Commit Graph

965 Commits

Author SHA1 Message Date
Sean Bright 312abaa1fe res_pjsip_endpoint_identifier_ip.c: Add port matching support
Adds source port matching support when IP matching is used:

  [example]
  type = identify
  match = 1.2.3.4:5060/32, 1.2.3.4:6000/32, asterisk.org:4444

If the IP matches but the source port does not, we reject and search for
alternatives. SRV lookups are still performed if enabled (srv_lookups = yes),
unless the configured FQDN includes a port number in which case just a host
lookup is performed.

ASTERISK-28639 #close
Reported by: Mitch Claborn

Change-Id: I256d5bd5d478b95f526e2f80ace31b690eebba92
2020-01-08 08:37:53 -06:00
George Joseph a4fd89536d Merge "app_bridgeaddchan.c: Make BridgeAdd be more like Bridge" 2020-01-07 14:29:27 -06:00
George Joseph 6b7334a311 Merge "app_chanisavail.c: Simplify dialplan using ChanIsAvail." 2020-01-07 14:28:55 -06:00
Friendly Automation 5b815fe1ac Merge "app_dial.c: Simplify dialplan using Dial." 2020-01-07 11:48:57 -06:00
Richard Mudgett fe3cce816c app_chanisavail.c: Simplify dialplan using ChanIsAvail.
Dialplan has to be careful about passing an empty device list or empty
positions in the list.  As a result, dialplan has to check for these
conditions before using ChanIsAvail.  Simplify dialplan by making
ChanIsAvail handle these conditions gracefully.

* Made tolerate empty positions in the device list.

* Simplified the code and eliminated some unnecessary indention.

ASTERISK-28638

Change-Id: I9e4b67e2cbf26b2417c2d03485b8568e898931d3
2020-01-06 19:11:58 -06:00
Richard Mudgett 19069f7db7 app_bridgeaddchan.c: Make BridgeAdd be more like Bridge
* Made BridgeAdd not hangup the call if there is a problem.
* Reduced message level from warning to verbose for normal exception
cases.
* Added a loop safety check to BridgeAdd.
* Made BridgeAdd set BRIDGERESULT with the status when dialplan is
resumed.

Change-Id: I374d39b8a3edcc794eeb5c6b9f31a01424cdc426
2020-01-05 21:32:01 -06:00
Richard Mudgett abcb4ab321 app_dial.c: Simplify dialplan using Dial.
Dialplan has to be careful about passing an empty destination list or
empty positions in the list.  As a result, dialplan has to check for
these conditions before using Dial.  Simplify dialplan by making Dial
handle these conditions gracefully.

* Made tolerate empty positions in the dialed device list.

* Reduced some message log levels from notice to verbose.

ASTERISK-28638

Change-Id: I6edc731aff451f8bdfaee5498078dd18c3a11ab9
2020-01-05 21:24:27 -06:00
Richard Mudgett d86a6ac5ce app_page.c: Simplify dialplan using Page.
Dialplan has to be careful about passing an empty destination list or
empty positions in the list.  As a result, dialplan has to check for
these conditions before using Page.  Simplify dialplan by making Page
handle these conditions gracefully.

* Made tolerate empty positions in the paged device list.

* Reduced some warnings associated with the 's' option to verbose
messages.  The warning level for those messages really serves no purpose
as that is why the 's' option exists.

ASTERISK-28638

Change-Id: I95b64a6d6800cd1a25279c88889314ae60fc21e3
2020-01-05 21:21:21 -06:00
Jean Aunis 034ac357ad ARI: Ability to inhibit COLP frames when adding channels to a bridge
This patch adds a new flag "inhibitConnectedLineUpdates" to the 'addChannel'
operation in the Bridges REST API. When set, this flag avoids generating COLP
frames when the specified channels enter the bridge.

ASTERISK-28629

Change-Id: Ib995d4f0c6106279aa448b34b042b68f0f2ca5dc
2020-01-02 15:06:15 +00:00
George Joseph be93537382 Merge "res_fax: wrap v21 detected Asterisk initiated negotiation with config option" 2020-01-02 08:43:21 -06:00
Joshua C. Colp 89b7144fbd confbridge: Add support for specifying maximum sample rate.
ConfBridge has the ability to move between different sample
rates for mixing the conference bridge. Up until now there has
only been the ability to set the conference bridge to mix at
a specific sample rate, or to let it move between sample rates
as necessary. This change adds the ability to configure a
conference bridge with a maximum sample rate so it can move
between sample rates but only up to the configured maximum.

ASTERISK-28658

Change-Id: Idff80896ccfb8a58a816e4ce9ac4ebde785963ee
2019-12-16 09:54:21 -06:00
Kevin Harwell b6f5607359 res_fax: wrap v21 detected Asterisk initiated negotiation with config option
A previous patch:

Gerrit Change-Id: I73bb24799bfe1a48adae9c034a2edbae54cc2a39

made it so a T.38 Gateway tries to negotiate with both sides by sending T.38
negotiation request to both endpoints supported T.38 versus the previous
behavior of forwarding negotiation to the "other" channel once a preamble
was detected.

This had the unfortunate side effect of breaking some setups. Specifically
ones that set the max datagram option on an endpoint configuration (configured
max datagram was not propagated since Asterisk now initiates negotiations).

This patch adds a configuration option, "negotiate_both", that when enabled
makes it so Asterisk initiates the negotiation requests to both endpoints vs.
the previous behavior of waiting, and forwarding the request.

The default is disabled keeping with the old behavior.

ASTERISK-28660

Change-Id: I5deb875f3485e20bc75119ec743090655d864a1a
2019-12-13 14:24:10 -06:00
Pascal Cadotte Michaud bf4dd3d837
PJSIP_CONTACT: add missing argument documentation
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
2019-12-11 12:05:27 -05:00
George Joseph 91cdb9537e Merge "Revert "PJSIP_CONTACT: add missing argument documentation"" 2019-12-11 10:36:58 -06:00
Joshua Colp d0b198b330 Revert "PJSIP_CONTACT: add missing argument documentation"
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
2019-12-11 07:01:32 -06:00
Friendly Automation 2587b7e45f Merge "PJSIP_CONTACT: add missing argument documentation" 2019-12-04 18:25:19 -06:00
George Joseph 7e3a6e158f manager.c: Prevent the Originate action from running the Originate app
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
2019-11-21 09:41:07 -06:00
Pascal Cadotte Michaud 7e3015d779
PJSIP_CONTACT: add missing argument documentation
add missing argument "rtt" and "status" to the documentation

ASTERISK-28626
Change-Id: I8419e4c8203e411b87d93dc395acdbcf7526dedf
2019-11-21 09:20:22 -05:00
Martin Tomec d257a0898e func_curl.c: Support custom http headers
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
2019-11-15 09:41:59 -05:00
Sean Bright 7362647e2f Revert "app_voicemail: Cleanup stale lock files on module load"
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
2019-10-08 06:35:05 -05:00
Torrey Searle b43cdc7f1e channel/chan_pjsip: add dialplan function for music on hold
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
2019-10-01 02:06:45 -05:00
George Joseph b0b3bd627f Merge "res_musiconhold: Add new 'playlist' mode" 2019-09-27 08:57:41 -05:00
Ben Ford 4c3655ecfd taskprocessor.c: Added "like" support to 'core show taskprocessors'
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
2019-09-25 14:01:49 -05:00
Sean Bright 966488ab52 res_musiconhold: Add new 'playlist' mode
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
2019-09-25 06:24:07 -05:00
Ben Ford 4de1e6d0e6 taskprocessor.c: Add CLI commands to reset taskprocessor stats.
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
2019-09-24 10:42:23 -05:00
Joshua Colp e79a3b428a Merge "func_jitterbuffer: Add audio/video sync support." 2019-09-19 08:23:15 -05:00
Joshua Colp 7298a785ad func_jitterbuffer: Add audio/video sync support.
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
2019-09-18 20:22:50 +00:00
Florian Floimair c18983207d core: Add H.265/HEVC passthrough support
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
2019-09-17 13:42:26 +02:00
sungtae kim cf364cd007 res_musiconhold: Added unregister realtime moh class
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
2019-09-10 21:29:48 +02:00
George Joseph 2ae1a22e0e ARI: External Media
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
2019-09-10 10:44:16 -05:00
George Joseph 19045db392 chan_rtp: Accept hostname as well as ip address as destination
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
2019-08-22 07:39:50 -05:00
Sean Bright 64906c4c9b audiohook.c: Substitute silence for unavailable audio frames
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
2019-08-20 08:44:00 -05:00
Asterisk Development Team 5e6e1175d5 Update CHANGES and UPGRADE.txt for 17.0.0 2019-07-29 11:38:30 -05:00
George Joseph 799c4cf494 Merge "chan_pjsip: Transmit REFER waits for the REFER result setting TRANSFERSTATUS" 2019-07-23 09:18:42 -05:00
George Joseph c781806e26 Build: Separate header install/uninstall
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
2019-07-16 08:17:36 -06:00
Chris-Savinovich 6b1f6ea2c4 app_voicemail.c: Build all three variants for app_voicemail at the same time
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
2019-06-28 07:32:03 -06:00
Dan Cropp e52fbae00f chan_pjsip: Transmit REFER waits for the REFER result setting TRANSFERSTATUS
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
2019-06-25 09:54:41 -05:00
Joshua Colp a8e5cf557d res_rtp_asterisk: Add support for DTLS packet fragmentation.
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
2019-06-13 07:51:57 -06:00
George Joseph 93ccff25c6 Merge "app_attended_transfer: new application AttendedTransfer" 2019-06-12 10:44:06 -05:00
Friendly Automation 9bb9700e07 Merge "app_blind_transfer: new application BlindTransfer" 2019-06-12 09:31:36 -05:00
Alexei Gradinari 3eaeb3e6c4 app_attended_transfer: new application AttendedTransfer
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
2019-06-11 08:17:06 -06:00
Alexei Gradinari 745cbab501 app_blind_transfer: new application BlindTransfer
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
2019-06-07 08:26:37 -06:00
Kirsty Tyerman bcaa01b024 pbx_dundi: added IPv4/IPv6 dual bind support to DUNDi
ASTERISK-28234
Reported-by: Kirsty Tyerman

Change-Id: I5d6e6b52dbe51415046bb3953fd16f5b421bc2e1
2019-06-05 11:56:36 -06:00
Friendly Automation 5750c97eb9 Merge "app_readexten: new option 'p' to stop reading on '#' key" 2019-06-03 10:05:11 -05:00
Ben Ford 9969c77bc2 build: Fix file format in CHANGES-staging.
One of the change files doesn't conform to the format that the release
scripts need in order to parse it.

Change-Id: Ie0b634cf27e4cbc671b9fe92993b6f2ecf60254c
2019-05-24 08:03:47 -06:00
Alexei Gradinari 408210bd4c app_readexten: new option 'p' to stop reading on '#' key
This patch adds the 'p' option.
The extension entered will be considered complete when a # is entered.

Change-Id: If77c40c9c8b525885730821e768f5dea71cf04c1
2019-05-23 08:37:08 -06:00
George Joseph be83591f99 res_rtp_asterisk: Add ability to propose local address in ICE
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
2019-05-17 17:50:06 -06:00
Joshua Colp 80dba268ea app_confbridge: Add "all" variants of REMB behavior.
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
2019-05-02 07:29:08 -06:00
Antoni Goldstein 8e21c25ce5 app_dial.c: RINGTIME, PROGRESSTIME and ms resolution dial timings
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
2019-04-24 06:27:41 -06:00
Joshua Colp 9c070a4ae3 Merge "res_pjsip: Added a norefersub configuration setting" 2019-04-19 08:30:14 -05:00
Dan Cropp cffa2a74cb res_pjsip: Added a norefersub configuration setting
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
2019-04-17 10:18:40 -05:00
Ben Ford a4ab7f5f80 build: Revise CHANGES and UPGRADE.txt handling.
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
2019-04-09 09:45:04 -05:00
Ben Ford d5d8448ce5 build: Add staging directories for future changes.
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
2019-03-27 12:32:54 -06:00
Corey Farrell 879e592baf Build System: Enable python3 compatibility.
* 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
2018-04-09 10:07:38 -04:00
Corey Farrell 0ad13949c1 doc/lang/language-criteria.txt: Link to wiki.
This document is out of date and is superseded by content on the
Asterisk wiki.

ASTERISK-24386 #close

Change-Id: Idbf95b27b096c205251e1bbb560c79224ba81822
2018-02-20 04:32:44 -06:00
Sean Bright fc98843d4b appdocsxml.xslt: Add Language to channel snapshot transformation
Change-Id: I8f494b0c895a69b8bc94656d0c6ceebecb0394d8
2018-02-01 15:03:03 -06:00
Sean Bright fd0ca1c3f9 Remove as much trailing whitespace as possible.
Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0
2017-12-22 09:23:22 -05:00
Sean Bright 9415ec2877 docs: Remove old API changes documentation
Change-Id: I1bc7957121cc7ae27dca04acc3613f4e1858476a
2017-12-20 14:55:13 -06:00
Corey Farrell 9316a064fd README: Send people to secure websites where available.
We should be sending people to secure web URL's where available.
Update README's and docs.

Change-Id: Id5b1e049b0b18b49a784f1254605aefa244ce19a
2017-11-17 21:10:55 -05:00
Matt Jordan a72ef38113 res/ari/resource_bridges: Add the ability to manipulate the video source
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
2016-11-14 17:03:09 -05:00
Corey Farrell 824a4e84d1 Refactor usage pattern of xmldoc info tag.
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
2016-08-16 10:42:46 -05:00
Matt Jordan ddab42e296 func_channel: Reorganize documentation
* 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
2016-08-15 07:39:19 -05:00
George Joseph 1d3191b118 progdocs: Exclude ./third-party from documentation generation
We don't need pjproject's documentation embedded in Asterisk's.

Change-Id: Iea6f5a621c0f4e3168dda3321eaab258d9f24a17
2016-03-19 17:51:41 -05:00
Richard Mudgett c2d48a2a28 AMI: Add Linkedid to the standard channel snapshot AMI event headers.
ASTERISK-25189 #close
Reported by: John Hardin

Change-Id: I2b1778c3fdc1dca0ed55db4e3a639eddfb16c2ac
2015-06-26 10:34:31 -05:00
Corey Farrell 57386dcb67 Allow command-line options to override asterisk.conf.
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
2015-05-12 12:44:12 -04:00
George Joseph cf637f2510 doc: Make progdocs play nice with git
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
2015-05-07 22:26:31 -06:00
Corey Farrell c232ff3af0 Git Migration: Create doc/rest-api when needed.
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
2015-04-29 06:24:51 -05:00
George Joseph b35e184d41 Add .gitignore and .gitreview files
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
2015-04-11 19:43:43 -06:00
Walter Doekes b8c1130ed1 docs: Escape unescaped minus sign in asterisk.8 manpage.
ASTERISK-23768 #close
Reported by: Jeremy Lainé
Patches:
  escape_manpage_hyphen.patch uploaded by Jeremy Lainé (License #6561)
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2014-09-26 08:26:24 +00:00
Matthew Jordan 664f83a03b doc: Add a manpage for the smsq utility
This patch adds a manpage for the smsq utility. Note that this is one of
the patches the Debian distro applies for the Asterisk project, as per
ASTERISK-24191.

Review: https://reviewboard.asterisk.org/r/3895/

ASTERISK-24171 #close
Reported by: Jeremy Laine
patches:
  smsq.8 uploaded by Jeremy Laine (License 6561)
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2014-08-29 19:40:34 +00:00
Matthew Jordan 81598fa082 doc: Add a manpage for the aelparse utility
This patch adds a manpage for the aelparse utility. Note that this is one of
the patches the Debian distro applies for the Asterisk project, as per
ASTERISK-24191.

Review: https://reviewboard.asterisk.org/r/3896/

ASTERISK-24171 #close
Reported by: Jeremy Laine
patches:
  aelparse.8 uploaded by Jeremy Laine (License 6561)
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2014-08-29 19:35:43 +00:00
Matthew Jordan 922e3203a9 xmldocs: Add support for an <example> tag in the Asterisk XML Documentation
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
2014-07-31 11:49:40 +00:00
Kinsey Moore 4445ee7fc0 AMI: Allow for command response documentation
Allow for responses to AMI actions/commands to be documented properly
in XML and displayed via the CLI. Response events are documented
exactly as standard AMI events are documented.

Review: https://reviewboard.asterisk.org/r/3812/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419342 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-24 13:00:59 +00:00
Rusty Newton f6647d2362 Documentation: doc fixes across various parts of the code for ASTERISK issues 23061,23028,23046,23027
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)
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2014-01-17 17:16:14 +00:00
Jonathan Rose b0bb03e916 bridging: Give bridges a name and a known creator
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404043 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-17 23:25:49 +00:00
Matthew Jordan f46b30bd36 func_pjsip_endpoint: Add PJSIP_ENDPOINT function for querying endpoint details
This patch adds a new function, PJSIP_ENDPOINT, which lets the dialplan query,
for any endpoint, any property configured on an endpoint. This function is a
companion to the CHANNEL function, which can be used to extract the endpoint
name for a channel.

Review: https://reviewboard.asterisk.org/r/3035
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2013-12-11 12:31:57 +00:00
Tzafrir Cohen e2204a1e61 man pages for astdb2bdb and astdb2sqlite3
Review: https://reviewboard.asterisk.org/r/2898/
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2013-10-02 18:28:03 +00:00
Kinsey Moore 03090a88ba Fix documentation replication issues
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
2013-08-01 17:07:52 +00:00
Mark Michelson 4710869133 Add documentation for features configuration.
Review: https://reviewboard.asterisk.org/r/2616

(closes issue ASTERISK-21542)
Reported by Matt Jordan



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392729 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-24 18:33:13 +00:00
Matthew Jordan d04ab3c645 Add CLI configuration documentation
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
2013-02-15 13:38:12 +00:00
Matthew Jordan 472e29df62 Let documentation reference links specify which module they're linking to
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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2013-01-16 17:46:15 +00:00
Matthew Jordan b84d37a711 Multiple revisions 379209-379210
........
  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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2013-01-16 15:33:05 +00:00
Andrew Latham fd98835f1f Doxygen Updates Janitor Work
* 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
2012-09-22 20:43:30 +00:00
Matthew Jordan 4781be6e93 Restore CODING-GUIDELINES to doc folder
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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2012-08-30 12:50:03 +00:00
Matthew Jordan 86ff5585fd Add the ability to specify technology specific documentation
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
2012-07-19 22:17:13 +00:00
Matthew Jordan 2ffae5745d Add some additional documentation for core AMI events
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
2012-07-10 22:26:27 +00:00
Matthew Jordan 82a7409c15 Add AMI event documentation
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
2012-06-25 17:59:34 +00:00
Matthew Nicholson bb07ca66a1 Merged revisions 340109 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  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
  
  ........
    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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2011-10-10 14:16:27 +00:00
Jason Parker 5fe104ddb9 Merged revisions 331143 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r331143 | qwell | 2011-08-09 10:59:54 -0500 (Tue, 09 Aug 2011) | 9 lines
  
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  ........
    r331142 | qwell | 2011-08-09 10:58:16 -0500 (Tue, 09 Aug 2011) | 1 line
    
    Regenerate asterisk man page from sgml.
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331144 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-08-09 16:02:45 +00:00
Jason Parker 873962f772 Merged revisions 331139 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  r331139 | qwell | 2011-08-09 10:50:07 -0500 (Tue, 09 Aug 2011) | 19 lines
  
  Merged revisions 306999 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r306999 | lathama | 2011-02-08 14:22:35 -0600 (Tue, 08 Feb 2011) | 12 lines
    
    Documentation Updates
    
    Note default polling setting in voicemail.conf
    Add missing config to asterisk.conf
    Update manpage
    
    (issue #16505)
    Reported by: tzafrir
    Patches: 
          asterisk_sgml_fixes_demo.diff uploaded by tzafrir (license 46)
    Tested by: lathama, tzafrir
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331141 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-08-09 15:53:26 +00:00
Jason Parker 19c8278815 Merged revisions 331138 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

........
  r331138 | qwell | 2011-08-09 10:47:20 -0500 (Tue, 09 Aug 2011) | 1 line
  
  Revert merge of r306999, due to merge conflict.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331140 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-08-09 15:51:38 +00:00
Andrew Latham a350924700 Documentation Updates
Note default polling setting in voicemail.conf
Add missing config to asterisk.conf
Update manpage

(issue #16505)
Reported by: tzafrir
Patches:
     asterisk_sgml_fixes_demo.diff uploaded by tzafrir (license 46)
Tested by: lathama, tzafrir



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307041 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-08 20:31:13 +00:00
Andrew Latham c4271440ab Documentation Updates.
Start updates to the man pages.

(issue #16505)
Reported by: tzafrir
Tested by: lathama


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306827 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-08 02:05:03 +00:00
Russell Bryant b1aa9069fe Merged revisions 294745 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r294745 | russell | 2010-11-11 16:17:57 -0600 (Thu, 11 Nov 2010) | 6 lines
  
  Remove CCSS architecture PDF.
  
  It has been moved to:
  
  https://wiki.asterisk.org/wiki/display/AST/CCSS+Architecture
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294749 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-11 22:18:33 +00:00
Russell Bryant 893ca656af Merged revisions 294740 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  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.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294741 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-11 22:14:25 +00:00
Russell Bryant 6b1019418a Merged revisions 291725 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r291725 | russell | 2010-10-14 07:08:43 -0500 (Thu, 14 Oct 2010) | 2 lines
  
  Fix a typo - s/seucre/secure/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@291726 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-10-14 12:10:29 +00:00
Richard Mudgett e100d8f649 Merged revisions 288082 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r288082 | rmudgett | 2010-09-21 16:03:28 -0500 (Tue, 21 Sep 2010) | 1 line
  
  Add note in party manipulation chapter on interception macros.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@288083 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-21 21:04:04 +00:00
Richard Mudgett 5fa454010e Merged revisions 286647 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r286647 | rmudgett | 2010-09-14 10:30:49 -0500 (Tue, 14 Sep 2010) | 1 line
  
  Corrected documented CONNECTED_LINE and REDIRECTING party manipulation macro names.
........


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2010-09-14 15:31:51 +00:00
David Ruggles c142342f12 Merged revisions 285992 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r285992 | diruggles | 2010-09-10 09:13:16 -0400 (Fri, 10 Sep 2010) | 1 line
  
  Added missing documentation for ExternalIVR feature added in January 2010
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@285993 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-10 13:20:16 +00:00
Richard Mudgett df6d017aa1 Merged revisions 284698 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r284698 | rmudgett | 2010-09-02 11:34:32 -0500 (Thu, 02 Sep 2010) | 5 lines

  Added documentation for CONNECTEDLINE and REDIRECTING functions.

  (closes issue #17808)
  Reported by: jtodd

  Review: https://reviewboard.asterisk.org/r/875/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284699 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-02 16:35:39 +00:00
Leif Madsen fc55d25510 Merged revisions 282470 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  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
  
  ........
    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
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282471 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-16 18:02:29 +00:00
Tilghman Lesher ee6592684b Merged revisions 280740 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r280740 | tilghman | 2010-08-03 13:42:24 -0500 (Tue, 03 Aug 2010) | 9 lines
  
  Merged revisions 280739 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r280739 | tilghman | 2010-08-03 13:39:28 -0500 (Tue, 03 Aug 2010) | 2 lines
    
    Document -B and -W flags and regenerate manpage from sgml
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280741 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-03 18:43:37 +00:00
Tzafrir Cohen 76c1992f38 Some left-over hyphen-minus fixes in the man page
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278947 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-23 16:07:53 +00:00
Olle Johansson e129b31fc6 Add ability to configure the Max-Forwards header in the dialplan, as well as in
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
2010-07-16 10:00:58 +00:00
Tim Ringenbach e19a6c248f Fix documentation for pgsql cel and cdr, and slightly improve pgsql_cel.
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
2010-07-14 16:09:11 +00:00
TransNexus OSP Development d6a70b619c Changed OSP TCP port from 1080 to 5045.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274492 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-07 07:07:08 +00:00
Russell Bryant 910cd8be1d Use the underscore package so that underscores do not need to be escaped.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272684 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-28 15:33:32 +00:00
Tilghman Lesher 7aa382b53f Merged revisions 272562 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r272562 | tilghman | 2010-06-25 15:17:37 -0500 (Fri, 25 Jun 2010) | 5 lines
  
  Make the structure of the table specified before match the queries and results.
  
  (closes issue #17557)
   Reported by: cmaj
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272568 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-25 20:18:47 +00:00
Paul Belanger 5d2bbe86ad Update formatting for channelvariables.tex
(closes issue #17511)
Reported by: klaus3000
Patches:
      channelvariables.tex-patch.txt uploaded by klaus3000 (license 65)
Tested by: pabelanger



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270801 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-16 15:05:11 +00:00
Tilghman Lesher 81c15adfa2 Add distributed devicestate via the XMPP protocol.
(closes issue #15757)
 Reported by: Marquis
 Patches: 
       distributed_devstate-XMPP.txt uploaded by lmadsen (license 10)
 Tested by: Marquis, lmadsen, marcelloceschia
 
Review: https://reviewboard.asterisk.org/r/351/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270519 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-15 17:06:23 +00:00
Paul Belanger f9bcb15db3 Merged revisions 270078 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r270078 | pabelanger | 2010-06-12 14:54:20 -0400 (Sat, 12 Jun 2010) | 2 lines
  
  Fix typo in example
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270079 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-12 18:55:47 +00:00
Mark Michelson 326c783685 Add documentation explaining PLC in Asterisk.
Review: https://reviewboard.asterisk.org/r/688/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269749 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-10 17:14:38 +00:00
Terry Wilson 857814f435 Add SRTP support for Asterisk
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
2010-06-08 05:29:08 +00:00
David Vossel 3280a5c0af Update CHANGES and aoc help doc to reflect AOC additions
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267181 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-02 19:33:56 +00:00
Richard Mudgett afd4454c44 Generic Advice of Charge.
Asterisk Generic AOC Representation
- Generic AOC encode/decode routines.
  (Generic AOC must be encoded to be passed on the wire in the AST_CONTROL_AOC frame)
- AST_CONTROL_AOC frame type to represent generic encoded AOC data
- Manager events for AOC-S, AOC-D, and AOC-E messages

Asterisk App Support
- app_dial AOC-S pass-through support on call setup
- app_queue AOC-S pass-through support on call setup

AOC Unit Tests
- AOC Unit Tests for encode/decode routines
- AOC Unit Test for manager event representation.

SIP AOC Support
- Pass-through of generic AOC-D and AOC-E messages to snom phones via the
  snom AOC specification.
- Creation of chan_sip page3 flags for the addition of the new
  'snom_aoc_enabled' sip.conf option.

IAX AOC Support
- Natively supports AOC pass-through through the use of the new
  AST_CONTROL_AOC frame type

DAHDI AOC Support
- ETSI PRI full AOC Pass-through support
- 'aoc_enable' chan_dahdi.conf option for independently enabling
  pass-through of AOC-S, AOC-D, AOC-E.
- 'aoce_delayhangup' option for retrieving AOC-E on disconnect.
- DAHDI A() dial string option for requesting AOC services.
  example usage:
  ;requests AOC-S, AOC-D, and AOC-E on call setup
  exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e))

Review:	https://reviewboard.asterisk.org/r/552/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267096 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-02 18:10:15 +00:00
Terry Wilson 0390dae08d Merge the rest of the FullyBooted patch
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265467 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-24 22:21:58 +00:00
Terry Wilson 3c9a8ebadb Add the FullyBooted AMI event
It is possible to connect to the manager interface before all Asterisk modules
are loaded. To ensure that an application does not send AMI actions that might
require a module that has not yet loaded, the application can listen for the
FullyBooted manager event. It will be sent upon connection if all modules have
been loaded, or as soon as loading is complete. The event:

   Event: FullyBooted
   Privilege: system,all
   Status: Fully Booted

Review: https://reviewboard.asterisk.org/r/639/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265320 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-24 19:06:40 +00:00
Leif Madsen 3a1cabb13f Merged revisions 260569 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r260569 | lmadsen | 2010-05-03 09:57:39 -0500 (Mon, 03 May 2010) | 1 line
  
  Minor typo pointed out by pabelanger on IRC.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260570 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-03 14:58:23 +00:00
Eliel C. Sardanons a753e8878b Asterisk data retrieval API.
This module implements an abstraction for retrieving and exporting
asterisk data.
Developed by:
	Brett Bryant <brettbryant@gmail.com>
	Eliel C. Sardanons (LU1ALY) <eliels@gmail.com>
For the Google Summer of code 2009 Project.
Documentation can be found in doxygen format and inside the
header include/asterisk/data.h

Review: https://reviewboard.asterisk.org/r/275/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258517 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-22 18:07:02 +00:00
Russell Bryant 8096f0fecc Add MEETMEBOOKID from r256019.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258515 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-22 17:36:34 +00:00
Leif Madsen 5258e5e683 Missed this when reverting the bad version change in asterisk.tex.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258387 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-21 19:45:33 +00:00
Leif Madsen 8ea4ecd58a Fix change in asterisk.tex that got merged in after testing.
(issue #17220)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258383 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-21 19:27:41 +00:00
Leif Madsen 8b11ae2e4f Add ability to generate ASCII documentation from the TeX files.
These changes add the ability to run 'make asterisk.txt' just like the existing
'make asterisk.pdf' commands to generate a text document from the TeX files we
have in the doc/tex/ directory. I've also updated a few of the .tex files because
they weren't properly escaping certain characters so they would show up as Unicode
characters (like [U+021C]). Made changes to the configure scripts so it would
detect the catdvi program which is required to convert the .dvi file generated
by latex.

I've also added a few lines to the build_tools/prep_tarball script so that the
text documentation gets generated and added to future tarballs of Asterisk
releases.

(closes issue #17220)
Reported by: lmadsen
Patches: 
      asterisk.txt.patch uploaded by lmadsen (license 10)
      asterisk.txt.patch-v4 uploaded by pabelanger (license 224)
Tested by: lmadsen, pabelanger

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258351 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-21 19:18:35 +00:00
Julian Lyndon-Smith 4b7c56ccef fix whitespace issue
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258256 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-21 13:24:28 +00:00
Julian Lyndon-Smith 81fd235286 Added NEW ACTIONS entry for new MixMonitorMute AMI command.
Added State and Direction variables for new MixMonitorMute AMI command.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258228 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-21 13:08:44 +00:00
Leif Madsen bef07ecc16 Merged revisions 257426 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r257426 | lmadsen | 2010-04-15 14:40:33 -0500 (Thu, 15 Apr 2010) | 13 lines
  
  Update backtrace.txt documentation.
  
  Update the backtrace.txt documentation so it conforms to the same layout as
  other documents we've been working on recently. Additionally, add a bunch of
  new information about gathering backtraces for crashes and deadlocks, along
  with ways of verifying your file before uploading it. Create a couple of one
  line commands for people to generate the files we need.
  
  (closes issue #17190)
  Reported by: lmadsen
  Patches: 
        backtrace.txt.patch-2 uploaded by lmadsen (license 10)
  Tested by: lmadsen, pabelanger
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257427 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-15 19:41:05 +00:00
Leif Madsen 28b78c5e67 Merged revisions 257342 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r257342 | lmadsen | 2010-04-15 08:41:45 -0500 (Thu, 15 Apr 2010) | 1 line
  
  Update address of the bug tracker.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257343 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-15 13:44:38 +00:00
Leif Madsen 47d8fdae97 Merged revisions 256900 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r256900 | lmadsen | 2010-04-12 12:29:26 -0500 (Mon, 12 Apr 2010) | 15 lines
  
  Add How-To document on collecting debugging info for issues.asterisk.org
  
  Paul Belanger has been helping a lot with bug tracking recently and created
  this document that we can now point to when additional debugging information
  is required. This document will help those filing issues to know how to get
  the information required when filing their issues. This will make things
  easier on the developers.
  
  Initial text and changes by pabelanger. Tweaks and editing by myself.
  
  (closes issue #17159)
  Reported by: pabelanger
  Patches: 
        HOWTO_collect_debug_information.txt.patch uploaded by lmadsen (license 10)
  Tested by: tzafrir, pabelanger, lmadsen
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256901 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-12 17:29:53 +00:00
Tzafrir Cohen 16774318f9 fix hyphen vs. minus in man pages
In troff '-' is used for a hyphen. A minus is denoted by '\-' . This is
normally also used for a dash.

This patch converts all '-'-s that are minuses or dashes to '\-'.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256704 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-10 08:33:57 +00:00
Richard Mudgett d66b44b4ca Merge CCSS architecture document from CCSS branch.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256608 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-09 19:46:54 +00:00
Mark Michelson e24661fd18 Merge Call completion support into trunk.
From Reviewboard:
CCSS stands for Call Completion Supplementary Services. An admittedly out-of-date
overview of the architecture can be found in the file doc/CCSS_architecture.pdf
in the CCSS branch. Off the top of my head, the big differences between what is
implemented and what is in the document are as follows:

1. We did not end up modifying the Hangup application at all.
2. The document states that a single call completion monitor may be used across
   multiple calls to the same device. This proved to not be such a good idea
   when implementing protocol-specific monitors, and so we ended up using one
   monitor per-device per-call.
3. There are some configuration options which were conceived after the document
   was written. These are documented in the ccss.conf.sample that is on this
   review request.
		      
For some basic understanding of terminology used throughout this code, see the
ccss.tex document that is on this review.

This implements CCBS and CCNR in several flavors.

First up is a "generic" implementation, which can work over any channel technology
provided that the channel technology can accurately report device state. Call
completion is requested using the dialplan application CallCompletionRequest and can
be canceled using CallCompletionCancel. Device state subscriptions are used in order
to monitor the state of called parties.

Next, there is a SIP-specific implementation of call completion. This method uses the
methods outlined in draft-ietf-bliss-call-completion-06 to implement call completion
using SIP signaling. There are a few things to note here:

* The agent/monitor terminology used throughout Asterisk sometimes is the reverse of
  what is defined in the referenced draft.

* Implementation of the draft required support for SIP PUBLISH. I attempted to write
  this in a generic-enough fashion such that if someone were to want to write PUBLISH
  support for other event packages, such as dialog-state or presence, most of the effort
  would be in writing callbacks specific to the event package.

* A subportion of supporting PUBLISH reception was that we had to implement a PIDF
  parser. The PIDF support added is a bit minimal. I first wrote a validation
  routine to ensure that the PIDF document is formatted properly. The rest of the
  PIDF reading is done in-line in the call-completion-specific PUBLISH-handling
  code. In other words, while there is PIDF support here, it is not in any state
  where it could easily be applied to other event packages as is.

Finally, there are a variety of ISDN-related call completion protocols supported. These
were written by Richard Mudgett, and as such I can't really say much about their
implementation. There are notes in the CHANGES file that indicate the ISDN protocols
over which call completion is supported.

Review: https://reviewboard.asterisk.org/r/523


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-09 15:31:32 +00:00
Leif Madsen 3a4baef6d9 Fix for localchannel.tex to allow PDFs to be generated again.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256161 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-05 15:14:53 +00:00
Leif Madsen b6e783d97c Update to new Local channel documentation.
Add same changes as commit to 1.4, but convert to TeX.

(issue #16963)
Reported by: kobaz
Patches: 
      localchannel-2.txt uploaded by kobaz (license 834)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253256 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-18 15:46:52 +00:00
Leif Madsen 06ff72a78e Update existing Local channel documentation.
A complete re-write of the Local channel documentation has been performed, with
the existing information from localchannel.txt and localchannel.tex merged in.

(closes issue #16637)
Reported by: kobaz
Patches: 
      localchannel.tex uploaded by lmadsen (license 10)
      localchannel.txt uploaded by lmadsen (license 10)
Tested by: lmadsen, jsmith, mmichelson

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250609 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-03 21:22:55 +00:00
Leif Madsen 491ea82b6d Update IMAP documentation.
Update the IMAP documentation to make it clear that storing voicemails
in the same folder as a large number of emails could potentially cause
significant slow downs when writing or retrieving voicemails.

(issue #16704)
Reported by: TimeHider
Tested by: lmadsen, TimeHider

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250051 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-02 21:09:27 +00:00
Leif Madsen aec5c6f840 Update documentation to not imply we support overriding options.
(closes issue #16855)
Reported by: davidw

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250037 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-02 20:36:10 +00:00
Tilghman Lesher 873989db91 Enable SendText to send strings in encoded format.
See http://lists.digium.com/pipermail/asterisk-users/2010-January/243462.html


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241364 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-19 22:41:36 +00:00
David Ruggles cfb0ba32ee Updated ExternalIVR documentation
Rewrote a large portion of the existing documentation
and added information about the TCP/IP socket interface


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240973 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-18 17:51:09 +00:00
Leif Madsen 712b500ee2 Add documentation about how to build queues.
Add a how-to set of documentation about building queues with Asterisk.
This documentation is based on Asterisk 1.6.2 but should work on most
versions with minor modifications.

(closes issue #16237)
Reported by: lmadsen
Patches:
      Building Queues (FINAL).txt uploaded by lmadsen (license 10)
Tested by: pdhales, lmadsen, cmdrwalrus

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240039 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-14 14:38:01 +00:00
TransNexus OSP Development ce80ff55c8 Updated channel variable list of osplookup application.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239625 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-13 07:02:13 +00:00
Jason Parker f93071483f Add app_voicemail and say.c support for Vietnamese.
Also add an XXX comment that I'm baffled nobody has ever complained about.  We
say "first message", and then we go into language-specific stuff where we
proceed to say..."first message".

(closes issue #15053)
Reported by: dinhtrung
Patches:
      vietnamese.ods uploaded by dinhtrung (license 776)
      app_voicemail.c.diff uploaded by dinhtrung (license 776)

(closes issue #15626)
Reported by: dinhtrung
Patches:
      say.c.diff uploaded by dinhtrung (license 776)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237050 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-30 22:30:21 +00:00
Leif Madsen 7ab15342ac Update IMAP build documentation.
Update the IMAP build documentation to show how to build on 64-bit
platforms.


(issue #16433)
Reported by: shrift
Tested by: lmadsen

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234631 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-14 17:19:58 +00:00
Joshua Colp d0e431ce3d Add an 'X' option to the asterisk application which enables #exec for configuration files.
This option can be used to enable #exec support in the asterisk.conf configuration file.

(closes issue #16260)
Reported by: atis
Patches:
      exec_includes.patch uploaded by atis (license 242)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232510 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-02 20:10:07 +00:00
Tilghman Lesher 23b4a700f4 Reorder option flags. Change guidelines so that example code is consistent with guidelines
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231369 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-26 02:09:58 +00:00
David Ruggles 6e27759d33 Fix/Implement error events for non-existing files
also include a better cmd define for S command

Review: https://reviewboard.asterisk.org/r/430/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230584 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-20 17:28:01 +00:00
David Ruggles 91b9c8fca1 Remove non-functional feature from ExternalIVR documentation
Remove non-functional socket implementation of ExternalIVR from documentation

(closes issue #16225)
Reported by: thedavidfactor
Patches:
      externalivr.txt.20091111.1542.patch uploaded by thedavidfactor (license 903)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229568 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-11 20:47:06 +00:00
David Ruggles baee8b9a54 Merged revisions 229355 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r229355 | diruggles | 2009-11-10 16:45:15 -0500 (Tue, 10 Nov 2009) | 9 lines
  
  Fix ExternalIVR Documentation
  
  Remove documentation for event that doesn't function
  
  (closes issue #16220)
  Reported by: thedavidfactor
  Patches:
        externalivr.txt.20091110.1622.patch uploaded by thedavidfactor (license 903)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229356 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-10 22:01:50 +00:00
David Ruggles 00b269af48 Merged revisions 229191 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r229191 | diruggles | 2009-11-10 12:23:59 -0500 (Tue, 10 Nov 2009) | 11 lines
  
  Document ExternalIVR event tag collision
  
  ExternalIVR uses the D tag for two different event types. This documents that
  behavior and how to differentiate between the two cases. Also includes a minor
  spelling fix and clarification
  
  (closes issue #16211)
  Reported by: thedavidfactor
  Patches:
        externalivr.txt.20091109.1507.patch uploaded by thedavidfactor (license 903)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229228 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-10 17:33:47 +00:00
Joshua Colp 7c4fd85293 Fix the localchannel.tex file.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228499 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-06 17:52:00 +00:00
Tilghman Lesher d8e0c58437 Expand codec bitfield from 32 bits to 64 bits.
Reviewboard: https://reviewboard.asterisk.org/r/416/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-04 14:05:12 +00:00
Joshua Colp b9c370da86 Merged revisions 226531 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r226531 | file | 2009-10-29 15:11:26 -0300 (Thu, 29 Oct 2009) | 6 lines
  
  Add an option to enabling passing music on hold start and stop requests through instead of
  acting on them in chan_local.
  
  (closes issue #14709)
  Reported by: dimas
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226532 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-29 18:13:42 +00:00
Leif Madsen 6b6fee6c95 Merged revisions 226377 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r226377 | lmadsen | 2009-10-28 14:48:29 -0500 (Wed, 28 Oct 2009) | 7 lines
  
  Update CALLINGSUBADDR channel variable documentation.
  
  (closes issue #15734)
  Reported by: alecdavis
  Patches:
        channelvariables.tex.diff.txt uploaded by alecdavis (license 585)
  Tested by: alecdavis
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-28 19:50:00 +00:00
Leif Madsen b544c53778 Merged revisions 225484 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r225484 | lmadsen | 2009-10-22 16:51:52 -0500 (Thu, 22 Oct 2009) | 11 lines
  
  Clean valgrind output by suppressing false errors.
  Update valgrind.txt documentation and add valgrind.supp file in order to
  allow those who are creating valgrind output to have less false errors in
  the logfile.
  
  (closes issue #16007)
  Reported by: atis
  Patches:
        valgrind.txt.diff uploaded by atis (license 242)
        asterisk2.supp uploaded by atis (license 242)
  Tested by: atis, amorsen
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225485 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-22 21:52:30 +00:00
Sean Bright 117d3fd910 Add the programs in utils/ to menuselect.
Nothing in utils/ is now built by default except for astcanary.

Review: https://reviewboard.asterisk.org/r/353/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225440 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-22 19:33:32 +00:00
Kevin P. Fleming 87ff40d3f3 Add 'mohsuggest' configuration option to 'sip show peer' CLI command and
SIPShowPeer AMI action.

(closes issue #15990)
Reported by: _brent_
Patches:
      sip_peer_info_mohsuggest-r3.patch uploaded by brent (license 388)

Review: https://reviewboard.asterisk.org/r/381/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225245 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-21 21:15:40 +00:00
Tilghman Lesher 2b82d051ca Remove a completed project and add another
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224527 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-19 15:50:31 +00:00
Philippe Sultan b11b94a083 Add JABBER_RECEIVE as a dialplan function, implement SendText in Jingle channels
JABBER_RECEIVE (along with JabberSend) makes Asterisk interact with users over
XMPP to process calls.
SendText can be used instead of JabberSend in the context of XMPP based voice
channels (chan_gtalk and chan_jingle).

(closes issue #12569)
Reported by: eech55
Tested by: phsultan, asannucci, lmadsen, jtodd, maxgo

Review: https://reviewboard.asterisk.org/r/88/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220457 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-25 10:54:42 +00:00
Tilghman Lesher 1a50d01ed8 Update fax number to the legal fax, not the generic fax.
(closes issue #15946)
 Reported by: jtodd
 Patches: 
       leif-is-a-wuss.txt uploaded by jtodd (license 870)
 Tested by: jparker, tilghman, jtodd, russellb, mmichelson, seanbright, kpfleming, and the rest of the usual suspects


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-23 23:19:14 +00:00
Olle Johansson 0cebb5047b Fixing formatting
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216956 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-07 20:23:19 +00:00
Olle Johansson fd736b1b8f Add new actions under "new actions" and not in the top of the document
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216955 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-07 20:19:37 +00:00
Olle Johansson 773e22596a Adding to the janitor list.
For new readers: The janitor list is a list of tasks we need help with in the Asterisk project. Taking up 
one of these is often a good way to get into Asterisk development and getting a lot of developers in 
the project to be grateful. It's stuff we could spend time on when the bug tracker is empty, when our
employers hasn't filled our task lists and our servers is running bugfree and happily without any issues.

If you want to start working on one of these small projects, feel free to ask for help in the #asterisk-dev
channel on IRC or asterisk-dev mailing list. We'll be more than happy to help you to start and reach
goal.

Thank you for your help.

</end of long commit message>


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216335 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-04 12:05:46 +00:00
Russell Bryant 3bad695fed Merged revisions 216263 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

................
  r216263 | russell | 2009-09-04 05:48:00 -0500 (Fri, 04 Sep 2009) | 9 lines
  
  Merged revisions 216262 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.2
  
  ........
    r216262 | russell | 2009-09-04 05:47:37 -0500 (Fri, 04 Sep 2009) | 2 lines
    
    Add a plain text version of the IAX2 security document.
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216264 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-04 10:48:44 +00:00
Russell Bryant 5fdc6292a3 Merged revisions 216008 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

................
  r216008 | russell | 2009-09-03 13:44:58 -0500 (Thu, 03 Sep 2009) | 9 lines
  
  Merged revisions 216005 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.2
  
  ........
    r216005 | russell | 2009-09-03 13:42:24 -0500 (Thu, 03 Sep 2009) | 2 lines
    
    Add IAX2 security document related to AST-2009-006.
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216009 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-03 18:45:54 +00:00
Kevin P. Fleming 7f745ecd73 Document language prompt submission process.
This patch adds a document describing the language prompt submission process,
licensing terms and other issues related to that process. In addition, it
modifies the sound file searching process to support language codes with
any number of suffices (not limited to just "xx" or "xx_YY"), so that prompts
can be named with gender, customer/company, etc. suffices as well.

(closes issue #15771)
Reported by: jtodd
Patches:
      language-criteria.txt uploaded by jtodd



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216006 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-03 18:42:38 +00:00
Michiel van Baak b3e97dc95c Document that SIPshowpeer and SKINNYshowline now include
the configured parkinglot in their response.

Prodded by snuff-work on #asterisk-dev IRC channel


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215838 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-03 05:57:23 +00:00
Jeff Peeler 08df1b85bf Add forgotten documentation for new channel variables added in 214309.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214355 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-27 15:57:47 +00:00
Kevin P. Fleming 0e70c71c25 Convert this branch to Opsound music-on-hold.
For more details:
http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@212922 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-18 20:29:37 +00:00
Tilghman Lesher 4344e26b09 Merged revisions 211583 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r211583 | tilghman | 2009-08-10 14:48:48 -0500 (Mon, 10 Aug 2009) | 1 line
  
  Conversion specifiers, not format specifiers
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@211584 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-10 19:49:41 +00:00
Tilghman Lesher 642bec4d6f AST-2009-005
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@211539 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-10 19:20:57 +00:00
Leif Madsen b8f1c9c4c3 Merged revisions 210563 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r210563 | lmadsen | 2009-08-05 13:46:21 -0500 (Wed, 05 Aug 2009) | 11 lines
  
  Update imapstorage.txt documentation.
  Updated the imapstorage.txt documentation to reflect that issues with
  c-client versions older than 2007 seem to cause crashing issues that
  are not seen with more recent versions. Documentation has been updated
  to reflect this.
  
  (closes issue #14496)
  Reported by: vbcrlfuser
  Patches:
        __20090727-imap-documentation-patch.txt uploaded by lmadsen (license 10)
  Tested by: lmadsen, mmichelson, dbrooks
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@210564 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-05 18:49:58 +00:00
Kevin P. Fleming e9d22f802e Rename 'canreinvite' option to 'directmedia', with backwards compatibility.
It is clear from multiple mailing list, forum, wiki and other sorts of posts
that users don't really understand the effects that the 'canreinvite' config
option actually has, and that in some cases they think that setting it to 'no'
will actually cause various other features (T.38, MOH, etc.) to not work properly,
when in fact this is not the case. This patch changes the proper name of the
option to what it should have been from the beginning ('directmedia'), but
preserves backwards compatibility for existing configurations.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@210190 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-03 20:48:48 +00:00
Bradley Latus 9eb46f3286 Update documentation in relation to UnixODBC
(closes issue #15516)
Reported by: snuffy
Patches: 
      bug_odbc_tex_update_v2.diff uploaded by snuffy (license 35)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209959 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-01 23:33:31 +00:00
Bradley Latus 0a47411334 (closes issue #15515)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209958 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-01 23:25:16 +00:00
Russell Bryant ced2554f60 Note that we use tabs instead of spaces for indentation.
I'm surprised this was never actually in here...


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207925 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-21 22:22:18 +00:00
Kevin P. Fleming 96e4e31eeb Merged revisions 207647 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r207647 | kpfleming | 2009-07-21 08:04:44 -0500 (Tue, 21 Jul 2009) | 12 lines
  
  Ensure that user-provided CFLAGS and LDFLAGS are honored.
  
  This commit changes the build system so that user-provided flags (in ASTCFLAGS
  and ASTLDFLAGS) are supplied to the compiler/linker *after* all flags provided
  by the build system itself, so that the user can effectively override the
  build system's flags if desired. In addition, ASTCFLAGS and ASTLDFLAGS can now
  be provided *either* in the environment before running 'make', or as variable
  assignments on the 'make' command line. As a result, the use of COPTS and LDOPTS
  is no longer necessary, so they are no longer documented, but are still supported
  so as not to break existing build systems that supply them when building Asterisk.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207680 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-21 13:28:04 +00:00
Tilghman Lesher 165ff1314b Add flag here, too (as requested by jsmith)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207255 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-17 22:29:50 +00:00
Tilghman Lesher aa379bb741 Document the "flag" field in the voicemessages table.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207224 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-17 22:04:43 +00:00
Russell Bryant 4cf8a968fd Add an API for reporting security events, and a security event logging module.
This commit introduces the security events API.  This API is to be used by
Asterisk components to report events that have security implications.
A simple example is when a connection is made but fails authentication.  These
events can be used by external tools manipulate firewall rules or something
similar after detecting unusual activity based on security events.

Inside of Asterisk, the events go through the ast_event API.  This means that
they have a binary encoding, and it is easy to write code to subscribe to these
events and do something with them.

One module is provided that is a subscriber to these events - res_security_log.
This module turns security events into a parseable text format and sends them
to the "security" logger level.  Using logger.conf, these log entries may be
sent to a file, or to syslog.

One service, AMI, has been fully updated for reporting security events.
AMI was chosen as it was a fairly straight forward service to convert.
The next target will be chan_sip.  That will be more complicated and will
be done as its own project as the next phase of security events work.

For more information on the security events framework, see the documentation
generated from doc/tex/.  "make asterisk.pdf"

Review: https://reviewboard.asterisk.org/r/273/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206021 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-11 19:15:03 +00:00
Russell Bryant c511a26749 Move Asterisk-addons modules into the main Asterisk source tree.
Someone asked yesterday, "is there a good reason why we can't just put these
modules in Asterisk?".  After a brief discussion, as long as the modules are
clearly set aside in their own directory and not enabled by default, it is
perfectly fine.

For more information about why a module goes in addons, see README-addons.txt.

chan_ooh323 does not currently compile as it is behind some trunk API updates.
However, it will not build by default, so it should be okay for now.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-30 16:40:38 +00:00
Mark Michelson e403a7fdfb Add timestamp to response to "Ping" manager action.
(closes issue #14596)
Reported by: JimDickenson
Patches:
      pong2.diff uploaded by JimDickenson (license 710)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203783 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-26 20:52:19 +00:00
Russell Bryant 0264eef115 Merge the new Channel Event Logging (CEL) subsystem.
CEL is the new system for logging channel events.  This was inspired after
facing many problems trying to represent what is possible to happen to a call
in Asterisk using CDR records.  For more information on CEL, see the built in
HTML or PDF documentation generated from the files in doc/tex/.

Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard
work developing this code.  Also, thanks to Matt Nicholson (mnicholson) and
Sean Bright (seanbright) for their assistance in the final push to get this
code ready for Asterisk trunk.

Review: https://reviewboard.asterisk.org/r/239/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203638 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-26 15:28:53 +00:00
Sean Bright 5059530d62 Change some section names in the CDR tex documentation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202841 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-23 23:57:07 +00:00
Sean Bright 233f1bc8da Remove some trailing whitespace before making content changes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202840 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-23 23:53:45 +00:00
Russell Bryant 1fe9c73c19 Clean up section hierarchy for the CDR chapter.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202804 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-23 22:47:26 +00:00
David Vossel 82e4b25230 ast_channel_datastore_alloc is no longer used. updating datastores.txt to reflect that.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201453 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-17 20:00:51 +00:00
Russell Bryant dcc651d99b Add missing closure of verbatim environment.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200762 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-16 01:26:20 +00:00
Kevin P. Fleming aaeec3b40f Last batch of 'static' qualifiers for module-level global variables.
Fix up modules in the 'apps' directory, and also correct the bad example of
enum definitions in include/asterisk/app.h, which many developers followed
(thanks for reading the documentation!). In addition, add some basic usage
examples of the 'pahole' and 'pglobal' tools to the coding guidelines.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200656 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-15 19:10:10 +00:00
Tilghman Lesher f3534a5a67 Add sigaction janitor
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199696 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-08 22:08:44 +00:00
Mark Michelson 298d745fb4 Add the ability to execute connected line interception macros.
When connected line updates are received or generated in the middle
of an application call, it is now possible to execute a macro to
manipulate the connected line data. This way, phone numbers may be
manipulated to be more presentable to users, names may be changed 
for...whatever reason, or whatever else needs to be done may be.

Review: https://reviewboard.asterisk.org/r/256

AST-165



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198727 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-01 20:57:31 +00:00
Terry Wilson ce004fbf1f Add some TeX docs for calendaring.
I still need to set up tests to make sure my examples are completely correct,
but I ran out of time tonight and felt that they at least would give an idea as
to how to use calendaring. I will try to test the examples and do some cleanup
on the docs tomorrow night.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197926 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-29 05:15:40 +00:00
Sean Bright f51bb019bb Update references to bugs.digium.com and reviewboard.digium.com to the new URLs.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197824 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-28 21:50:27 +00:00
Gavin Henry a5fc03b683 closes issue #15156
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197406 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-28 10:43:51 +00:00
Eliel C. Sardanons 2c882626a0 Implement a new element in AstXML for AMI actions documentation.
A new xml element was created to manage the AMI actions documentation,
using AstXML.
To register a manager action using XML documentation it is now possible
using ast_manager_register_xml().
The CLI command 'manager show command' can be used to show the parsed
documentation.

Example manager xml documentation:
<manager name="ami action name" language="en_US">
    <synopsis>
        AMI action synopsis.
    </synopsis>
    <syntax>
        <xi:include xpointer="xpointer(...)" /> <-- for ActionID
        <parameter name="header1" required="true">
	    <para>Description</para>
	</parameter>
	...
    </syntax>
    <description>
        <para>AMI action description</para>
    </description>
    <see-also>
    	...
    </see-also>
</manager>



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196308 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-22 17:52:35 +00:00
Eliel C. Sardanons 2f996a3ab4 Allow to include sections of other parts of the xml documentation.
Avoid duplicating xml documentation by allowing to include other parts of
the xml documentation using XInclude.
Example:
   <xi:include xpointer="xpointer(/docs/function[@name='CHANNEL']/synopsis)" />
(Insert this line to include the synopsis of the CHANNEL function xml
documentation).

It is also possible to include documentation from other files in the
'documentation/' directory using the href="" attribute inside a xinclude
element.

(closes issue #15107)
Reported by: lmadsen

(issue #14444)
Reported by: ewieling



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194982 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-16 20:01:22 +00:00
Eliel C. Sardanons 77c41d700e Allow to specify an enumlist inside an enum.
It was not possible to use an enumlist inside an enum:
<enumlist>
   <enum name="aa">
      <enumlist>
         ...
      </enumlist>
   </enum>
</enumlist>
Now we will be able to insert as many levels as we want.

(closes issue #15112)
Reported by: lmadsen



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194635 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-15 13:23:37 +00:00
Tilghman Lesher 829807cd2c Add numbers in Urdu, the national language of Pakistan
(closes issue #15034)
 Reported by: nasirq
 Patches: 
       ast_say_number_full_ur-patch.c uploaded by nasirq (license 772)
       urdu.ods uploaded by nasirq (license 772)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-06 16:28:02 +00:00
Tilghman Lesher 4bf441ca39 UniqueID column has a maximum size of 150
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190904 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-28 17:31:43 +00:00
Mark Michelson 1d941ad821 Allow for a position to be specified when entering a queue.
This would allow for one to add a caller to a specific place in the
queue instead of just placing the caller in the back every time. To help
facilitate some interesting manipulations, a new channel variable called
QUEUEPOSITION has been added. When a caller is removed from a queue, his
position in that queue is stored in the QUEUEPOSITION variable. One such
strategy an administrator can employ is to allow for the removal of a caller
from one queue followed by the insertion of the same caller into a separate
queue in the same position.

Review: http://reviewboard.digium.com/r/189



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190626 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-27 16:37:51 +00:00
Olle Johansson 34080f9cbe New actions should go under "New Actions", not "new events"
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@188284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-14 14:22:39 +00:00
Joshua Colp 4eaa651a8a Add support for changing the outbound codec on a SIP call using
a dialplan variable.

This adds a dialplan variable (SIP_CODEC_OUTBOUND) which controls
the codec offered for an outgoing SIP call. This is much like the
SIP_CODEC dialplan variable and has the same restrictions. The codec
set must be one that is configured for the call.

(closes issue #13243)
Reported by: samdell3
Patches:
      13243.diff uploaded by file (license 11)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186624 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-06 16:15:30 +00:00
Russell Bryant fe0323f2f7 Replace contents of this doc with a pointer to its new home
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181292 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-11 16:19:38 +00:00
Russell Bryant 71a361e001 tabs to spaces
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181099 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-11 02:25:24 +00:00
Russell Bryant 4b9a0c8aed Add some notes on getting in contact with the dev community
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180942 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-10 22:03:16 +00:00
Russell Bryant 0576d57d49 Remove difficulty and language specifiers
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180938 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-10 21:55:49 +00:00
Russell Bryant e3a339512e Expand upon documentation of manager event project
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180935 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-10 21:45:54 +00:00
Russell Bryant 093b469ef5 add more projects
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180862 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-10 19:36:21 +00:00
Russell Bryant 04fbb5aaa6 add more project ideas
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180859 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-10 19:23:41 +00:00
Russell Bryant 91fef42422 Add current mentors list, and first pass on a project list broken out of "PineMango"
I will work on adding projects that have been sent to be via email tomorrow.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180750 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-09 22:00:42 +00:00
Russell Bryant dea550a292 Add skeleton for GSoC ideas list
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180684 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-09 14:14:34 +00:00
Mark Michelson d8d5e38f65 Add documentation for timing modules used in Asterisk
This document specifies the timing modules available in Asterisk beginning
with Asterisk 1.6.1. The document goes into detail about the differences
between each and gives a general overview of what timing is used for in
Asterisk. There is also a section which can be used to help customize
your setup or to troubleshoot timing issues you may have.

I also added messages to the DAHDI timing test used in res_timing_dahdi.c
that points to this new documentation if people experience problems.

Big thanks to all who contributed comments on this.

(closes issue #14490)
Reported by: mmichelson
Patches:
      timing.txt uploaded by mmichelson (license 60)

Review: http://reviewboard.digium.com/r/164/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179937 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-03 20:59:16 +00:00
Russell Bryant d2c5b0f1de Mark res_ais as experimental, as the binary event format is subject to change.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179164 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-27 21:47:18 +00:00
Jason Parker 796ec24ed5 Update documentation for DIALEDTIME and ANSWEREDTIME variables.
(closes issue #14566)
Reported by: klaus3000
Patches:
      ANSWEREDTIME-1.4-patch.txt uploaded by klaus3000 (license 65)
      ANSWEREDTIME-trunk-patch.txt uploaded by klaus3000 (license 65)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179057 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-27 19:04:57 +00:00
David Vossel 641dd68c4d Allows manager command to see if IAX link is trunked and encrypted. Displays what kind of encryption is enabled as well.
Manager command "iaxpeers" now shows if a link is trunked and encrypted.  Instead of encryption saying simply "yes" or "no", it now displays what type of encryption is enabled and if keyrotation is on or not.  

(closes issue #14427)
Reported by: snuffy
Patches:
	iax_show_trunks.diff uploaded by snuffy (license 35)
	2009022200_iax2_show_trunkencryption.diff.txt uploaded by mvanbaak (license 7)
Tested by: mvanbaak, dvossel, snuffy
Review: http://reviewboard.digium.com/r/173/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178300 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-24 17:42:37 +00:00
Michiel van Baak c880aa936b Add a couple of manager commands to chan_skinny
Added:
SKINNYdevices
SKINNYshowdevice
SKINNYlines
SKINNYshowline

(closes issue #14521)
Reported by: mvanbaak

Review: http://reviewboard.digium.com/r/170/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-22 23:04:37 +00:00
Tilghman Lesher 1e19ca567c Document how to use database transactions
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177506 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-19 19:46:13 +00:00
Michiel van Baak 115c6abef4 Merged revisions 175921 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r175921 | mvanbaak | 2009-02-16 00:37:03 +0100 (Mon, 16 Feb 2009) | 3 lines
  
  fix mis-spelling of the word registered.
  Reported by De_Mon on #asterisk-dev.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175952 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-16 00:26:59 +00:00
Mark Michelson 3c9667ae12 Merge queue-reset branch to Asterisk
From a user point-of-view, this adds new CLI commands and Manager Actions to
better facilitate the reloading of queues and the resetting of their statistics.

The new CLI commands are the "queue reload" and "queue reset stats" commands.

The new manager actions are the QueueReload and QueueReset commands.

Review: http://reviewboard.digium.com/r/115



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175663 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-13 20:57:37 +00:00
Mark Michelson b5bde28d9f Add manager events for chanspy starting or stopping
(closes issue #14469)
Reported by: caio1982
Patches:
      chanspy_events2.diff uploaded by caio1982 (license 22)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175655 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-13 20:35:26 +00:00
Richard Mudgett 7ed9ece337 Fix asterisk.pdf generation if branch name has an underscore in it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170794 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-23 23:10:34 +00:00
Russell Bryant ac7bd2ddd9 Don't blow up if a branch name has an underscore in it
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170790 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-23 22:58:37 +00:00
Michiel van Baak 2c6a7907ff remove duplicated sentence.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169793 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-21 22:04:16 +00:00
Sean Bright e1f941d7f6 Mostly just whitespace, but also convert 'CVS' to 'SVN' in a couple
places and fix a few typos I found in the CODING_GUIDELINES.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167061 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-31 23:07:14 +00:00
Russell Bryant 0ab0896516 argsep is used as an attribute for an argument, as well
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165145 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-17 17:56:25 +00:00
Tilghman Lesher 27cbfc1bd5 Add timezone to the possible fields in a timespec.
(closes issue #14028)
 Reported by: mostyn
 Patches: 
       timezone-v2.patch uploaded by mostyn (license 398)
       (with additional code guideline fixes and a memory leak fix by me - license 14)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164976 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-16 22:57:17 +00:00
Sean Bright 45bc07c439 Use tables instead of ASCII art. Also change a bit of minor formatting.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164565 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-16 01:52:32 +00:00
Sean Bright c0fc8edbbd Use a \picture instead of ASCII art.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164137 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-14 15:26:37 +00:00
Sean Bright 3b96ae826e This shouldn't have gotten commited. We might want to generate this into a separate file instead of the version controlled one.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163952 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-13 03:03:15 +00:00
Sean Bright 3f12b282b6 Use actual tables instead of ASCII art ones.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-13 03:00:26 +00:00
Sean Bright 3b6fc0369f Fix some of the grammar issues in doc/tex/qos.tex.
(closes issue #14049)
Reported by: kshumard
Patches:
      doc.tex.qos.tex.patch uploaded by kshumard (license 92)
      (Slight modifications by seanbright)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163037 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-11 15:40:07 +00:00
Mark Michelson 5e24d64e0b Merged revisions 162659 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r162659 | mmichelson | 2008-12-10 10:10:25 -0600 (Wed, 10 Dec 2008) | 8 lines

Add missing documentation to misdn.txt

(closes issue #14052)
Reported by: festr
Patches:
      misdn.txt.patch uploaded by festr (license 443)


........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162667 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-10 16:39:10 +00:00
Joshua Colp 135bb29ba6 Finish conversion to using ARRAY_LEN and remove it as a janitor project.
(closes issue #14032)
Reported by: bkruse
Patches:
      14032.patch uploaded by bkruse (license 132)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162542 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-10 01:09:06 +00:00
Eliel C. Sardanons a22928b853 Introduce XML documentation for:
- MeetMe()
  - MeetMeCount()
  - MeetMeChannelAdmin()
  - MeetMeAdmin()
  - SLAStation()
  - SLATrunk()

- Add an attribute to optionlist 'hasparams' with the same functionality as the one
we have in <parameter> and <argument> (the DTD was updated)
- Fix a leak when getting an attribute while parsing an <optionlist>.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156575 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-13 15:46:06 +00:00
Eliel C. Sardanons 5b25c26318 - Add 'database del', 'database put' and 'set music' AGI commands XML documentation.
- Add to the DTD the possibility to put a parameter inside an <enum>.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156087 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-12 02:20:05 +00:00
Eliel C. Sardanons 6f31fed83f Implement AGI XML documentation parsing functions.
A new <agi> element is used to describe the XML documentation.
We have the usual synopsis,syntax,description and seealso for AGI commands.
The CLI 'agi show commands' command was changed to show all the documentation se
ctions.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156051 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-12 00:17:43 +00:00
Tilghman Lesher 7841475abe Merged revisions 155803 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r155803 | tilghman | 2008-11-10 14:49:59 -0600 (Mon, 10 Nov 2008) | 1 line
  
  I got tired of saying this in every single bugnote referring to this file.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@155804 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-10 20:50:50 +00:00
Eliel C. Sardanons f3d95a2131 We now can have a reference to a filename inside a <see-also> tag.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154617 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-05 13:54:21 +00:00
Russell Bryant 5b168ee34b Merge changes from team/group/appdocsxml
This commit introduces the first phase of an effort to manage documentation of the
interfaces in Asterisk in an XML format.  Currently, a new format is available for
applications and dialplan functions.  A good number of conversions to the new format
are also included.

For more information, see the following message to asterisk-dev:

http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-01 21:10:07 +00:00
Olle Johansson 5aa23add8c Adding a small new feature.
Setting _SIPFROMDOMAIN in a channel will set the domain we use for the URI in the outbound call leg.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@151739 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-23 15:30:16 +00:00
BJ Weschke 7a8344bac6 The QueueEntry event now has the uniqueid of the channel included.
(closes issue #13731)
 reported and patched by: caio1982



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150773 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-18 00:25:18 +00:00
Mark Michelson 32ef7bedd3 Add an IAXregistry manager command. See doc/manager_1_1.txt
for more details of this command.

(closes issue #13326)
Reported by: ib2
Patches:
      bug13326_trunk_20080822.diff uploaded by snuffy (license 35)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-17 00:18:01 +00:00
Leif Madsen f813260584 Add missing documentation for SipShowRegistry action and RegistryEntry event.
(closes issue #13342)
Reported and patch by: Laureano



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@149040 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-14 19:35:54 +00:00
Steve Murphy e235a07376 (closes issue #13557)
Reported by: nickpeirson
Patches:
      pbx.c.patch uploaded by nickpeirson (license 579)
      replace_bzero+bcopy.patch uploaded by nickpeirson (license 579)
Tested by: nickpeirson, murf

1. replaced all refs to bzero and bcopy to memset and memmove instead.
2. added a note to the CODING-GUIDELINES
3. add two macros to asterisk.h to prevent bzero, bcopy from creeping
   back into the source
4. removed bzero from configure, configure.ac, autoconfig.h.in




git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147807 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-09 14:17:33 +00:00
Richard Mudgett e3311c9875 * Miscellaneous formatting changes to make v1.4 and trunk
more merge compatible in the mISDN area.

channels/chan_misdn.c
*  Eliminated redundant code in cb_events() EVENT_SETUP



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-30 21:00:54 +00:00
Michiel van Baak abed7e031a fix privacymanager example so it shows how to use the
PRIVACYMRGSTATUS variable


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@143843 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-21 09:53:01 +00:00
Michiel van Baak 11f9bf9ad0 document the new context argument for privacymanager
so people can do pattern matching on the input


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@143840 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-21 09:31:54 +00:00
Michiel van Baak faa847f0ed fix privacy documentation. We no longer do priority jumping +101
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@143837 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-21 09:27:08 +00:00
Tilghman Lesher 6dd5b8609f Optional light colored background, for those who use black on white terminals.
(closes issue #13306)
 Reported by: Corydon76
 Patches: 
       20080814__bug13306__3.diff.txt uploaded by Corydon76 (license 14)
 Tested by: Corydon76, pkempgen


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@139981 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-25 23:13:32 +00:00
Tilghman Lesher 8d62c61678 Realtime capabilities for the Find-Me-Follow-Me application.
(closes issue #13295)
 Reported by: Corydon76
 Patches: 
       20080813__followme_realtime_enabled.diff.txt uploaded by Corydon76 (license 14)
 Tested by: dferrer


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@139775 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-25 16:02:56 +00:00
Richard Mudgett 1678a005b6 channels/chan_misdn.c
*  Made bearer2str() use allowed_bearers_array[]
*  Made use the causes.h defines instead of hardcoded numbers.
*  Made use Asterisk presentation indicator values if either of the
mISDN presentation or screen options are negative.
*  Updated the misdn_set_opt application option descriptions.
*  Renamed the awkward Caller ID presentation misdn_set_opt
application option value not_screened to restricted.
Deprecated the not_screened option value.

channels/misdn/isdn_lib.c
*  Made use the causes.h defines instead of hardcoded numbers.
*  Fixed some spelling errors and typos.
*  Added all defined facility code strings to fac2str().

channels/misdn/isdn_lib.h
*  Added doxygen comments to struct misdn_bchannel.

channels/misdn/isdn_lib_intern.h
*  Added doxygen comments to struct misdn_stack.

channels/misdn_config.c
configs/misdn.conf.sample
*  Updated the mISDN presentation and screen parameter descriptions.

doc/tex/misdn.tex
*  Updated the misdn_set_opt application option descriptions.
*  Fixed some spelling errors and typos.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@138738 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-18 21:07:28 +00:00
Russell Bryant 3662d60974 Fix a bashism that causes an error when trying to build the pdf on ubuntu
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137987 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-14 22:43:15 +00:00
Sean Bright 16ff15a47a Merged revisions 137405 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r137405 | seanbright | 2008-08-13 10:33:49 -0400 (Wed, 13 Aug 2008) | 1 line

Update docs to reflect the change to cdr_tds
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137406 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-13 14:41:49 +00:00
Russell Bryant 8fb13c6803 Grammar hax from Qwell
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137301 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-12 19:48:38 +00:00
Russell Bryant f6bda9def2 Note that developer documentation belongs in doxygen, and not integrated with
the user manual stuff in doc/tex/.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137299 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-12 19:40:35 +00:00
Sean Bright 6cf6d9eca5 Merge in changes that allow Asterisk to be built against the Hoard
memory allocator.  See doc/hoard.txt for more details.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135405 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-03 16:14:14 +00:00
Russell Bryant f4c4469f38 Fix some tex errors
(closes issue #13211)
Reported by: eliel
Patches:
      fixtexerrors.patch uploaded by eliel (license 64)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134803 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-31 16:37:08 +00:00
Tilghman Lesher d86fc7fcc1 Add %u and %g to the ASTERISK_PROMPT settings, for username and group,
respectively.  Also, take the opportunity to clean up the CLI prompt
generation code.
(closes issue #13175)
 Reported by: eliel
 Patches: 
       cliprompt.patch uploaded by eliel (license 64)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134353 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-30 15:30:18 +00:00
Kevin P. Fleming 6291cd19bf remove remaining Zaptel references in various places
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134086 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-28 16:42:00 +00:00
Tilghman Lesher c4d4c636d0 Hebrew syntax for voicemail prompts
(closes issue #13155)
 Reported by: greenfieldtech
 Patches: 
       app_voicemail.c.patch uploaded by greenfieldtech (license 369)
       hebrew.ods uploaded by greenfieldtech (license 369)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@133904 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-25 22:08:06 +00:00
Olle Johansson d231a9cf7d Merged revisions 132645 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r132645 | oej | 2008-07-22 22:10:26 +0200 (Tis, 22 Jul 2008) | 9 lines

The most common question on the #asterisk iRC channel and on mailing lists
seems to be in regards to an error message when retransmit fails. This
is frequently misunderstood as a failure of Asterisk, not a failure of
the network to reach the other party.

This document tries to assist the Asterisk user in sorting out these
issues by explaining the logic and pointing at some possible 
causes. Hopefully, we will get other questions now :-)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@132703 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-22 20:46:11 +00:00
Michiel van Baak 0d49cdae3e Make all sed calls Posix sed compatible.
To make sure nobody commits script-modified files we first make a backup
of asterisk.tex, run the script, generate the pdf and / or html,
and put the original asterisk.tex back.
This will guard us for the stuff that happened before that someone committed 
a locally modified asterisk.tex, with changes done by this script.

(closes issue #13062)
Reported by: mvanbaak
Patches:
      sed_without-i-v3.diff uploaded by mvanbaak (license 7)
Tested by: mvanbaak

Feedback from Corydon. Thanks for taking the time to go through this.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130578 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-13 23:14:03 +00:00
Michiel van Baak 8faa81be79 restore ASTERISKVERSION marker to asterisk.tex.
This got lost in commit 97634


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130479 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-13 14:58:40 +00:00
Kevin P. Fleming dd7630222c clean up a bunch more Zaptel-related references
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-11 16:18:01 +00:00
Mark Michelson a92e934075 Update a few instances of "extensions reload" to "dialplan reload"
in the documentation.

Patch provided by caio1982 (license 22)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128599 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-07 14:35:27 +00:00
Olle Johansson ef943d774d Adding documentation on the T.140 support in Asterisk. This is a function that we're
the reference implementation on now. :-)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128417 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-06 10:13:45 +00:00
Mark Michelson 55e4daef88 Add a janitor project to use ARRAY_LEN instead of in-line
sizeof() and division.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127566 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-02 21:09:18 +00:00
Russell Bryant 604b819b1c Add a locking section to the coding guidelines document.
This section covers some locking fundamentals, as well as some information on
locking as it is used in Asterisk.  It describes some of the ways that are used
and could be used to achieve deadlock avoidance.  It also demonstrates the
unfortunate conclusion that with the use of recursive locks, none of the
constructs in use today are failsafe from deadlocks.  Finally, it makes some
recommendations for new code being written.  As proper locking strategies is a
complex subject, this section still has room for expansion and improvement.

This is a result of collaboration between Luigi Rizzo and myself on the 
asterisk-dev mailing list.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127363 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-02 12:08:33 +00:00
Luigi Rizzo 52c6ee5155 add documentation on video console support
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@126959 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-01 16:14:28 +00:00
Russell Bryant dc5c54eaf2 a few minor updates and typo fixes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@126515 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-30 12:49:56 +00:00
Sean Bright dd0c0f3d38 Cast a few more strings to char *, so that we can compile cleanly against
FreeTDS 0.60.  Update the docs to reflect that we can now compile and run
against all modern releases of FreeTDS (0.60 through 0.82)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@126513 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-30 11:57:42 +00:00
Russell Bryant d4ddf14b6b Fix a typo. Someone on IRC copied this literally and then wondered why it
wasn't working.  :)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-27 19:19:08 +00:00
Jeff Gehlbach 7e9bb68ec8 Merged revisions 124372 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r124372 | jeffg | 2008-06-20 17:14:40 -0400 (Fri, 20 Jun 2008) | 1 line

Fix issues in digium-mib.txt and asterisk-mib.txt to placate smilint - bug 12905
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@124392 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-20 21:36:01 +00:00
Michiel van Baak 8e8359465b Older versions of GNU gcc do not allow 'NULL' as sentinel.
They want (char *)NULL as sentinel.
An example is OpenBSD (confirmed on 4.3) that ships with gcc 3.3.4

This commit introduces a contstant SENTINEL which is declared as:
#define SENTINEL ((char *)NULL)

All places I could test compile on my openbsd system are converted.
Update CODING-GUIDELINES to tell about this constant.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@124127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-19 20:48:33 +00:00
Sean Bright a41649d66b Update the queuelog.tex documentation as well.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@123830 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-19 14:28:56 +00:00
Tilghman Lesher a06ab7b648 Merged revisions 123769 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r123769 | tilghman | 2008-06-18 17:08:30 -0500 (Wed, 18 Jun 2008) | 8 lines

Add support for saying numbers in Hebrew.
(closes issue #11662)
 Reported by: greenfieldtech
 Patches: 
       say.c.patch-12042008 uploaded by greenfieldtech (license 369)
       Hebrew-Sounds.ods uploaded by greenfieldtech
       (with signficant changes to the spreadsheet by me)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@123770 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-18 22:17:17 +00:00
Jeff Peeler ef3b214728 Goodbye Zaptel, hello DAHDI. Removes Zaptel driver support with DAHDI. Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122234 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-12 17:27:55 +00:00
Jeff Peeler d373bd4430 Merged revisions 121804 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r121804 | jpeeler | 2008-06-11 11:11:09 -0500 (Wed, 11 Jun 2008) | 1 line

add instructions for logging gdb output via set logging on
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121805 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-11 16:11:40 +00:00
Russell Bryant 7123fb41c4 don't refer to asterisk-events, as that implies that the code was checked out from a branch
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121716 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-10 21:51:06 +00:00
Russell Bryant e9d72e0cb2 Merge another big set of changes from team/russell/events
This commit merges in the rest of the code needed to support distributed device
state.  There are two main parts to this commit.

Core changes:
 - The device state handling in the core has been updated to understand device
   state across a cluster of Asterisk servers.  Every time the state of a device
   changes, it looks at all of the device states on each node, and determines the
   aggregate device state.  That resulting device state is what is provided to
   modules in Asterisk that take actions based on the state of a device.

New module, res_ais:
 - A module has been written to facilitate the communication of events between
   nodes in a cluster of Asterisk servers.  This module uses the SAForum AIS
   (Service Availability Forum Application Interface Specification) CLM and EVT
   services (Cluster Management and Event) to handle this task.  This module
   currently supports sharing Voicemail MWI (Message Waiting Indication) and
   device state events between servers.  It has been tested with openais, though
   other implementations of the spec do exist.

For more information on testing distributed device state, see the following doc:
  - doc/distributed_devstate.txt


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-10 15:12:17 +00:00
Russell Bryant f4a8062e93 Merge another change from team/russell/events ...
DUNDi uses a concept called the Entity ID for unique server identifiers.  I have
pulled out the handling of EIDs and made it something available to all of Asterisk.
There is now a global Entity ID that can be used for other purposes as well, such
as code providing distributed device state, which is why I did this.  The global
Entity ID is set automatically, just like it was done in DUNDi, but it can also be
set in asterisk.conf.  DUNDi will now use this global EID unless one is specified
in dundi.conf.

The current EID for the system can be seen in the "core show settings" CLI command.
It is also available in the dialplan via the ENTITYID variable.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121439 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-10 12:48:50 +00:00
Jeff Peeler 3c6eb61008 add document describing API changes from 1.4.0 to 1.6.0
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@118178 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-24 01:14:41 +00:00
Tilghman Lesher ee29d6dc2b Merged revisions 118052 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r118052 | tilghman | 2008-05-23 07:59:16 -0500 (Fri, 23 May 2008) | 3 lines

Add information on using the Asterisk console, including tab command line
completion.  (Closes issue #12681)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@118053 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-23 13:00:10 +00:00
Tilghman Lesher ce8453f57c Enhance ExternalIVR with new options and commands.
(closes issue #12705)
 Reported by: ctooley
 Patches: 
       new_externalivr_argument_format-v2.diff uploaded by ctooley (license 136)
       new_externalivr_documentation.diff uploaded by ctooley (license 136)
       and a few additional fixes by me


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117725 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-22 05:10:01 +00:00
Mark Michelson b0e637791c Add a new manager event, AgentRingNoAnswer to
app_queue.

(closes issue #12591)
Reported by: CCHAsteria
Patches:
      app_queue_RNA_event.diff uploaded by CCHAsteria (license 477)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117625 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-21 20:27:45 +00:00
Olle Johansson e30c26c689 Add support for playing an audio file for caller and callee at start and stop of monitoring (one-touch monitor).
Keep messages short, since the other party is waiting while one party hear the message...



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115784 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-12 18:39:09 +00:00
Russell Bryant 44af1e23d0 Merge changes from team/russell/smdi-msg-searching
This commit adds some new features to the SMDI_MSG_RETRIEVE() dialplan function.
Previously, this function only allowed searching by the forwarding station.
I have added some options to allow you to also search for messages in the queue
by the message desk terminal ID, as well as the message desk number.

This originally came up as a suggestion on the asterisk-dev mailing list.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115021 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-01 19:05:36 +00:00
Steve Murphy c0b8f57b9d (closes issue #12467)
Reported by: atis
Tested by: murf

This upgrade adds the ~~ (concatenation) string operator to expr2.
While not needed in normal runtime pbx operation, it is needed when
raw exprs are being syntax checked. This plays into future syntax-
unification plans. By permission of atis, this addition in trunk 
and the reason of why things are as they are will suffice to close
this bug.

I also added a short note about the previous addition of "sip show sched"
to the CLI in CHANGES, which I discovered I forgot in a previous commit.




git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-21 21:13:02 +00:00
Russell Bryant e29b987b30 Add a simple janitor project
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114325 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-21 15:01:04 +00:00
Tilghman Lesher 631d8e5fbe fileio.h does not exist; io.h does, though.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114202 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-17 15:12:52 +00:00
Steve Murphy 5fb4b1bbe5 This is the scariest commit I've done in a long time. This is the astobj2-ification of chan_sip. I've tested a number of scenarios like crazy. It used to have 4x the call setup/teardown performance of trunk, but now it's roughly at parity. I will attempt to find the bottlenecks and get it back to the 4x mark. The changes made were somewhat invasive, but the value to the community of these upgrades outweighs waiting further for more testing. Every change being made to chan_sip was lousing this code up when we tried to merge. Peers, Users, Dialogs, are all now astobj2 objects, indexed via hashtables. Refcounting is used to track objects and free them at the bitter end of their lives. Please file issues on bugs.digium.com, and PLEASE, please, please be patient. One natural advantage to all the hash-table work is that loading large sip.conf files full of thousands of peers now goes much faster. One more please: PLEASE help thrash this code and test it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114190 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-16 23:53:27 +00:00
Russell Bryant afd8783577 Make some notes about common usage of pbx_builtin_getvar_helper() that is not
thread-safe.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111909 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-28 22:50:46 +00:00
Tilghman Lesher 3f78a375a0 Merged revisions 111605 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r111605 | tilghman | 2008-03-28 09:35:45 -0500 (Fri, 28 Mar 2008) | 3 lines

Update debugging text, since Valgrind eliminated the --log-file-exactly option.
(Closes issue #12320)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-28 14:37:28 +00:00
Donny Kavanagh 7a2d8fc309 update documentation to reflect the changes in the way configure detects net-snmp.
(closes issue #12067)
Reported by: juggie
Patches:
      12067_snmp_doc.patch uploaded by juggie (license 24)
Tested by: juggie


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110911 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-26 17:24:54 +00:00
Jeff Peeler 3c4c3c0dd2 documenting changes as a result of adding TCP functionality to ExternalIVR
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108639 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-13 23:12:59 +00:00
Jason Parker ce66adb981 Merged revisions 107826 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r107826 | qwell | 2008-03-11 18:37:05 -0500 (Tue, 11 Mar 2008) | 7 lines

Update documentation for pgsql ODBC voicemail.

(closes issue #12186)
Reported by: jsmith
Patches:
      vm_pgsql_doc_update.patch uploaded by jsmith (license 15)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107827 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-11 23:38:00 +00:00
Russell Bryant de31be8741 fix example usage
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106684 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-07 16:31:48 +00:00
Russell Bryant 5fa773a9f9 minor text changes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106518 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-07 01:19:02 +00:00
Russell Bryant 56e6a4b2bb Add updated SMDI documentation that I had only sitting in my email ... oops
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106507 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-07 01:15:36 +00:00
Mark Michelson 2ed30d47e8 Adding the Atxfer manager command. With this, you may initiate
an attended transfer over AMI

(closes issue #10585)
Reported by: ornati
Patches:
      atxfer-trunk-r90428.diff uploaded by ornati (license 210)
	  (with modifications from me)
Tested by: putnopvut



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106236 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-05 22:33:05 +00:00
Michiel van Baak eafa9b0de0 document var_metric usage to prevent bugreports that are actually configuration issues
(closes issue #12151)
Reported by: caio1982
Patches:
      DB_metric3.diff uploaded by caio1982 (license 22)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106186 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-05 21:19:06 +00:00
Russell Bryant 0e3ef53ddd note that the chan_sip conversion is already in progress
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104473 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-27 04:42:59 +00:00