add missing argument "rtt" and "status" to the documentation
The change to the dtd file allow an enumlist to contain one or many
configOptionToEnum or enum.
This is different from the previous patch I submitted when you could have a
configOptionToEnum or (a configOptionToEnum followed by one or manu enums) or
(one or many enums)
ASTERISK-28626
Change-Id: Ia71743ee7ec813f40297b0ddefeee7909db63b6d
This reverts commit 7e3015d779.
Reason for revert: Regression in XML validation.
validity error : Content model of enumlist is not determinist:
(configOptionToEnum | (configOptionToEnum , enum+) | enum+)
As we are preparing to do releases and this is not critical
I am reverting this for now until resolved.
Change-Id: I30c2295f9d7f0a0475674ee77071a7ebabf5b83f
If an AMI user without the "system" authorization calls the
Originate AMI command with the Originate application,
the second Originate could run the "System" command.
Action: Originate
Channel: Local/1111
Application: Originate
Data: Local/2222,app,System,touch /tmp/owned
If the "system" authorization isn't set, we now block the
Originate app as well as the System, Exec, etc. apps.
ASTERISK-28580
Reported by: Eliel Sardañons
Change-Id: Ic4c9dedc34c426f03c8c14fce334a71386d8a5fa
When user wants to send json data, the default Content-Type header
is incorect (application/x-www-form-urlencoded). This patch allows
to set any custom headers so the Content-Type header can be
overriden. User can set multiple headers by multiple calls of
curlopt(). This approach is not consistent with other parameters,
but is more readable in dialplan than one call with multiple
headers.
ASTERISK-28613
Change-Id: I4dd68c3f4e25362ef941d73a3861f58348dcfbf9
This reverts commit fd2e8d0da7.
Reason for revert: Problematic for users who store their voicemail
on network storage devices, or share voicemail storage between
multiple Asterisk instances.
ASTERISK-28567 #close
Change-Id: I3ff4ca983d8e753fe2971f3439bd154705693c41
Add a new dialplan function PJSIP_MOH_PASSTHROUGH that allows
the on-hold behavior to be controlled on a per-call basis
ASTERISK-28542 #close
Change-Id: Iebe905b2ad6dbaa87ab330267147180b05a3c3a8
Added "like" support for 'core show taskprocessors'. Now you
can specify a specific set of taskprocessors (or just one) by
adding the keyword "like" to the above command, followed by
your search criteria.
Change-Id: I021e740201e9ba487204b5451e46feb0e3222464
Allow the list of files to be played to be provided explicitly in the
music class's configuration. The primary driver for this change is to
allow URLs to be used for MoH.
Change-Id: I9f43b80b43880980b18b2bee26ec09429d0b92fa
Added two new CLI commands to reset stats for taskprocessors. You can
reset stats for a single, specific taskprocessor ('core reset
taskprocessor <taskprocessor>'), or you can reset all taskprocessors
('core reset taskprocessors'). These commands will reset the counter for
the number of tasks processed as well as the max queue size.
Change-Id: Iaf17fc4ae29396ab0c6ac92408fc7bdc2f12362d
This change adds support to the JITTERBUFFER dialplan function
for audio and video synchronization. When enabled the RTCP SR
report is used to produce an NTP timestamp for both the audio and
video streams. Using this information the video frames are queued
until their NTP timestamp is equal to or behind the NTP timestamp
of the audio. The audio jitterbuffer acts as the leader deciding
when to shrink/grow the jitterbuffer when adaptive is in use. For
both adaptive and fixed the video buffer follows the size of the
audio jitterbuffer.
ASTERISK-28533
Change-Id: I3fd75160426465e6d46bb2e198c07b9d314a4492
This change adds H.265/HEVC as a known codec and creates a cached
"h265" media format for use.
Note that RFC 7798 section 7.2 also describes additional SDP
parameters. Handling of these is not yet supported.
ASTERISK-28512
Change-Id: I26d262cc4110b4f7e99348a3ddc53bad0d2cd1f2
This fix allows a realtime moh class to be unregistered from the command
line. This is useful when the contents of a directory referenced by a
realtime moh class have changed.
The realtime moh class is then reloaded on the next request and uses the
new directory contents.
ASTERISK-17808
Change-Id: Ibc4c6834592257c4bb90601ee299682d15befbce
The Channel resource has a new sub-resource "externalMedia".
This allows an application to create a channel for the sole purpose
of exchanging media with an external server. Once created, this
channel could be placed into a bridge with existing channels to
allow the external server to inject audio into the bridge or
receive audio from the bridge.
See https://wiki.asterisk.org/wiki/display/AST/External+Media+and+ARI
for more information.
Change-Id: I9618899198880b4c650354581b50c0401b58bc46
The UnicastRTP channel driver provided by chan_rtp now accepts
"<hostname>:<port>" as an alternative to "<ip_address>:<port>"
in the destination. The first AAAA (preferred) or A record resolved
will be used as the destination. The lookup is synchronous so beware
of possible dialplan delays if you specify a hostname.
Change-Id: Ie6f95b983a8792bf0dacc64c7953a41032dba677
There are 4 scenarios to consider when capturing audio from a channel
with an audiohook:
1. There is no rx and no tx audio, so return nothing.
2. There is rx but no tx audio, so return rx.
3. There is tx but no rx audio, so return tx.
4. There is rx and tx audio, so mix them and return.
The file passed as the primary argument to MixMonitor will be written to
in scenarios 2, 3, and 4. However, if you pass the r() and t() options
to MixMonitor, a frame will only be written to the r() file if there was
rx audio and a frame will only be written to the t() file if there was
tx audio.
If you subsequently take the r() and t() files and try to mix them, the
sides of the conversation will 'drift' and be non-representative of the
user experience.
This patch adds a new 'S' option to MixMonitor that injects a frame of
silence on either the r() side or the t() side of the channel so that
when later mixed, there is no such drift.
Change-Id: Ibf5ed73a811087727bd561a89a59f4447b4ee20e
Asterisk headers are no longer installed and uninstalled
automatically when performing a "make install" or a
"make uninstall". To install/uninstall the headers, use
"make install-headers" and "make uninstall-headers".
The headers also continue to be uninstalled when performing a
"make uninstall-all".
Also corrects an issue where /usr/include/asterisk.h was never
being removed at all.
Change-Id: Ia7399f3a0203a4825fc4a9f43b9034dae9a2b643
Changes made to apps/Makefile to optionally build all three app_voicemail
variations at the same time: 1) file (default), 2) odbc, and 3) imap.
This functionality was requested by users. modules.conf.sample warns the
user to make sure only one voicemail is loaded at a time.
Change-Id: Iba3cd8ffb4b7e8b1c64a11dd383e1eafcd3ed0e7
Previously, when a Transfer (REFER) was performed, chan_pjsip would set
the TRANSFERSTATUS to SUCCESS when the REFER was queued up. This did not
reflect a successful/unsuccessful transfer the way chan_sip did.
Added a callback module to process the refer subscription information.
Now depends on res_pjsip_pubsub so call transfer progress can be monitored
and reported
ASTERISK-26968 #close
Reported-by: Dan Cropp
Change-Id: If6c27c757c66f71e8b75e3fe49da53ebe62395dc
This change adds support for larger TLS certificates by allowing
OpenSSL to fragment the DTLS packets according to the configured
MTU. By default this is set to 1200.
This is accomplished by implementing our own BIO method that
supports MTU querying. The configured MTU is returned to OpenSSL
which fragments the packet accordingly. When a packet is to be
sent it is done directly out the RTP instance.
ASTERISK-28018
Change-Id: If2d5032019a28ffd48f43e9e93ed71dbdbf39c06
AttendedTransfer queues up attended transfer to the given extension.
This application can be useful with Custom Dynamic Features.
For example to make attended transfer to a predefined number.
features.conf
;;;
[applicationmap]
my_atxfer => *7,self,GoSub,"my_atxfer,s,1",default
;;;
extensions.conf
;;;
[globals]
DYNAMIC_FEATURES=my_atxfer
TRANSFER_CONTEXT=my_transfer
[my_atxfer]
exten => s,1,AttendedTransfer(1234567890)
same => n,Return()
[my_transfer]
include => default
;;;
This application also can be used to completly redefine Attended transfer
feature using dialplan. For example:
features.conf
;;;
[featuremap]
atxfer => *7
[applicationmap]
custom_atxfer => *2,self,GoSub,"custom_atxfer,s,1",default
;;;
extensions.conf
;;;
[globals]
DYNAMIC_FEATURES=custom_atxfer
TRANSFER_CONTEXT=my_transfer
[custom_atxfer]
exten => s,1,
same => n,Playback(pbx-transfer)
same => n,Read(dest,dial,10,i,3,3)
same => n,AttendedTransfer(${dest})
same => n,Return()
[my_transfer]
include => default
;;;
Change-Id: Ie5cfa455d0813cffd5c85a6fb117f07d8f0b903b
BlindTransfer redirects all channels currently bridged to the
caller channel to the specified destination.
This application can be useful with Custom Dynamic Features.
For example to make blind transfer to a predefined number.
features.conf
;;;
[applicationmap]
my_blindxfer => *6,self,GoSub,"my_blindxfer,s,1",default
;;;
extensions.conf
;;;
[globals]
DYNAMIC_FEATURES=my_blindxfer
[my_blindxfer]
exten => s,1,BlindTransfer(1234567890,default)
same => n,Return()
;;;
This application also can be used to completly redefine Blind transfer
feature using dialplan. For example:
features.conf
;;;
[featuremap]
blindxfer =>
[applicationmap]
custom_blindxfer => ##,self,GoSub,"custom_blindxfer,s,1",default
;;;
extensions.conf
;;;
[globals]
DYNAMIC_FEATURES=custom_blindxfer
[custom_blindxfer]
exten => s,1,
same => n,Playback(pbx-transfer)
same => n,Read(dest,dial,10,i,3,3)
same => n,BlindTransfer(${dest},default)
same => n,Return()
;;;
Change-Id: I9d55e7f69ccfd4472dec00d62771d6de8803215a
One of the change files doesn't conform to the format that the release
scripts need in order to parse it.
Change-Id: Ie0b634cf27e4cbc671b9fe92993b6f2ecf60254c
This patch adds the 'p' option.
The extension entered will be considered complete when a # is entered.
Change-Id: If77c40c9c8b525885730821e768f5dea71cf04c1
You can now add the "include_local_address" flag to an entry in
rtp.conf "[ice_host_candidates]" to include both the advertized
address and the local address in ICE negotiation:
[ice_host_candidates]
192.168.1.1 = 1.2.3.4,include_local_address
This causes both 192.168.1.1 and 1.2.3.4 to be advertized.
Change-Id: Ide492cd45ce84546175ca7d557de80d9770513db
When producing a combined REMB value the normal behavior
is to have a REMB value which is unique for each sender
based on all of their receivers. This can result in one
sender having low bitrate while all the rest are high.
This change adds "all" variants which produces a bridge
level REMB value instead. All REMB reports are combined
together into a single REMB value that is the same for
each sender.
ASTERISK-28401
Change-Id: I883e6cc26003b497c8180b346111c79a131ba88c
Added RINGTIME, RINGTIME_MS, PROGRESSTIME, PROGRESSTIME_MS variables filled
at the earliest received PROGRESS or RINGING.
Added millisecond versions of DIALEDTIME and ANSWEREDTIME.
Added millisecond versions of ast_channel_get_up_time and
ast_channel_get_duration in channel.c.
ASTERISK-28363
Change-Id: If95f1a7d8c4acbac740037de0c6e3109ff6620b1
Added a new PJSIP global setting called norefersub.
Default is true to keep support working as before.
res_pjsip_refer: Configures PJSIP norefersub capability accordingly.
Checks the PJSIP global setting value.
If it is true (default) it adds the norefersub capability to PJSIP.
If it is false (disabled) it does not add the norefersub capability
to PJSIP.
This is useful for Cisco switches that do not follow RFC4488.
ASTERISK-28375 #close
Reported-by: Dan Cropp
Change-Id: I0b1c28ebc905d881f4a16e752715487a688b30e9
This changes the way that we handle adding changes to CHANGES and
UPGRADE.txt. The reason for this is because whenever someone needed to
make a change to one of these files and someone else had already done
so, you would run into merge conflicts. With this new setup, there will
never be merge conflicts since all changes will be documented in the
doc/<file>-staging directory. The release script is now responsible for
merging all of these changes into the appropriate files.
There is a special format that these files have to follow in order to be
parsed. The files do not need to have a meaningful name, but it is
strongly recommended. For example, if you made a change to pjsip, you
may have something like this "res_pjsip_relative_title", where
"relative_title" is something more descriptive than that. Inside each
file, you will need a subject line for your change, followed by a
description. There can be multiple subject lines. The file may look
something like this:
Subject: res_pjsip
Subject: Core
A description that explains the changes made and why. The release
script will handle the bulleting and section separators!
You can still separate with new lines within your
description.
The headers ("Subject" and "Master-Only") are case sensative, but the
value for "Master-Only" ("true" or "True") is not.
For more information, check out the wiki page:
https://wiki.asterisk.org/wiki/display/AST/CHANGES+and+UPGRADE.txt
ASTERISK-28111 #close
Change-Id: I19cf4b569321c88155a65e9b0b80f6d58075dd47
This is the first step in changing the release process so that changes
made to the CHANGES and UPGRADE.txt files do not result in merge
conflicts every time multiple people modify these files. The changes
made will go in these new directories: doc/CHANGES-staging and
doc/UPGRADE-staging. The README.md files explain how things will work,
but here's a little overview. When you make a change that would go in
either CHANGES or UPGRADE.txt, this should instead be documented in a
new file in the doc/CHANGES-staging or doc/UPGRADE-staging directory,
respectively. The format will look like this:
Subject: res_pjsip
A description that explains the changes made and why. The release
script will handle the bulleting and section separators! The
'Subject:' header is case-sensitive.
You can still separate with new lines within your description.
Subject: res_ari
Master-Only: true
You can have more than one subject, and they don't have to be the
same! Also, the 'Master-Only' header should always be true and is
also case-sensitive (but the value is not - you can have 'true' or
'True'). This header will only ever be present in the master branch.
For more information, check out the wiki page:
https://wiki.asterisk.org/wiki/display/AST/CHANGES+and+UPGRADE.txt
This is an initial change for ASTERISK_28111. Functionally, this will
make no difference, but it will prep the directories for when the
changes from CHANGES and UPGRADE.txt are extracted.
Change-Id: I8d852f284f66ac456b26dcb899ee46babf7d15b6
* Consistently use spaces in rest-api-templates/asterisk_processor.py.
* Exclude third-party from docs/full-en_US.xml.
* Add docs/full-en_US.xml to .gitignore.
* Use list() to convert python3 view.
* Use python3 print function.
* Replace cmp() with equivalent equation.
* Replace reference to out of scope subtype variable with name
parameter.
* Use unescaping triple bracket notation in mustache templates where
needed. This causes behavior of Python2 to be maintained when using
Python3.
* Fix references to has_websocket / is_websocket in
res_ari_resource.c.mustache.
* Update calculation of has_websocket to use any().
* Use unicode mode for writing output file in transform.py.
* Replace 'from swagger_model import *' with explicit import of required
symbols.
I have not tested spandspflow2pcap.py or voicemailpwcheck.py, only the
print syntax has been fixed.
Change-Id: If5c5b556a2800d41a3e2cfef080ac2e151178c33
This document is out of date and is superseded by content on the
Asterisk wiki.
ASTERISK-24386 #close
Change-Id: Idbf95b27b096c205251e1bbb560c79224ba81822
In multi-party bridges, Asterisk currently supports two video modes:
* Follow the talker, in which the speaker with the most energy is shown
to all participants but the speaker, and the speaker sees the
previous video source
* Explicitly set video sources, in which all participants see a locked
video source
Prior to this patch, ARI had no ability to manipulate the video source.
This isn't important for two-party bridges, in which Asterisk merely
relays the video between the participants. However, in a multi-party
bridge, it can be advantageous to allow an external application to
manipulate the video source.
This patch provides two new routes to accomplish this:
(1) setVideoSource: POST /bridges/{bridgeId}/videoSource/{channelId}
Sets a video source to an explicit channel
(2) clearVideoSource: DELETE /bridges/{bridgeId}/videoSource
Removes any explicit video source, and sets the video mode to talk
detection
ASTERISK-26595 #close
Change-Id: I98e455d5bffc08ea5e8d6b84ccaf063c714e6621
This updates func_channel.c and main/message.c to use a generic xpointer
include instead of including info from each channel driver. Now the
name attribute of info is CHANNEL or CHANNEL_EXAMPLES to be included in
documentation for func_channel. Setting the name attribute of info to
MessageToInfo or MessageFromInfo causes it to be included in the
MessageSend application and AMI action.
Change-Id: I89fd8276a3250824241a618009714267d3a8d1ea
* Following the example of the PJSIP channel driver, the channel
technology specific documentation has been moved to the respective
channel drivers that provide that functionality. This has the benefit
of locating the documentation of items with those modules that provide
it.
* Examples of using the CHANNEL function for both standard items as well
as for PJSIP have been added.
* The 'max_forwards' standard item has been documented.
Change-Id: Ifaa79a232c8ac99cf8da6ef6cc7815d398b1b79b
Previous versions of Asterisk processed command-line options before
processing asterisk.conf. This meant that if an option was set in
asterisk.conf, it could not be overridden with the equivelent command
line option. This change causes Asterisk to process the command-line
twice. First it processes options that are needed to load asterisk.conf,
then it processes the remaining options after the config is read.
This changes the function of -X slightly. Previously using -X without
disabling execincludes in asterisk.conf caused #exec to be usable in any
config. Now -X only enables #exec for the load of asterisk.conf, if it
is wanted in the rest of the system it must be enabled with execincludes
in asterisk.conf. Updated 'asterisk -h' and 'man asterisk' to reflect
the limited function of -X.
ASTERISK-25042 #close
Reported by: Corey Farrell
Change-Id: I1450d45c15b4467274b871914d893ed4f6564cd7
Moved contrib/asterisk-ng-doxygen to doc/asterisk-ng-doxygen.in
Changed /Makefile to copy asterisk-ng-doxygen.in to
asterisk-ng-doxygen then modify it with version instead of
modifying asterisk-ng-doxygen directly. Updated clean
targets as well.
Updated /.gitignore and doc/.gitignore.
Change-Id: I38712d3e334fa4baec19d30d05de8c6f28137622
Create the directory './doc/rest-api' at the start of 'make ari-stubs'
to prevent an error when documentation is generated. The directory is
also added to git ignores.
ASTERISK-25027
Reported by: Corey Farrell
Change-Id: Iaccc7f0138501c23aa78feaca2f3cce9e68cbc1b
Add the .gitignore and .gitreview files to the asterisk repo.
NB: You can add local ignores to the .git/info/exclude file
without having to do a commit.
Common ignore patterns are in the top-level .gitignore file.
Subdirectory-specific ignore patterns are in their own .gitignore
files.
Change-Id: I842a1588ff27d8a0189f12d597f0a7af033d6c69
Tested-by: George Joseph
This patch adds support for an <example /> tag in the XML documentation schema.
For CLI help, this doesn't change the formatting too much:
- Preceeding white space is removed
- Unlike with para elements, new lines are preserved
However, having an <example /> tag in the XML schema allows for the wiki
documentation generation script to surround the documentation with {code} or
{noformat} tags, generating much better content for the wiki - and allowing us
to put dialplan examples (and other code snippets, if desired) into the
documentation for an application/function/AMI command/etc.
Review: https://reviewboard.asterisk.org/r/3807/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419822 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Fixes typos of "transfered" instead of "transferred" in various code. Fixes incorrect gosub param help text for app_queue.
Fixes Asterisk man pages containing unquoted minus signs. Adds note about the "textsupport" option in sip.conf.sample.
(issue ASTERISK-23061)
(issue ASTERISK-23028)
(issue ASTERISK-23046)
(issue ASTERISK-23027)
(closes issue ASTERISK-23061)
(closes issue ASTERISK-23028)
(closes issue ASTERISK-23046)
(closes issue ASTERISK-23027)
Reported by: Eugene, Jeremy Laine, Denis Pantsyrev
Patches:
transferred.patch uploaded by Jeremy Laine (license 6561)
hyphen.patch uploaded by Jeremy Laine (license 6561)
sip.conf.sample.patch uploaded by Eugene (license 6360)
........
Merged revisions 405791 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 405792 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 405829 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405830 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Bridges have two new optional properties, a creator and a name.
Certain consumers of bridges will automatically provide bridges that
they create with these properties. Examples include app_bridgewait,
res_parking, app_confbridge, and app_agent_pool. In addition, a name
may now be provided as an argument to the POST function for creating
new bridges via ARI.
(closes issue AFS-47)
Review: https://reviewboard.asterisk.org/r/3070/
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Merged revisions 404042 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404043 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This prevents XML documentation duplication by expanding channel and
bridge snapshot tags into channel and bridge snapshot parameter sets
with a given prefix or defaulting to no prefix. This also prevents
documentation from becoming fractured and out of date by keeping all
variations of the documentation in template form such that it only
needs to be updated once and keeps maintenance to a minimum.
Review: https://reviewboard.asterisk.org/r/2708/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395985 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch allows a module to define its configuration in XML in source, such
that it can be parsed by the XML documentation engine. Documentation is
generated in a two-pass approach:
1. The documentation is first generated from the XML pulled from the source
2. The documentation is then enhanced by the registration of configuration
options that use the configuration framework
This patch include configuration documentation for the following modules:
* chan_motif
* res_xmpp
* app_confbridge
* app_skel
* udptl
Two new CLI commands have been added:
* config show help - show configuration help by module, category, and item
* xmldoc dump - dump the in-memory representation of the XML documentation to
a new XML file.
Review: https://reviewboard.asterisk.org/r/2278
Review: https://reviewboard.asterisk.org/r/2058
patches:
on review 2058 uploaded by twilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381527 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Again, since res_jabber/res_xmpp have duplicate APIs, their documentation ref
links have to specify which reference they're referring to. The various
documentation parsers can interpret the module attribute however they want
in order to construct the appropriate links.
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Merged revisions 379228 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379229 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r379209 | mjordan | 2013-01-16 09:27:44 -0600 (Wed, 16 Jan 2013) | 8 lines
Add module tags to documentation for res_jabber/res_xmpp
Since res_jabber/res_xmpp provide the same APIs (app/func/manager/etc.),
the XML documentation for each needs to call out which module is providing
the documentation. The module attribute has been added to the various XML
fragments for this purpose.
........
r379210 | mjordan | 2013-01-16 09:30:20 -0600 (Wed, 16 Jan 2013) | 4 lines
Update the dtd to actually *support* the module attribute in all elements
Mea culpa.
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Merged revisions 379209-379210 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379211 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Whitespace, doc-blocks, spelling, case, missing and incorrect tags.
* Add cleanup to Makefile for the Doxygen configuration update
* Start updating Doxygen configuration for cleaner output
* Enable inclusion of configuration files into documentation
* remove mantisworkflow...
* update documentation README
* Add markup to Tilghman's email and talk with him about updating his email, he knows...
* no code changes on this commit other than the mentioned Makefile change
(issue ASTERISK-20259)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In r294740, the CODING-GUIDELINES was removed from the doc folder in favor
of the content on the Asterisk wiki. Some folks still look in the doc folder
initially for coding guideline suggestions; as such, this patch adds a
CODING-GUIDELINES file back into the doc folder. The content of the file
merely points to the correct page on the Asterisk wiki where the coding
guidelines currently live.
(closes issue ASTERISK-20279)
Reported by: Andrew Latham
Patches:
CODING-GUIDELINES.diff uploaded by Andrew Latham (license 5985)
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Merged revisions 371961 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 371962 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 371963 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371964 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A number of applications/AMI commands in Asterisk have specific behavioral
differences depending on the resource or channel technology those
applications are executed on. For example, the MessageSend application/
command is technology agnostic, but how the channel drivers that support
that functionality behave is dependant on the protocols and channel
driver implementation. Prior to this patch, those details were either
documented in the application/command documentation itself, or were left
undocumented.
This patch adds a new element to the documentation schema, <info/>. An info
node is essentially a piece of technology specific reference information that
can be included by any top level XML documentation node. For example, the
MessageSend application can now include XMPP/SIP specific information, where
that technology specific information can be defined in chan_motif/res_xmpp/
chan_sip. Likewise, that information can also be included in the MessageSend
AMI command.
Review: https://reviewboard.asterisk.org/r/2049
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370278 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds some basic documentation for a number of modules. This
includes core source files in Asterisk (those in main), as well as
chan_agent, chan_dahdi, chan_local, sig_analog, and sig_pri. The DTD
has also been updated to allow referencing of AMI commands.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369905 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
https://origsvn.digium.com/svn/asterisk/branches/10
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r340109 | mnicholson | 2011-10-10 09:15:41 -0500 (Mon, 10 Oct 2011) | 18 lines
Merged revisions 340108 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r340108 | mnicholson | 2011-10-10 09:14:48 -0500 (Mon, 10 Oct 2011) | 11 lines
Load the proper XML documentation when multiple modules document the same application.
This patch adds an optional "module" attribute to the XML documentation spec
that allows the documentation processor to match apps with identical names from
different modules to their documentation. This patch also fixes a number of
bugs with the documentation processor and should make it a little more
efficient. Support for multiple languages has also been properly implemented.
ASTERISK-18130
Review: https://reviewboard.asterisk.org/r/1485/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340110 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r294740 | russell | 2010-11-11 16:13:38 -0600 (Thu, 11 Nov 2010) | 11 lines
Remove most of the contents of the doc dir in favor of the wiki content.
This merge does the following things:
* Removes most of the contents from the doc/ directory in favor
of the wiki - http://wiki.asterisk.org/
* Updates the build_tools/prep_tarball script to know how to export
the contents of the wiki in both PDF and plain text formats so that
the documentation is still included in Asterisk release tarballs.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294741 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r282470 | lmadsen | 2010-08-16 13:01:00 -0500 (Mon, 16 Aug 2010) | 15 lines
Merged revisions 282469 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r282469 | lmadsen | 2010-08-16 13:00:09 -0500 (Mon, 16 Aug 2010) | 7 lines
Add information about creating sounds files using
the sounds tools publically available so that others can create their
own sounds prompts using the same tools we use to generate sounds releases.
This allows people creating their own prompts to sound consistent with
the prompts available from the open source project.
SWP-595
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282471 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
Change the documented pgsql schema to use "timestamp" instead of "time",
as the latter is only a time without a date.
Added some missing columns for cel's pgsql schema, and corrected spelling
on some others. Updated cel's uniqueid size to be the same as the cdr.
Added id column to cel's pgsql schema and updated code to allow unknown
columns to get their default value instead of forcing 0 or empty string.
Added microseconds to the timestamp cel logs to pgsql.
Review: https://reviewboard.asterisk.org/r/734
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276349 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Pascal Cadotte Michaud
George Joseph
Joshua Colp
Friendly Automation
Martin Tomec
Sean Bright
Torrey Searle
Ben Ford
Florian Floimair
sungtae kim
Asterisk Development Team
Chris-Savinovich
Dan Cropp
Alexei Gradinari
Kirsty Tyerman
Antoni Goldstein
Corey Farrell
Matt Jordan
George Joseph
Richard Mudgett
Walter Doekes
Kinsey Moore
Rusty Newton
Jonathan Rose
Tzafrir Cohen
Mark Michelson
Andrew Latham
Matthew Nicholson
Jason Parker
Russell Bryant
David Ruggles
Leif Madsen
Tilghman Lesher
Olle Johansson
Tim Ringenbach
TransNexus OSP Development
Paul Belanger