The patch for CDRs moved around a lot of content in CHANGES to try and
organize the areas that were affected. This missed some changes that went
in with a merge and removed some updates - this patch adds them back in.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392004 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch is the initial push to update Asterisk's CDR engine for the new
bridging framework. This patch guts the existing CDR engine and builds the new
on top of messages coming across Stasis. As changes in channel state and bridge
state are detected, CDRs are built and dispatched accordingly. This
fundamentally changes CDRs in a few ways.
(1) CDRs are now *very* reflective of the actual state of channels and bridges.
This means CDRs track well with what an actual channel is doing - which
is useful in transfer scenarios (which were previously difficult to pin
down). It does, however, mean that CDRs cannot be 'fooled'. Previous
behavior in Asterisk allowed for CDR applications, channels, and other
properties to be spoofed in parts of the code - this no longer works.
(2) CDRs have defined behavior in multi-party scenarios. This behavior will not
be what everyone wants, but it is a defined behavior and as such, it is
predictable.
(3) The CDR manipulation functions and applications have been overhauled. Major
changes have been made to ResetCDR and ForkCDR in particular. Many of the
options for these two applications no longer made any sense with the new
framework and the (slightly) more immutable nature of CDRs.
There are a plethora of other changes. For a full description of CDR behavior,
see the CDR specification on the Asterisk wiki.
(closes issue ASTERISK-21196)
Review: https://reviewboard.asterisk.org/r/2486/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391947 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer
channel driver specific. If the channel variable is set on the
transferrer channel, the sound will be played to the target of an attended
transfer.
* The channel variable BRIDGEPEER becomes a comma separated list of peers
in a multi-party bridge. The BRIDGEPEER value can have a maximum of 10
peers listed. Any more peers in the bridge will not be included in the
list. BRIDGEPEER is not valid in holding bridges like parking since those
channels do not talk to each other even though they are in a bridge.
* The channel variable BRIDGEPVTCALLID is only valid for two party bridges
and will contain a value if the BRIDGEPEER's channel driver supports it.
* The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and
is removed. The more useful DYNAMIC_WHO_ACTIVATED gives the channel name
that activated the dynamic feature.
* The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are
set only on the channel executing the dynamic feature. Executing a
dynamic feature on the bridge peer in a multi-party bridge will execute it
on all peers of the activating channel.
(closes issue ASTERISK-21555)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2582/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390771 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The attended transfer API call can complete the attended transfer in a number of ways
depending on the current bridged states of the channels involved.
The hiding of masquerades is done in some bridging-related functions, such as the manager
Bridge action and the Bridge dialplan application. In addition, call pickup was edited
to "move" a channel rather than masquerade it.
Review: https://reviewboard.asterisk.org/r/2511
(closes issue ASTERISK-21334)
Reported by Matt Jordan
(closes issue Asterisk-21336)
Reported by Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389848 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch moves a number of AMI events over to the Stasis-Core message bus.
This includes:
* ChanSpyStart/Stop
* MonitorStart/Stop
* MusicOnHoldStart/Stop
* FullyBooted/Reload
* All Voicemail/MWI related events
In addition, it adds some Stasis-Core and AMI support for generic AMI messages,
refactors the message router in AMI to use a single router with topic
forwarding for the topics that AMI cares about, and refactors MWI message
types and topics to be more name compliant.
Review: https://reviewboard.asterisk.org/r/2532
(closes issue ASTERISK-21462)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389733 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Breaks many things until they can be reworked. A partial list:
chan_agent
chan_dahdi, chan_misdn, chan_iax2 native bridging
app_queue
COLP updates
DTMF attended transfers
Protocol attended transfers
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change adds a framework in res_stasis for handling events from
channel topics. JSON event generation and validation code is created
from event documentation in rest-api/api-docs/events.json to assist in
JSON event generation, ensure consistency, and ensure that accurate
documentation is available for ALL events that are received by
res_stasis applications.
The userevent application has been refactored along with the code that
handles userevent channel blob events to pass the headers as key/value
pairs in the JSON blob. As a side-effect, app_userevent now handles
duplicate keys by overwriting the previous value.
Review: https://reviewboard.asterisk.org/r/2428/
(closes issue ASTERISK-21180)
Patch-By: Kinsey Moore <kmoore@digium.com>
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388275 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change does the following:
1. Adds the sorcery realtime module
2. Adds unit tests for the sorcery realtime module
3. Changes the realtime core to use an ast_variable list instead of variadic arguments
4. Changes all realtime drivers to accept an ast_variable list
Review: https://reviewboard.asterisk.org/r/2424/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The settings saved on the channel for FEATURE()/FEATUREMAP() were only
for that channel. This patch adds the ability to have these settings
inherited to child channels if you set FEATURE(inherit)=yes.
Closes issue ASTERISK-21306.
Review: https://reviewboard.asterisk.org/r/2415/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385088 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does the following:
* A new Stasis payload has been defined for multi-channel messages. This
payload can store multiple ast_channel_snapshot objects along with a single
JSON blob. The payload object itself is opaque; the snapshots are stored
in a container keyed by roles. APIs have been provided to query for and
retrieve the snapshots from the payload object.
* The Dial AMI events have been refactored onto Stasis. This includes dial
messages in app_dial, as well as the core dialing framework. The AMI events
have been modified to send out a DialBegin/DialEnd events, as opposed to
the subevent type that was previously used.
* Stasis messages, types, and other objects related to channels have been
placed in their own file, stasis_channels. Unit tests for some of these
objects/messages have also been written.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384910 65c4cc65-6c06-0410-ace0-fbb531ad65f3
HangupRequest and SoftHangupRequest are now ast_channel_blob Stasis
messages, with the cause code as an optional field in the blob.
NewCallerid now simply watches for changes in the callerid information
in channel snapshots, and creates the AMI event appropriately.
Since the original NewCallerid event honored the channelvars setting
in manager.conf, the channel variables configured there had to become
a part of the channel snapshot. These are now a part of every snapshot
based event, making the configuration description "every time a
channel-oriented event is emitted" less of a lie.
There a a few other changes wrapped up in here as well.
* When ast_channel_topic() is given NULL for a channel, it returns
the ast_channel_topic_all() topic instead of NULL. This can clean
up a lot of NULL checking we're doing currently.
* The fields Cause and Cause-txt were removed from the base channel
information and put only on the Hangup events, since those fields
are meaningless outside of a Hangup event.
* Removed the pipe-delimiter processing of the channelvars field,
since that's been deprecated forever.
(closes issue ASTERISK-21096)
Review: https://reviewboard.asterisk.org/r/2405/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383726 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch started out simply as fixing the bouncing tests introduced
in r382685, but required some other changes to give it a decent
implementation.
To fix the bouncing tests, the UserEvent and Newexten AMI events
needed to be refactored to dispatch via Stasis. Dispatching directly
to AMI resulted in those events sometimes getting ahead of the
associated Newchannel events, which would understandably confuse anyone.
I found that instead of creating a zillion different message types and
structures associated with them, it would be preferable to define a
message type that has a channel snapshot and a blob of structured data
with a small bit of additional information. The JSON object model
provides a very nice way of representing structured data, so I went
with that.
* Move JSON support from res_json.c to main/json.c
* Made libjansson-dev a required dependency
* Added an ast_channel_blob message type, which has a channel
snapshot and JSON blob of data.
* Changed UserEvent and Newexten events so that they are dispatched
via ast_channel_blob messages on the channel's topic.
* Got rid of the ast_channel_varset message; used ast_channel_blob
instead.
* Extracted the manager functions converting Stasis channel events to
AMI events into manager_channel.c.
(issue ASTERISK-21096)
Review: https://reviewboard.asterisk.org/r/2381/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change allows you to use XMPP buddy state in places where device state
can be used be used, such as dialplan hints. If at least one resource is
available the buddy is considered available. Now your phone can reflect
their IM status too!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383283 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Added an option "discard_remote_hold_retrieval" (default "no") that if set does
not trigger the music on hold event. This essentially stops telling the peer
to start music on hold.
(issue ABE-2899)
Reported by: Denis Alberto Martinez
Review: https://reviewboard.asterisk.org/r/2336/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382787 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds support for RFC 3327 "Path" headers. This can be enabled in
sip.conf using the 'supportpath' setting, either on a global basis or on a
peer basis. This setting enables Asterisk to route outgoing out-of-dialog
requests via a set of proxies by using a pre-loaded route-set defined by the
Path headers in the REGISTER request. This patch also adds Realtime support
for dynamically updating the Path information for a peer.
A huge thank-you to Klaus Darillion and Olle E Johansson for their efforts
in writing this patch.
Review: https://reviewboard.asterisk.org/r/2235/
Review: https://reviewboard.asterisk.org/r/991/
(closes issue ASTERISK-16884)
Reported by: klaus3000
Tested by: klaus3000, oej, mjordan
patches:
path-1.8.0-patch.txt uploaded by klaus3000 (License 5054)
oolong-path-support-trunk in team branch by oej (License 5267)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382440 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The original report had to do with a realtime peer behind NAT being pruned and
the peer's private address being used instead of its external address. Upon
debugging, it was discovered that this was being caused by the addition of
the auto_force_rport and auto_comedia settings.
This patch does the following:
* Adds a missing note to the CHANGES file indicating that the default global nat
setting is auto_force_rport
* Constify the 'req' parameter for check_via()
* Add calls to check_via() in a couple of places in order for the auto_*
settings to do their job in attempting to determine if NAT is involved
* Set the flags SIP_NAT_FORCE_RPORT and SIP_PAGE2_SYMMETRICRTP if the auto_*
settings are in use where it was needed
* Moves the copying of peer flags up in build_peer() to before they are used;
this fixes the realtime prune issue
* Update the contrib/realtime schemas to allow the nat column to handle the
different nat setting combinations we have
This patch received a review and "Ship It!" on the issue itself.
(closes issue ASTERISK-20904)
Reported by: JoshE
Tested by: JoshE, Michael L. Young
Patches:
asterisk-20904-nat-auto-and-rt-peersv2.diff Michael L. Young (license 5026)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382323 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For some channel drivers, specifically those that have a varying rate in the
number of audio samples, the audio quality for a MeetMe conference can be
exceedingly poor. This is due to a unilateral application of the DENOISE
function in func_speex to channels joining the conference.
The denoiser function in the speex library is initialized with the number of
audio samples in each sample that will be provided to it. If the number of
audio samples changes, the denoiser has to be thrown away and re-initialized.
While this could be worked around by removing func_speex, that doesn't help
if you actually use the denoiser with other channels on the system.
This patches does the following:
* Checks for the presence of func_speex as opposed to codec_speex when
determining if the DENOISE function is present (which is where the function
is actually implemented)
* Adds an option to MeetMe 'n' that causes the denoiser to not be applied
to a channel when it joins. This keeps the current behavior the default, but
let's users disable the denoiser if it causes problems on their system.
Review: https://reviewboard.asterisk.org/r/2358
(closes issue AST-1062)
Reported by: Thomas Arimont
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382232 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Currently, if one starts, stops, and then starts a recording again for a
conference the recorded data is appended to the file originally created
on the first record start. An option record_file_append has been added
that defaults to "yes", but when set to "no" will force creation of a new
file between every record start/stop.
(issue AST-1088)
Reported by: John Bigelow
Review: http://reviewboard.digium.internal/r/374/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381729 65c4cc65-6c06-0410-ace0-fbb531ad65f3
These two variables were previously not being set when comebacktoorigin=yes
and the example configs seemed to imply that they should be. Since there
is no harm in this and since calls that are sent back to origin are capable
of continuing in the dialplan, this seemed like a no-brainer. Also it
supports some bridging tests I've been working on.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381068 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds the capability for asynchronous manipulation of audio being
played back to a channel though a new AMI action "ControlPlayback". The
ControlPlayback action supports a number of operations, the availability of
which depend on the application being used to send audio to the channel.
When the audio playback was initiated using the ControlPlayback application
or CONTROL STREAM FILE AGI command, the audio can be paused, stopped,
restarted, reversed, or skipped forward. When initiated by other mechanisms
(such as the Playback application), the audio can be stopped, reversed, or
skipped forward.
Review: https://reviewboard.asterisk.org/r/2265/
(closes issue ASTERISK-20882)
Reported by: mjordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379830 65c4cc65-6c06-0410-ace0-fbb531ad65f3
From the patch author:
"First this patch adds general support for busy detection. It also adds support
for the ECAM command at Sony Ericsson phones and also signals busy when only
early media was received but the call got not answered."
Review: https://reviewboard.asterisk.org/r/323
(closes issue ASTERISK-14527)
Reported by: Artem Makhutov
Tested by: Artem Makhutov
patches:
busy-full5.patch uploaded by artem (license 5757)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379144 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When ringinuse=no queue members can receive more than one call if these
calls happen at nearly the same time.
* Fix so a queue member does not receive more than one call from a queue.
NOTE: This fix does not prevent multiple calls to a member if the member
is in more than one queue.
* Did some refactoring to eliminate some code redundancy.
(issue ASTERISK-16115)
Reported by: nik600
Patches:
jira_asterisk_16115_single_q_v1.8.patch (license #5621) patch uploaded by rmudgett
Modified
* Revert the -r341580 and -r341599 changes adding the queues.conf
check_state_unknown option as it was added in an attempt to fix this
problem. The fix did not need to be optional. The fix should not have
tried to explicitly set the device state. Setting the device state by
something other than the device introduces a race condition. I also could
not see how the change would be effective other than delaying the
app_queue code long enough for the device state to propagate to app_queue.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378688 65c4cc65-6c06-0410-ace0-fbb531ad65f3
BRIDGE_FEATURES did not previously support the automixmonitor feature. Now it
does. In addition, the BRIDGE_FEATURES variable would not apply features to
the proper party based on whether the feature option letter was in caps or
in lowercase (both ways would apply it to the caller). Now uppercase applies
to the caller while lowercase applies to the callee (like with the dial option)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378063 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The new field is will show up within the response if the requested peer has a
subscribe context set.
(closes issue ASTERISK-20626)
Reported by: Jaco Kroon
Patches:
asterisk-sip-ami-SubscrContext.patch uploaded by jkroon (license 5671)
-with modifications by jrose to conform to style guidelines
Review: https://reviewboard.asterisk.org/r/2195/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376219 65c4cc65-6c06-0410-ace0-fbb531ad65f3
manager show commands now shows the full name of the command being displayed
regardless of size. The privilege column has also been removed from this
display. It will also now use the full length of the terminal if curses is
available. Manager show command will now always display the privilege of
the manager command within the CLI.
(closes ASTERISK-20396)
Reported by: Johan Wilfer
Review: https://reviewboard.asterisk.org/r/2143/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375103 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk's DTMF Specifications are based on AT&T specs, which may not be compatible in other countries.
Various countries have different specifications for the maximum power level differences
between the DTMF low group and high group of frequencies.
Power level difference between frequencies for different Administrations/RPOAs
NTT = Max. 5 dB
AT&T = 4dB(reverse) to 8dB(normal)
Danish = Max. 6 dB
Australian = Max. 10 dB
Brazilian = Max. 9 dB
ETSI = Max. 6 dB from ETSI ES 201 235-3 V1.3.1 (2006-03)
Now allow 4 variables to be individually configured in dsp.conf, with reasonable min/max of 2dB to 20dB.
Default is AT&T specifications
Add's the following variables to dsp.conf
;dtmf_normal_twist=6.31
;dtmf_reverse_twist=2.51
;relax_dtmf_normal_twist=6.31
;relax_dtmf_reverse_twist=3.98
(closes issue ASTERISK-20442)
Reported by: tbsky
Tested by: tbsky,alecdavis
alecdavis (license 585)
Review https://reviewboard.asterisk.org/r/2141/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374387 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds an optional header to the PlayDTMF AMI action, Duration.
It allows the duration of the DTMF digit to be played on the channel to be
specified in milliseconds.
(closes issue ASTERISK-18172)
Reported by: Renato dos Santos
patches:
send-dtmf.patch uploaded by Renato dos Santos (license #6267)
Modified slightly for this commit for Asterisk 12.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373979 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Currently, if there are modifications to mailboxes that Asterisk is
not aware of, the user needs to add "pollmailboxes" to their mailbox
configuration, which repeatedly polls the subscribed mailboxes for
changes. This results in a lot of extra work for the CPU. This patch
introduces the AMI command VoicemailRefresh which permits external
applications to trigger the refresh themselves. The refresh can apply
to a specified mailbox only, an entire context, or all configured
mailboxes. Even a refresh performed on every mailbox would not consume
as much CPU as the pollmailboxes option, given that pollmailboxes runs
continuously and this only runs on demand.
(closes issue ASTERISK-17206)
(closes issue ASTERISK-19908)
Reported-by: Jeff Hutchins
Reported-by: Tilghman Lesher
Patch-by: Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373913 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This allows for the REDIRECTING dialplan function to be
used to set the reason to any string.
The SIP channel driver has been modified to set the redirecting
reason string to the value received in a Diversion header. In
addition, SIP 480 response reason text will set the redirecting
reason as well.
(closes issue AST-942)
reported by Malcolm Davenport
(closes issue AST-943)
reported by Malcolm Davenport
Review: https://reviewboard.asterisk.org/r/2101
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373701 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Sets INUSE when no free agents, NOT_INUSE when an agent is free.
modifes handle_statechange() scan members loop to scan for a free agent
and updates the Queue:queuename_avial devstate.
Previously exited early if the member was found in the queue.
Now Exits later when both a member was found, and a free agent was found.
alecdavis (license 585)
Reported by: Alec Davis
Tested by: alecdavis
Review: https://reviewboard.asterisk.org/r/2121/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373188 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Prior to this patch, if pause or unpause was issued on an interface
without specifying a specific queue, a PAUSEALL or UNPAUSEALL event
would be logged in the queue log even if that interface wasn't a
member of any queues. This patch changes it so that these events are
only logged when at least one member of any queue exists for that
interface.
(closes issue AST-946)
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/2079/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372148 65c4cc65-6c06-0410-ace0-fbb531ad65f3
(closes issue ASTERISK-18390)
Reported by: Peter Racz
Patches:
dundi_cli_cache.patch.v2 uploaded by Peter Racz (license #6290)
ASTERISK-18390_dundi_cli_cache_jrose_mods_v2.diff uploaded by Jonathan Rose (license #6182)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370978 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is based on the work done by Olle Johansson on review board.
The idea is that the channel specified in an AMI originate or call
file is typically not connected to the outgoing extension until the
channel has been answered. With this change, an EarlyMedia header can
be specified for AMI originates and an early_media option can
be specified in call files. With this option set, once early media is
received on a channel, it will be connected with the outgoing extension.
(closes issue ASTERISK-18644)
Reported by Olle Johansson
Review: https://reviewboard.asterisk.org/r/1472
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Implementation of a dialplan function for checking manager accounts. Right now
it only returns the number of logged in sessions for a manager account, but
other attributes can be added later.
Patch by: Olle Johansson
Review: https://reviewboard.asterisk.org/r/421/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370943 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds named calledgroups/pickupgroups to Asterisk. Named groups are
implemented in parallel to the existing numbered callgroup/pickupgroup
implementation. However, unlike the existing implementation, which is limited
to a maximum of 64 defined groups, the number of defined groups allowed for
named callgroups/pickupgroups is effectively unlimited.
Named groups are configured with the keywords "namedcallgroup" and
"namedpickupgroup". This corresponds to the numbered group definitions of
"callgroup" and "pickupgroup". Note that as the implementation of named groups
coexists with the existing numbered implementation, a defined named group of
"4" does not equate to numbered group 4.
Support for the named groups has been added to the SIP, DAHDI, and mISDN channel
drivers.
Review: https://reviewboard.asterisk.org/r/2043
Uploaded by:
Guenther Kelleter(license #6372)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370831 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is a patch from kkm from review board.
This is useful for adding headers to REFER requests that
emanate from a Transfer() dialplan application call.
This also fixes some uses of the Referred-by header, removing
an extra set of angle brackets.
I've modified the reporter's original patch to not require
any additions to the sip_refer header and to just remove the
referred_by_name from sip_refer since it is no longer needed
or used.
(closes Issue ASTERISK-17639)
reported by Kirill Katsnelson
Patches:
019059-sip-refer-addheaders-trunk-353549.diff
uploaded by Kirill Katsnelson (license #5845)
Review: https://reviewboard.asterisk.org/r/1159
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370691 65c4cc65-6c06-0410-ace0-fbb531ad65f3
With this option set, channel variables can be set on
every manager originate. The Variable header can still
be used to set additional channel variables for individual
calls if desired.
This work was completed by Olle Johansson on review board.
I have applied the review feedback and am committing it in
order to get this into trunk before Asterisk 11 is branched.
Review: https://reviewboard.asterisk.org/r/1412
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370681 65c4cc65-6c06-0410-ace0-fbb531ad65f3
From corruptor's review board posting:
"I've noticed that we can remove particular extension from context with
dialplan remove extension command but in order to remove all extensions
in the context we should delete them on by one. I've created dialplan
remove context command which uses ast_context_destroy to destroy the
whole context with all extensions. I've created to functions for in
pbx_config.c: handle_cli_dialplan_remove_context which actually removes
context and complete_dialplan_remove_context which completes input.
They are based on other similar functions and pretty trivial but I can be
mistaken somewhere.
"I've also modified dialplan add include <context2> into <context1>. I've
made it similar dialplan add extension ... command. It creates <context1>
if it doesn't exist and I've also modified complete_dialplan_add_include
and removed check for existance of <context2> because we can include
non-existent context into another one. (I usually include empty
(non-existent) contexts in advance). Should we raise warning in this case
as it's raised while reading extensions.conf?
"I use those functions with AMI. I think manager commands should be created
in addition to those CLI commands."
I've addressed the latest comments on review board and have made some other
coding guidelines-related cleanup. I also have modified the CHANGES file to
mention these new commands.
(closes issue ASTERISK-19292)
reported by Andrey Solovyev
Patches:
dialplan_add_include.patch
uploaded by Andrey Solovyev (license #5214)
dialplan_remove_context.patch
uploaded by Andrey Solovyev (license #5214)
Review: https://reviewboard.asterisk.org/r/2042
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370644 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch allows you to specify a port number for the MySQL server.
It's useful if a MySQL server is running on a non-standard port.
Even though this module is deprecated in favor of func_odbc, someone
asked for this feature and it seems pretty harmless to add.
It has been tested using a number of combinations of with/without a
port number specified in the dialplan and changing the port number
for mysqld.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370534 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch was submitted by mnicholson a while back. It adds a new AMI action
which allows users to request SIP peer status on demand similar to existing
PeerStatus events and to the output you would see from CLI with sip show peer
Review: https://reviewboard.asterisk.org/r/1098/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370518 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch changes the Asterisk configure script and build system to detect
the presence of the NetBSD editline library (libedit) on the system. If it is
found, it will be used in preference to the version included in the Asterisk
source tree.
(closes issue ASTERISK-18725)
Reported by: Jeffrey C. Ollie
Review: https://reviewboard.asterisk.org/r/1528/
Patches:
0001-Allow-linking-building-against-an-external-editline.patch uploaded by jcollie (license #5373) (heavily modified by kpfleming)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370481 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This updates the CHANGES file with things that were committed for
Asterisk 11, but were not noted in that file.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370353 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds Named ACL functionality to Asterisk. This allows system
administrators to define an ACL and refer to it by a unique name. Configurable
items can then refer to that name when specifying access control lists.
It also includes updates to all core supported consumers of ACLs. That includes
manager, chan_sip, and chan_iax2. This feature is based on the deluxepine-trunk
by Olle E. Johansson and provides a subset of the Named ACL functionality
implemented in that branch. For more information on this feature, see acl.conf
and/or the Asterisk wiki.
Review: https://reviewboard.asterisk.org/r/1978/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369959 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This channel driver is a replacement for both chan_gtalk and chan_jingle but adds additional features not found in either.
These features include full configuration reload, video, full codec support, bidirectional cause code mapping, hold,
unhold, and ringing indication. It is also compliant with the current published Jingle and Google Jingle specifications.
The original Google Talk protocol is also supported for Google Voice interoperability.
You may ask yourself though where the name motif comes from... and I would say to you... music!
motif: a perceivable or salient recurring fragment or succession of notes
Sorta like a jingle!
Review: https://reviewboard.asterisk.org/r/1917/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369769 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds a new CLI command, 'stun show status'. This command will show
a table describing all known STUN servers and statuses.
(closes issue ASTERISK-18046)
Reported by: Jeremy Kister
Tested by: Jeremy Kister
patches:
(stun-show-status-v4-trunk.patch license #6232 uploaded by Jeremy Kister)
Review: https://reviewboard.asterisk.org/r/2001
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369681 65c4cc65-6c06-0410-ace0-fbb531ad65f3
options are documented in config sample
sample config rename to proper name - ooh323.conf
To change media address ooh323 send empty TCS if there was
completed TCS exchange or send facility forwardedelements
with new fast start proposal if not.
Then close transmit logical channels and renew TCS exchange.
If new fast start proposal is received then ooh323 stack call back
channel driver routine to change rtp address in the rtp instance.
If empty TCS is received then close transmit logical channels and
renew TCS exchange
Review: https://reviewboard.asterisk.org/r/1607/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Hangup handlers are an alternative to the h extension. They can be used
in addition to the h extension. The idea is to attach a Gosub routine to
a channel that will execute when the call hangs up. Whereas which h
extension gets executed depends on the location of dialplan execution when
the call hangs up, hangup handlers are attached to the call channel. You
can attach multiple handlers that will execute in the order of most
recently added first.
(closes issue ASTERISK-19549)
Reported by: Mark Murawski
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/2002/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369493 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds the core changes necessary to support AMI event documentation
in the source files of Asterisk, and adds documentation to those AMI events
defined in the core application modules. Event documentation is built from
the source by two new python scripts, located in build_tools:
get_documentation.py and post_process_documentation.py.
The get_documentation.py script mirrors the actions of the existing AWK
get_documentation scripts, except that it will scan the entirety of a source
file for Asterisk documentation. Upon encountering it, if the documentation
happens to be an AMI event, it will attempt to extract information about the
event directly from the manager event macro calls that raise the event. The
post_process_documentation.py script combines manager event instances that
are the same event but documented in multiple source files. It generates
the final core-[lang].xml file.
As this process can take longer to complete than a typical 'make all', it
is only performed if a new make target, 'full', is chosen.
Review: https://reviewboard.asterisk.org/r/1967/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369346 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Adds a number of methods for controlling the setting of 'ringinuse'
which is basically the same concept as the old ignorebusy setting,
only now the per member setting always controls whether or not the
member is actually ringed while in use. A CLI command and a manager
action have been added to change a given queue member's ringinuse
option while Asterisk is running and the an argument has been added
for adding members with deliberately set ringinuse in queues.conf
Some effort has been made to ensure compatability with dialplans and
databases still referring to 'ignorebusy'.
(issue ASTERISK-19536)
reported by: Philippe Lindheimer
Review: https://reviewboard.asterisk.org/r/1919/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367080 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch improves the handling of call id logging significantly with regard
to transfers and adding APIs to better handle specific aspects of logging.
Also, changes have been made to chan_sip in order to better handle the creation
of callids and to enable the monitor thread to bind itself to a particular
call id when a dialog is determined to be related to a callid. It then unbinds
itself before returning to normal monitoring.
review: https://reviewboard.asterisk.org/r/1886/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366842 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If enabled using the keepalive option in sip.conf a small packet will be sent
at a regular interval to keep the NAT mapping open. This is lightweight as the
remote side does not need to parse and handle a SIP message.
(closes issue AST-783)
Review: https://reviewboard.asterisk.org/r/1756/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364500 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Add two new dialplan functions: FEATURE() and FEATUREMAP(). FEATURE()
lets you set some of the configuration options from the [general] section
of features.conf on a per-channel basis. FEATUREMAP() lets you customize
the key sequence used to activate built-in features, such as blindxfer,
and automon. See the built-in documentation for details.
Review: https://reviewboard.asterisk.org/r/1871/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364437 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk has a setting for the minimum allowed DTMF. If we get shorter
DTMF tones, these will be changed to the minimum on the outbound call
leg.
(closes issue ASTERISK-19772)
Review: https://reviewboard.asterisk.org/r/1882/
Reported by: oej
Tested by: oej
Patches by: oej
Thanks to the reviewers.
1.8 branch for this patch: agave-dtmf-duration-asterisk-conf-1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363558 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ISDN ETSI PTP and Q.SIG (And SS7 in future) have support for reporting who
was the original redirecting party of a call.
* Added support for the original redirecting party and reason to the
REDIRECTING function and the system core as well as to the stubbed
locations in sig_pri.c.
Review: https://reviewboard.asterisk.org/r/1829/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362779 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The current Security Events Framework API only supports IPv4 when it comes to
generating security events. This patch does the following:
* Changes the Security Events Framework API to support IPV6 and updates
the components that use this API.
* Eliminates an error message that was being generated since the current
implementation was treating an IPv6 socket address as if it was IPv4.
* Some copyright dates were updated on files touched by this patch.
(closes issue ASTERISK-19447)
Reported by: Michael L. Young
Tested by: Michael L. Young
Patches:
security_events_ipv6v3.diff uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/1777/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Hangup now can take a regular expression as the Channel option. If you want
to hangup multiple channels, use /regex/ as the Channel option. Existing
behavior to hanging up a single channel is unchanged, but if you pass a regex,
the manager will send you a list of channels back that were hung up.
(closes issue ASTERISK-19575)
Reported by: Mark Murawski
Tested by: Mark Murawski
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361038 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Added 'b' and 'B' options to Dial. These options will allow you to run
last-minute dialplan on the caller and callee channels while the Dial
application is executing, but before the call is started. For example you
can use the 'b' option to run dialplan on the callee channel to get the name
of the newly created channel right away.
Review: https://reviewboard.asterisk.org/r/1229/
(closes issue: ASTERISK-19548)
Reported by: Mark Murawski
Tested by: Mark Murawski, Stefan Schmidt
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Added ability to use multiple lines on phone, so for one device in configuration multiple lines can be defined, it allows to have multiple calls on one phone, callwaiting and switching between calls.
* Added ability for translation on-screen menu to multiple languages. Tested on Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4, ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen menu of phone
* Other described in CHANGES file
Testing done by issue tracker users: ibercom, scsiborg, idarwin, TeknoJuce, c0rnoTa.
Tested on production system by Jonn Taylor (jonnt) using phone models: Nortel i2004, 1120E and 1140E.
(closes issue ASTERISK-16890)
Review: https://reviewboard.asterisk.org/r/1243/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358766 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For analog lines, enables Asterisk to use dialtone detection per channel
if an incoming call was hung up before it was answered. If dialtone is
detected, the call is hung up.
no: Disabled. (Default)
yes: Look for dialtone for 10000 ms after answer.
<number>: Look for dialtone for the specified number of ms after answer.
always: Look for dialtone for the entire call. Dialtone may return
if the far end hangs up first.
dialtone_detect=yes
dialtone_detect=5000
dialtone_detect=always
(closes issue ASTERISK-19316)
Reported by: Jeremy Pepper
Patch by: Jeremy Pepper
Tested by: rmudgett,Jeremy Pepper
Review: https://reviewboard.asterisk.org/r/1737/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
send_diversion=no will prevent Diversion headers from being added to SIP
requests. This doesn't prevent Diversion from being added with dialplan
such as with SIPAddHeader.
(closes issue ASTERISK-16862)
Reported by: rsw686
Review: https://reviewboard.asterisk.org/r/1769/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356987 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change permits each verbose destination (consoles, logger) to have its
own concept of what the verbosity level is. The big feature here is that
the logger will now be able to capture a particular verbosity level without
condemning each console to need to suffer that level of verbosity.
Additionally, a stray 'core set verbose' will no longer change what will go
to the log.
Review: https://reviewboard.asterisk.org/r/1599/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds a variable AGIEXITONHANGUP for res_agi. If this variable is
set to "yes" on a channel, AGI() will exit immediately once a channel hangup
has been detected. This was the behavior of AGI() in Asterisk 1.4 and earlier
and is still desired by some people.
Review: https://reviewboard.asterisk.org/r/1734/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355102 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds the auto_force_rport and auto_comedia NAT options. It
also converts the nat= setting to a list of comma-separated combinable
options: no, force_rport, comedia, auto_force_rport, and auto_comedia.
nat=yes remains as an undocumented option equal to
"force_rport,comedia". The first instance of 'yes' or 'no' in the list
stops parsing and overrides any previously set options. If an auto_*
option is specified with its non-auto_ counterpart, the auto setting
takes precedence.
This patch builds upon the patch posted to ASTERISK-17860 by JIRA user
pedro-garcia.
(closes issue ASTERISK-17860)
Review: https://reviewboard.asterisk.org/r/1698/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354597 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch is based on the one by David Vossel, developer extrodinaire, at
https://reviewboard.asterisk.org/r/344/. If multiple peers are defined with the
same host/port, but differing callbackextensions, it chooses the peer with the
matching callbackextension. Since callbackextension creates an outbound
registration with the callbackextension as the Contact address, matching an
incoming request by that (in addition to the host/port) makes a lot of sense.
This patch also adds support for callbackextension to realtime by querying all
peers with callbackextensions on reload and adding registrations for them.
(closes issue ASTERISK-13456)
Review: https://reviewboard.asterisk.org/r/344/
Review: https://reviewboard.asterisk.org/r/1717/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354458 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch removes res_ais and introduces a new module, res_corosync.
The OpenAIS project is deprecated and is now just a wrapper around
Corosync. This module provides the same functionality using the same
core infrastructure, but without the use of the deprecated components.
Technically res_ais could have been used with an AIS implementation other
than OpenAIS, but that is the only one I know of that was ever used.
Review: https://reviewboard.asterisk.org/r/1700/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354046 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When Asterisk is used with various third-party libraries (CURL, PostgresSQL,
many others) that have the ability themselves to use OpenSSL, it is possible
for conflicts to arise in how the OpenSSL libraries are initialized and
shutdown. This patch addresses these conflicts by 'wrapping' the important
functions from the OpenSSL libraries in a new shared library that is part
of Asterisk itself, and is loaded in such a way as to ensure that *all*
calls to these functions will be dispatched through the Asterisk wrapper
functions, not the native functions.
This new library is optional, but enabled by default. See the CHANGES file
for documentation on how to disable it.
Along the way, this patch also makes a few other minor changes:
* Changes MODULES_DIR to ASTMODDIR throughout the build system, in order to
more closely match what is used during run-time configuration.
* Corrects some errors in the configure script where AC_CHECK_TOOLS was used
instead of AC_PATH_PROG.
* Adds a new variable for linker flags in the build system (DYLINK), used for
producing true shared libraries (as opposed to the dynamically loadable
modules that the build system produces for 'regular' Asterisk modules).
* Moves the Makefile bits that handle installation and uninstallation of the
main Asterisk binary into main/Makefile from the top-level Makefile.
* Moves a couple of useful preprocessor macros from optional_api.h to
asterisk.h.
Review: https://reviewboard.asterisk.org/r/1006/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353317 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fix authenticate MESSAGE losing custom headers added by the MESSAGE_DATA
function in the authorization attempt.
* Pass up better From header contents for SIP to use. Now is in the
"display-name" <URI> format expected by MessageSend. (Note that this is a
behavior change that could concievably affect some people.)
* Block user from adding standard headers that are added automatically.
(To, From,...)
* Allow the user to override the Content-Type header contents sent by
MessageSend.
* Decrement Max-Forwards header if the user transferred it from an
incoming message.
* Expand SIP short header names so the dialplan and other code only has to
deal with the full names.
* Documents what SIP expects in the MessageSend(from) parameter.
(closes issue ASTERISK-18992)
Reported by: Yuri
(closes issue ASTERISK-18917)
Reported by: Shaun Clark
Review: https://reviewboard.asterisk.org/r/1683/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352538 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is a feature patch which allows an 'announcement' option to be specified in
musiconhold.conf which should be set to the name of a sound. If a valid sound is
specified for this option, then it will be played on that music on hold class whenever
a channel bound to that class is put on hold as well as when Asterisk is able to detect
that a song has ended before starting the next song (excludes external players).
(closes ASTERISK-18977)
Reported by: Timo Teräs
Patches:
asterisk-moh-announcement.diff uploaded by Timo Teräs (license 5409)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352134 65c4cc65-6c06-0410-ace0-fbb531ad65f3
MixMonitor receives a new option i(channel_variable) which stores the unique id at said
variable. StopMixMonitor now accepts ID as an optional argument, which if included will
make StopMixMonitor specifically target the mixmonitor on that particular channel. CLI
commands and AMI actions have been ammended to work with the IDs as well. In addition,
monitors across a channel can now be listed be listed via CLI command "mixmonitor list
<channel>" which will display all of the mixmonitors active on that channel along with
the files they each have open. Created by Sergio González Martín.
(closes issue ASTERISK-19096)
Reported by: Sergio González Martín
Review: https://reviewboard.asterisk.org/r/1643/
Review: https://reviewboard.asterisk.org/r/1682/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352093 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Adds per-parking lot options comebackcontext and comebackdialtime
* Makes comebacktoorigin settable per parking lot
* Sets a PARKER channel variable when comebacktoorigin is disabled
(closes issue ASTERISK-16643)
Reported by: Mitch Sharp (bluecrow76)
Patches:
asterisk-1.6.2.17.2-park-features-comebackcontext-consolidated-v3.diff by Mitch Sharp (bluecrow76) license 5231
with updates by me.
Review: https://reviewboard.asterisk.org/r/1674
Review: https://reviewboard.asterisk.org/r/963
Reviewed by Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351913 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In order to better handle RTP sources with strictrtp enabled (which is now default in 10)
using the learning mode to figure out new sources when they change is handled by checking
for a number of consecutive (by sequence number) packets received to an rtp struct
based on a new configurable value called 'probation'. Also, during learning mode instead
of liberally accepting all packets received, we now reject packets until a clear source
has been determined.
Review: https://reviewboard.asterisk.org/r/1663/
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Merged revisions 351289 from http://svn.asterisk.org/svn/asterisk/branches/10
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This patch allows the imapserver, imapport, and imapflags settings to be
overridden for any voicemail user. It also documents the settings in
the sample voicemail.conf file, and updates the voicemail schema to
allow storage of those columns.
(closes issue ASTERISK-16489)
Reporter: Hubert Mickael
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1614/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349106 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch moves destruction of sip peers to immediately after the general section of
sip.conf is read so that autocreatepeer setting can be read before deletion of peers.
If autocreatepeer=persist at reload, then peers created by the autocreatepeer setting
will be skipped when purging the current SIP peer list.
(closes ASTERISK-16508)
Reported by: Kirill Katsnelson
Patches:
017797-kkm-persist-autopeers-1.8.patch uploaded by Kirill Katsnelson (license 5845)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349097 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In order to check the availability of the caller's name, app_voicemail will check for an
audio file in <astspooldir>/recordings/callerids/
This change sets a precedent for where to put recordings of names. Currently the idea is
that recordings here could also be used for applications like confbridge and meetme to
find recorded names in this folder from callerid (when another recording isn't available)
(closes issue ASTERISK-18565)
Reporter: Russell Brown
Patches:
r uploaded by Russel Brown (license 6182)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348416 65c4cc65-6c06-0410-ace0-fbb531ad65f3
During testing, it was discovered that there were a number of side effects
introduced by r346391 and subsequent check-ins related to it (r346429,
r346617, and r346655). This included the /main/stdtime/ test 'hanging',
as well as the remote console option failing to receive the appropriate output
after a period of time.
I only backed out the changes to main/ and utils/, as this was adequate
to reverse the behavior experienced.
(issue ASTERISK-18974)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347997 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a caller sends DTMF while the SayUnixTime application is saying the time, The call
would jump to the next extension much like it does during Background(). This patch adds
option 'j' to SayUnixTime which when used employs the old behavior. Also, this patch
allows arguments to sayunixtime to not be used as empty strings in the case of something
like 'sayunixtime(,,,j)' or 'sayunixtime(,,pattern).
(closes issue ASTERISK-16675)
Reported by: jlpedrosa
Patches:
patch_SayUnixTime_noJump.patch uploaded by jlpedrosa (license 5959)
Review: https://reviewboard.asterisk.org/r/956/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347866 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds two new menu features to app_confbridge, admin_toggle_menu_
participants and participant_count. The admin action will globally mute /
unmute all non-admin participants on a converence, while the participant
count simply exposes the existing participant count function to the
conference bridge menu.
This also adds configuration options to change the sound played when the
conference is globally muted / unmuted, as well as the necessary config
hooks to place these functions in the DTMF menus.
(closes issue ASTERISK-18204)
Reported by: Kevin Reeves
Tested by: Matt Jordan
Patches:
app_confbridge.c.patch.txt, conf_config_parser.c.patch.txt,
confbridge.h.patch.txt uploaded by Kevin Reeves (license 6281)
Review: https://reviewboard.asterisk.org/r/1518/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345560 65c4cc65-6c06-0410-ace0-fbb531ad65f3
You can now use the built in variables , , and
within a dynamic weight. For example, this could be useful when you want
to pass requested lookup number to the SHELL() function which could be
used to execute a script to dynamically set the weight of the result.
(Closes issue ASTERISK-13657)
Reported by: Joel Vandal
Tested by: Leif Madsen, Russell Bryant
Patches:
asterisk-1.6-dundi-varhead.patch uploaded by Joel Vandal (License #5374)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343693 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/10
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r341580 | irroot | 2011-10-20 19:13:23 +0200 (Thu, 20 Oct 2011) | 15 lines
Add option to check state when state is unknown
r341486 reverts r325483 this is a rework of the patch.
optimize to minimize load.
add option check_state_unknown to control whether a member with unknown
device state is checked there is a small % chance that calls will be sent
to the member when they on a call.
app_queue will see a device with unknown state as available and does not
try verify the state without this option enabled.
Review: https://reviewboard.asterisk.org/r/1535/
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Added func FAXOPT(faxdetect)=yes,cng,t38[,timeout]/no
to enable dialplan faxdetect allowing more flexibility.
as soon as a fax tone is detected the framehook is removed.
there is a penalty involved in running this framehook on
non G711 channels as they will be transcoded.
CNG tone is suppresed using the SQUELCH flag to allow
WaitForNoise to be run on the channel to detect Voice.
(Closes issue ASTERISK-18569)
Reported by: Myself
Reviewed by: Matthew Nicholson, Kevin Fleming
Review: https://reviewboard.asterisk.org/r/1116/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339465 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/10
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r337595 | jrose | 2011-09-22 10:35:50 -0500 (Thu, 22 Sep 2011) | 12 lines
Generate Security events in chan_sip using new Security Events Framework
Security Events Framework was added in 1.8 and support was added for AMI to generate
events at that time. This patch adds support for chan_sip to generate security events.
(closes issue ASTERISK-18264)
Reported by: Michael L. Young
Patches:
security_events_chan_sip_v4.patch (license #5026) by Michael L. Young
Review: https://reviewboard.asterisk.org/r/1362/
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r337597 | jrose | 2011-09-22 10:47:05 -0500 (Thu, 22 Sep 2011) | 10 lines
Forgot to svn add new files to r337595
Part of Generating security events for chan_sip
(issue ASTERISK-18264)
Reported by: Michael L. Young
Patches:
security_events_chan_sip_v4.patch (License #5026) by Michael L. Young
Reviewboard: https://reviewboard.asterisk.org/r/1362/
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r337219 | oej | 2011-09-21 11:32:50 +0200 (Ons, 21 Sep 2011) | 13 lines
Make ast_pbx_run() not default to s@default if extension is not found
Review: https://reviewboard.asterisk.org/r/1446/
This is a bug - or architecture mistake - that has been in Asterisk for a
very long time. It was exposed by the AMI originate action and possibly
some other applications. Most channel drivers checks if an extension
exists BEFORE starting a pbx on an inbound call, so most calls will
not depend on this issue.
Thanks everyone involved in the review and on IRC and the mailing list
for a quick review and all the feedback.
(closes issue ASTERISK-18578)
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r337178 | oej | 2011-09-21 10:51:41 +0200 (Ons, 21 Sep 2011) | 14 lines
Change strictrtp option to default to yes in the RTP module
Suggested by Kapejod on Facebook
Review: https://reviewboard.asterisk.org/r/1448/
(closes issue ASTERISK-18587)
Thanks for quick feedback to kpfleming and Tilghman
--Denna och nedanstående rader kommer inte med i loggmeddelandet--
M CHANGES
M configs/rtp.conf.sample
M res/res_rtp_asterisk.c
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r336936 | irroot | 2011-09-20 18:51:59 +0200 (Tue, 20 Sep 2011) | 14 lines
Allow Setting Auth Tag Bit length Based on invite or config option
Update the SIP SRTP API to allow use of 32 or 80 bit taglen.
Curently only 80 bit is supported.
The outgoing invite will use the taglen of the incoming invite preventing
one-way audio.
(Closes issue ASTERISK-17895)
Review: https://reviewboard.asterisk.org/r/1173/
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r336042 | oej | 2011-09-15 14:46:38 +0200 (Tor, 15 Sep 2011) | 12 lines
Meetme: Introducing a new option "k" to kill a conference if there's only a single member left.
When using Meetme as a modular call bridge from third party applications, it's handy to make
it behave like a normal call bridge. When the second to last person exists, the last person
will be kicked out of the conference when this option is enabled.
(closes issue ASTERISK-18234)
Review: https://reviewboard.asterisk.org/r/1376/
Patch by oej, sponsored by ClearIT, Solna, Sweden
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This permits the list of codecs to be specified in one configuration line,
instead of two or more, generally with the aim of either allowing all codecs
with the exception of a few or disallowing most but permitting a few.
Review: https://reviewboard.asterisk.org/r/1411/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334574 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds a CDR option to cdr.conf that will allow CDR files to log calls ending
with congestion in a way that is unique from other unanswered calls.
(closes issue ASTERISK-14842)
Reported by: Alec Davis
Patches:
cdr_congestion.diff.txt (License #5546) patch uploaded by Alec Davis
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332760 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r332101 | rmudgett | 2011-08-16 12:17:28 -0500 (Tue, 16 Aug 2011) | 140 lines
Merged revisions 332100 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r332100 | rmudgett | 2011-08-16 11:31:36 -0500 (Tue, 16 Aug 2011) | 133 lines
Fix multiple parking issues.
JIRA ASTERISK-17183
Multi-parkinglot directs calls to wrong parkinglot.
JIRA ASTERISK-17870
Cannot retrieve parked calls.
JIRA ASTERISK-17430
ParkedCall() with no extension should pickup first available call and does not.
JIRA AST-576
Issues with parking lots
* Removed searching for parking lots by extension. Parking lots can only
be found by the parking lot name since parking lot access extensions and
spaces are not guaranteed to be unique.
* Added parking_lot_name option to the Park and ParkedCall applications.
Updated documentation for Park and ParkedCall applications.
* Add parkext_exclusive configuration option to make parking entry
extensions specify which parking lot they access.
(closes issue ASTERISK-17183)
Reported by: David Cabrejos
Tested by: rmudgett, David Cabrejos
(closes issue ASTERISK-17870)
Reported by: Remi Quezada
(closes issue ASTERISK-17430)
Reported by: Philippe Lindheimer
JIRA ASTERISK-17452
Parking_offset not used
JIRA AST-624
'next' setting for findslot does nothing
* Reimplemented since findslot feature option broken by -r114655.
(closes issue ASTERISK-17452)
Reported by: David Woolley
Tested by: rmudgett
JIRA ASTERISK-15792
Dialplan continues execution after transfer to park.
This happens for DTMF attended transfer, DTMF blind transfer, and DTMF
one-touch-parking if the party initiating these features also initiated
the call.
* Fixed the return code from the affected builtin features when parking a
call.
(closes issue ASTERISK-15792)
Reported by: Mat Murdock
Tested by: rmudgett, twilson
JIRA AST-607
The courtesytone is not playing to the expected call when picking up a
parked call.
This is mostly a documentation problem. However, the option is not reset
to the default when features.conf is reloaded.
* Updated features.conf.sample documentation for courtesytone and
parkedplay options.
* Reset the parkedplay option to default when features.conf is reloaded.
JIRA AST-615
AMI Park action followed by features reload results in orphaned channels
in parking lot.
* Reloading features.conf will not touch parking lots that have calls
still parked in them. Reload again at a later time.
Misc additional fixes:
* Added unit test for parking lot dialplan usage checking.
* Made update connected line when a parked call is retrieved from a
parking lot.
* Made retrieved parked call stop ringing or MOH depending upon how the
call was waiting in the parking lot.
* Made CLI "features show" indicate if the parking lot is enabled for use.
* Added PARKINGDYNEXTEN channel variable to allow dynamic parking lots to
specify the parking lot access extension.
* Made AMI ParkedCalls action ParkedCall events have a Parkinglot header.
* Made AMI ParkedCalls action ParkedCallsComplete event have a Total
header.
* Fixed potential deadlock from AMI Park action holding channel locks
while calling masq_park_call().
* Fixed several places where ast_strdupa() were used inside of loops.
(Mostly fixed by refactoring the loop body into its own function.)
* Fixed copy_parkinglot() copying too much from the source parking lot.
Extracted the parking lot configuration settings into struct
parkinglot_cfg.
* Refactored courtesytone playing code to put the channel not playing the
tone in autoservice.
* Fix when pbx-parkingfailed is played that the other channel is put in
autoservice if it exists.
* Fixed parkinglot reference leak in parked_call_exec() error paths.
* Fixed parkinglot_unref() use of parkinglot after it was unreffed.
* Made destroy the struct ast_parkinglot parkings lock when done.
* Refactored the features.conf parking lot configuration code to eliminate
redundancy.
* Fixed feature reload to better protect parking lots.
* Fixed parking lot container reference leak in handle_parkedcalls().
* Fixed the total count in handle_parkedcalls().
Review: https://reviewboard.asterisk.org/r/1358/
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r332022 | mnicholson | 2011-08-16 09:40:37 -0500 (Tue, 16 Aug 2011) | 16 lines
In 10 and trunk this option is disabled by default.
Merged revisions 332021 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r332021 | mnicholson | 2011-08-16 09:20:43 -0500 (Tue, 16 Aug 2011) | 7 lines
Added the 'storesipcause' option to sip.conf to allow the user to disable the
setting of HASH(SIP_CAUSE,<chan name>) on the channel.
Having chan_sip set HASH(SIP_CAUSE,<chan name>) on the channel carries a
significant performance penalty because of the usage of the MASTER_CHANNEL()
dialplan function.
AST-580
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In order to address a compatability issue with certain features on certain devices
which rely on display name content to change behavior, initreqprep in chan_sip.c
has been changed to no longer substitute cid_number into the display name when
cid_name isn't present. Instead, it will send no display name in that case.
(closes issue ASTERISK-16198)
Reported by: Walter Doekes
Review: https://reviewboard.asterisk.org/r/1341/
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r331097 | twilson | 2011-08-08 17:59:01 -0500 (Mon, 08 Aug 2011) | 5 lines
Bump the AMI protocol version to 1.2
As a result of converting Unlink events that were missed in the AMI
1.1 update to Bridge events, the AMI protocol version is being incremented.
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r331041 | twilson | 2011-08-08 16:12:51 -0500 (Mon, 08 Aug 2011) | 6 lines
Replace AMI Unlink events with Bridge events
A previous update converted some of the Link and Unlink events to
Bridge events, but a couple of Unlink events were missed. This patch
rectifies the situation.
(closes issues ASTERISK-17455)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There were some bugs in the very ancient version of Berkeley DB that Asterisk
used. Instead of spending the time tracking down the bugs in the Berkeley code
we move to the much better documented SQLite 3.
Conversion of the old astdb happens at runtime by running the included
astdb2sqlite3 utility. The ast_db API with SQLite 3 backend should behave
identically to the old Berkeley backend, but in the future we could offer a
much more robust interface.
We do not include the SQLite 3 library in the source tree, but instead rely
upon the distribution-provided libraries. SQLite is so ubiquitous that this
should not place undue burden on administrators.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds a new action, FilterAdd to the manager interface that allows control over event filters for the current session
(closes issue ASTERISK-16795)
Reported by: kobaz
Tested by: kobaz,loloski
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326267 65c4cc65-6c06-0410-ace0-fbb531ad65f3
terminal). Can be enabled on a channel by setting FAXOPT(gateway)=yes in the
dialplan.
Big thanks to irroot for porting this code to use the framehooks api.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325816 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Added general option negative_penalty_invalid default off. when set
members are seen as invalid/logged out when there penalty is negative.
for realtime members when set remove from queue will set penalty to -1.
* Added queue option autopausedelay when autopause is enabled it will be
delayed for this number of seconds since last successful call if there
was no prior call the agent will be autopaused immediately.
* Added member option ignorebusy this when set and ringinuse is not
will allow per member control of multiple calls as ringinuse does for
the Queue.
- Mark QUEUE_MEMBER_PENALTY Deprecated it never worked for realtime members
- QUEUE_MEMBER is now R/W supporting setting paused, ignorebusy and penalty.
(closes issue ASTERISK-17421)
(closes issue ASTERISK-17391)
Reported by: irroot
Tested by: irroot, jrose
Review: https://reviewboard.asterisk.org/r/1119/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325483 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
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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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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287648 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
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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/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
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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/
........
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
Jonathan Rose
Matthew Jordan
Jason Parker
Richard Mudgett
Mark Michelson
Kinsey Moore
Olle Johansson
Joshua Colp
Russell Bryant
David M. Lee
Kevin Harwell
Michael L. Young
Alec L Davis
Terry Wilson
Kevin P. Fleming
Igor Goncharovskiy
Alexandr Anikin
Sean Bright
Mark Murawki
Tilghman Lesher
Walter Doekes
Paul Belanger
Leif Madsen
Gregory Nietsky
Matthew Nicholson
David Vossel
Jeff Peeler
Andrew Parisio
David Ruggles
Tzafrir Cohen
Bradley Latus