie (license 940)
modified slightly by me
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260757 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Gregory Nietsky
Kinsey Moore
David Vossel
Russell Bryant
Richard Mudgett
Jonathan Rose
Matthew Nicholson
Tilghman Lesher
Leif Madsen
Terry Wilson
Jeff Peeler
Paul Belanger
Andrew Parisio
Mark Michelson
David Ruggles
Olle Johansson
Tzafrir Cohen
Bradley Latus
Kevin P. Fleming