Commit Graph

1637 Commits

Author SHA1 Message Date
Matthew Jordan bb955e37fb Provide proper ring tone in indications.conf for Malaysia
The ring tone provided in the sample indications.conf was incorrect. This patch
modifies the sample ring tone to be what it should:
  ring = 425/400,0/200,425/400,0/2000

This brings it in line with the tone definition in DAHDI 2.7.0. (zonedata.c)

(closes issue ASTERISK-21997)
Reported by: Filip Jenicek
patches:
  malaysia_ring.patch uploaded by phill (License 6277)
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Merged revisions 394940 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 394941 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394942 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-21 18:17:17 +00:00
Matthew Jordan 54803338b4 Always install safe_asterisk; add configuration file support
This patch modifies the behavior of safe_asterisk in two ways:
(1) It modifies the Asterisk Makefile such that safe_asterisk is always
    installed on a 'make install'. This was done as bugfixes in the
    safe_asterisk script were not applied in previous version of Asterisk
    without first removing the old version of the script.
(2) In order to keep a newly installed version of safe_asterisk from impacting
    local modifications, a new config file - safe_asterisk.conf.sample - has
    been provided. Settings that were previously modified in safe_asterisk can
    be set there instead.

(closes issue ASTERISK-21965)
Reported by: Jeremy Kister
patches:
  safe_asterisk.patch uploaded by jkister (License 6232)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-21 18:12:00 +00:00
Matthew Jordan 75e83bdbab Document connectedline parameter for chan_iax2
The connectedline parameter for a chan_iax2 peer was undocumented. This patch
documents the options in the sample configuration file.

(closes issue ASTERISK-21953)
Reported by: Birger "WIMPy" Harzenetter
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Merged revisions 394886 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 394890 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-21 02:26:31 +00:00
Richard Mudgett d43b17a872 Replace chan_agent with app_agent_pool.
The ill conceived chan_agent is no more.  It is now replaced by
app_agent_pool.

Agents login using the AgentLogin() application as before.  The
AgentLogin() application no longer does any authentication.
Authentication is now the responsibility of the dialplan.  (Besides, the
authentication done by chan_agent did not match what the voice prompts
asked for.)

Sample extensions.conf
[login]
; Sample agent 1001 login
; Set COLP for in between calls so the agent does not see the last caller COLP.
exten => 1001,1,Set(CONNECTEDLINE(all)="Agent Waiting" <1001>)
; Give the agent DTMF transfer and disconnect features when connected to a caller.
same => n,Set(CHANNEL(dtmf-features)=TX)
same => n,AgentLogin(1001)
same => n,NoOp(AGENT_STATUS is ${AGENT_STATUS})
same => n,Hangup()

[caller]
; Sample caller direct connect to agent 1001
exten => 800,1,AgentRequest(1001)
same => n,NoOp(AGENT_STATUS is ${AGENT_STATUS})
same => n,Hangup()

; Sample caller going through a Queue to agent 1001
exten => 900,1,Queue(agent_q)
same => n,Hangup()

Sample queues.conf
[agent_q]
member => Local/800@caller,,SuperAgent,Agent:1001

Under the hood operation overview:
1) Logged in agents wait for callers in an agents holding bridge.
2) Caller requests an agent using AgentRequest()
3) A basic bridge is created, the agent is notified, and caller joins the
   basic bridge to wait for the agent.
4) The agent is either automatically connected to the caller or must ack
   the call to connect.
5) The agent is moved from the agents holding bridge to the basic bridge.
6) The agent and caller talk.
7) The connection is ended by either party.
8) The agent goes back to the agents holding bridge.

To avoid some locking issues with the agent holding bridge, I needed to
make some changes to the after bridge callback support.  The after bridge
callback is now a list of requested callbacks with the last to be added
the only active callback.  The after bridge callback for failed callbacks
will always happen in the channel thread when the channel leaves the
bridging system or is destroyed.

(closes issue ASTERISK-21554)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/2657/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394417 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-15 23:20:55 +00:00
David M. Lee 684481b74c Change ARI user config to use a type field
When I initially wrote the configuration support for ARI users, I
determined the section type by a category prefix (i.e., [user-admin]).

This is neither idiomatic Asterisk configuration, nor is it really
that user friendly. This patch replaces the category prefix with a
type field in the section, which is much cleaner.

Review: https://reviewboard.asterisk.org/r/2664/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394076 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-11 14:39:55 +00:00
Russell Bryant 0bfe2d4cc4 astobj2-ify the SLA code
The SLA code within app_meetme was written before asotbj2 had been
merged into Asterisk.  Worse, support for reloads did not exist at first
and was added later as a bolt-on feature.  I knew at the time that
reloading was not safe at all while SLA was in use, so the reload would
be queued up to execute when the system was idle.  Unfortunately, this
approach was still prone to errors beyond the fact that this was the
only place in Asterisk where configuration was not reloaded
instantly when requested.

This patch converts various SLA objects to be reference counted objects
using astobj2.  This allows reloads to be processed while the system is
in use.  The code ensures that the objects will not disappear while one
of the other threads is using them.  However, they will be immediately
removed from the global trunk and station containers so no new calls
will use them if removed from configuration.

Review: https://reviewboard.asterisk.org/r/2581/
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Merged revisions 393928 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 393929 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393930 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-10 01:56:15 +00:00
Richard Mudgett 02f55a36a0 Revert accidental overcommit.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393632 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-03 23:57:37 +00:00
Richard Mudgett b4e9a3fc2f Add BUGBUG note for ASTERISK-22009
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393631 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-03 23:55:53 +00:00
David M. Lee c4adaf9106 Configuration for Stasis threadpool
The appropriate settings for the Stasis threadpool is very system
specific, depending upon both workload and system configuration.

This patch adds a stasis.conf file which can be used to configure the
key attributes of the threadpool for the Stasis message bus.

(closes issue ASTERISK-21280)
Review: https://reviewboard.asterisk.org/r/2651/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393542 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-03 17:20:43 +00:00
David M. Lee 9ba976b19c ARI authentication.
This patch adds authentication support to ARI.

Two authentication methods are supported. The first is HTTP Basic
authentication, as specified in RFC 2617[1]. The second is by simply
passing the username and password as an ?api_key query parameter
(which allows swagger-ui[2] to authenticate more easily).

ARI usernames and passwords are configured in the ari.conf file
(formerly known as stasis_http.conf). The user may be set to
`read_only`, which will prohibit the user from issuing POST, DELETE,
etc. Also, the user's password may be specified in either plaintext,
or encrypted using the crypt() function.

Several other notes about the patch.

 * A few command line commands for seeing ARI config and status were
   also added.
 * The configuration parsing grew big enough that I extracted it to
   its own file.

 [1]: http://www.ietf.org/rfc/rfc2617.txt [2]:
 https://github.com/wordnik/swagger-ui

(closes issue ASTERISK-21277)
Review: https://reviewboard.asterisk.org/r/2649/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393530 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-03 16:33:13 +00:00
Jonathan Rose f306dbd841 bridge_features: Support One touch Monitor/MixMonitor
In addition to porting those features, they now enjoy greater feature parity
with one another. Specifically, AutoMixMon now has a start and stop
message that can be specified with TOUCH_MIXMONITOR_MESSAGE_START and
TOUCH_MIXMONITOR_MESSAGE_STOP.

(closes issue ASTERISK-21553)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2620/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393309 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-01 16:01:24 +00:00
Kinsey Moore 909ee4bfb9 Refactor extraneous channel events
This change removes JitterBufStats, ChannelReload, and ChannelUpdate
and refactors the following events to travel over Stasis-Core:
* LocalBridge
* DAHDIChannel
* AlarmClear
* SpanAlarmClear
* Alarm
* SpanAlarm
* DNDState
* MCID
* SIPQualifyPeerDone
* SessionTimeout

Review: https://reviewboard.asterisk.org/r/2627/
(closes issue ASTERISK-21476)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-01 13:16:09 +00:00
Joshua Colp 77002bc377 Merge in current pimp_my_sip work, including:
1. Security events
2. Websocket support
3. Diversion header + redirecting support
4. An anonymous endpoint identifier
5. Inbound extension state subscription support
6. PIDF notify generation
7. One touch recording support (special thanks Sean Bright!)
8. Blind and attended transfer support
9. Automatic inbound registration expiration
10. SRTP support
11. Media offer control dialplan function
12. Connected line support
13. SendText() support
14. Qualify support
15. Inband DTMF detection
16. Call and pickup groups
17. Messaging support

Thanks everyone!

Side note: I'm reminded of the song "How Far We've Come" by Matchbox Twenty.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392565 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-22 14:03:22 +00:00
Matthew Jordan c2e29abcbf Add announce-to-first-user option for app_queue
In r386792, the ability to play prompts to the first caller in a call queue was
added. While this is arguably a bug fix for those who expect the first caller
to continue receiving prompts while the agent is dialed, it has the side effect
of preventing the first caller from hearing the agent immediately upon
bridging. This may not be a problem for those who really want this option, but
for those who didn't care whether or not the first caller in queue heard their
position, it was an issue.

This patch disables the ability for the first caller in the queue to hear
prompts and adds a new option, announce-to-first-user, to queues.conf. Those
who the behavior can enable it by setting this value to True.

Note that if we ever implement the ability to have the prompts be stopped
upon bridging, this option can be removed.

(closes issue ASTERISK-21782)
Reported by: Remi Quezada
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Merged revisions 391215 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391245 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-10 14:36:15 +00:00
Jason Parker a2d02edca5 Make app_queue AMI events more consistent. Give Join/Leave more useful names.
This also removes the eventwhencalled and eventmemberstatus configuration
options.  These events can just be filtered via manager.conf blacklists.

(closes issue ASTERISK-21469)
Review: https://reviewboard.asterisk.org/r/2586/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390901 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-07 19:51:19 +00:00
Richard Mudgett bad8caa8c6 Reimplement bridging and DTMF features related channel variables in the bridging core.
* The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer
channel driver specific.  If the channel variable is set on the
transferrer channel, the sound will be played to the target of an attended
transfer.

* The channel variable BRIDGEPEER becomes a comma separated list of peers
in a multi-party bridge.  The BRIDGEPEER value can have a maximum of 10
peers listed.  Any more peers in the bridge will not be included in the
list.  BRIDGEPEER is not valid in holding bridges like parking since those
channels do not talk to each other even though they are in a bridge.

* The channel variable BRIDGEPVTCALLID is only valid for two party bridges
and will contain a value if the BRIDGEPEER's channel driver supports it.

* The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and
is removed.  The more useful DYNAMIC_WHO_ACTIVATED gives the channel name
that activated the dynamic feature.

* The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are
set only on the channel executing the dynamic feature.  Executing a
dynamic feature on the bridge peer in a multi-party bridge will execute it
on all peers of the activating channel.

(closes issue ASTERISK-21555)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/2582/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390771 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-06 22:46:54 +00:00
Richard Mudgett 3d63833bd6 Merge in the bridge_construction branch to make the system use the Bridging API.
Breaks many things until they can be reworked.  A partial list:
chan_agent
chan_dahdi, chan_misdn, chan_iax2 native bridging
app_queue
COLP updates
DTMF attended transfers
Protocol attended transfers


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-21 18:00:22 +00:00
Damien Wedhorn 01d6e8dbc9 Add call forward no answer to skinny and cleanup general callfwd handling.
CallforwardNoAnswer uses a sched to determine when to forward the call. 
Defaults to 20secs but configurable in skinny.conf.

Adds dialType to each subchannel structure to be used to differentiate
between normal dials that result in a call being placed (default) and
other uses for the skinny_dialer (such as cfwd digit collection).
Restructured all cfwd handling to use this new arrangement.

(closes issue ASTERISK-21292)
Reported by: wedhorn
Tested by: myself
Patches: 
    skinny-callfwdnoans03.diff uploaded by wedhorn (license 5019)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389097 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-18 23:20:53 +00:00
David M. Lee 946eb5ede0 Example of how to use the Stasis message bus
In order to get people familiar with the Stasis message bus, it would
be useful to have something of a tutorial. Since I'm not clever enough
to think of some cool integration we could do with Twitter, I settled
for something that might actually be useful.

This patch adds a res_statsd.so module, which implements a basic
statsd[1] client. Statsd is a very simple statistics gathering server,
which can publish its results to a backend graphing engine, like
Graphite[2]. There are several different Statsd server
implementations[3], so you can pick what works best for your
environment.

The actual example of how to use the Stasis message bus is in
res_chan_stats.so. This module demonstrates how to use subscriptions
and the message router by monitoring messages and posting channels
stats to the statsd server.

A wiki page walking through res_chan_stats.so is forthcoming.

 [1]: https://github.com/etsy/statsd/
 [2]: http://graphite.readthedocs.org/en/latest/
 [3]: http://joemiller.me/2011/09/21/list-of-statsd-server-implementations/

Review: https://reviewboard.asterisk.org/r/2460/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386624 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-04-26 20:05:15 +00:00
Joshua Colp 8c1f423cf7 Don't bind to anything in the sample configuration so we don't clash with chan_sip on a "make samples" right now.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386577 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-04-25 21:32:48 +00:00
Mark Michelson 74f2318051 Merge the pimp_my_sip branch into trunk.
The pimp_my_sip branch is being merged at this point because
it offers basic functionality, and from an API standpoint, things
are complete.

SIP work is *not* feature-complete; however, with the completion
of the SUBSCRIBE/NOTIFY API, all APIs (except a PUBLISH API) have
been created, and thus it is possible for developers to attempt
to create new SIP work.

API documentation can be found in the doxygen in the code, but
usability documentation is still lacking.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386540 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-04-25 18:25:31 +00:00
David M. Lee 1c21b8575b This patch adds a RESTful HTTP interface to Asterisk.
The API itself is documented using Swagger, a lightweight mechanism for
documenting RESTful API's using JSON. This allows us to use swagger-ui
to provide executable documentation for the API, generate client
bindings in different languages, and generate a lot of the boilerplate
code for implementing the RESTful bindings. The API docs live in the
rest-api/ directory.

The RESTful bindings are generated from the Swagger API docs using a set
of Mustache templates.  The code generator is written in Python, and
uses Pystache. Pystache has no dependencies, and be installed easily
using pip. Code generation code lives in rest-api-templates/.

The generated code reduces a lot of boilerplate when it comes to
handling HTTP requests. It also helps us have greater consistency in the
REST API.

(closes issue ASTERISK-20891)
Review: https://reviewboard.asterisk.org/r/2376/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386232 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-04-22 14:58:53 +00:00
Richard Mudgett 13e2aae2ef Fix 'pri intense debug span' alias.
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Merged revisions 385313 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-04-11 16:53:21 +00:00
Rusty Newton 98f2318559 Modified the list of keys for the driver backends for sake of sample clarity
Added a line showing the mapping of "mysql" to res_config_mysql available in add-ons. We used "mysql" as an example driver key in the sample, but didn't show what module it mapped too. Also added a subtitle above the list of keys for driver backends.
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Merged revisions 385047 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385049 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-04-08 23:38:08 +00:00
Richard Mudgett 6a25d49296 chan_dahdi: Change inband_on_proceeding option default to no/disabled.
(issue ASTERISK-21151)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384711 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-04-03 20:27:11 +00:00
Richard Mudgett 79818112fd chan_dahdi: Add inband_on_proceeding compatibility option.
The new inband_on_proceeding option causes Asterisk to assume inband audio
may be present when a PROCEEDING message is received.

Q.931 Section 5.1.2 says the network cannot assume that the CPE side has
attached to the B channel at this time without explicitly sending the
progress indicator ie informing the CPE side to attach to the B channel
for audio.  However, some non-compliant ISDN switches send a PROCEEDING
without the progress indicator ie indicating inband audio is available and
assume that the CPE device has connected the media path for listening to
ringback and other messages.

ASTERISK-17834 which causes this issue was dealing with a non-compliant
network switch.

(closes issue ASTERISK-21151)
Reported by: Gianluca Merlo
Tested by: rmudgett
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Merged revisions 384685 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384696 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-04-03 20:20:09 +00:00
David M. Lee 641fc7ea54 Sample config file for stasis-core.
(issue ASTERISK-20887)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383225 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-15 16:42:05 +00:00
Matthew Jordan 8d5c36c9bb Add RFC 3327 Path header support to chan_sip
This patch adds support for RFC 3327 "Path" headers. This can be enabled in
sip.conf using the 'supportpath' setting, either on a global basis or on a
peer basis. This setting enables Asterisk to route outgoing out-of-dialog
requests via a set of proxies by using a pre-loaded route-set defined by the
Path headers in the REGISTER request. This patch also adds Realtime support
for dynamically updating the Path information for a peer.

A huge thank-you to Klaus Darillion and Olle E Johansson for their efforts
in writing this patch.

Review: https://reviewboard.asterisk.org/r/2235/
Review: https://reviewboard.asterisk.org/r/991/

(closes issue ASTERISK-16884)
Reported by: klaus3000
Tested by: klaus3000, oej, mjordan
patches:
  path-1.8.0-patch.txt uploaded by klaus3000 (License 5054)
  oolong-path-support-trunk in team branch by oej (License 5267)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382440 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-05 13:14:43 +00:00
Matthew Jordan 33e4c6115f Ensure that the default bridge/user profiles are always available
ConfBridge and Page require that there always be a default bridge and user
profile available. While properties of the default profiles can be overriden
in the configuration file, removing them can create situations where neither
application can function properly.

This patch ensures that if an administrator removes the profiles from the
confbridge.conf configuration file, the profiles are added upon load.
Documentation clarifying this has been added to the confbridge.conf.sample file.

Review: https://reviewboard.asterisk.org/r/2356/

(closes issue AST-1115)
Reported by: John Bigelow
Tested by: John Bigelow
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-26 15:26:16 +00:00
Damien Wedhorn 0d553eece8 Add serviceURL stuff to skinny.
Patch adds all the packet and structure stuff to skinny to enable setting 
service URLs in skinny, such as corporate directories.

This stuff is only relevant during load/unload as when activated. Also 
some minor changes removing duplicated counting of addons and speedials in 
handle_skinny_show_devices.

Review: https://reviewboard.asterisk.org/r/2321/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381718 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-19 06:54:23 +00:00
Walter Doekes d4d1d10307 Remove "registertrying" and add "rtp_engine" from/to sip.conf.sample
The "registertrying" option was removed in r343220. The "rtp_engine"
option was added in r186078 but erroneously named "engine" in the sample.
Note that there is no global sip setting for a different engine.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381670 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-18 20:31:56 +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
Jonathan Rose 1a70d513f1 Call Parking: Set PARKINGLOT and PARKINGSLOT variables on all parked calls
These two variables were previously not being set when comebacktoorigin=yes
and the example configs seemed to imply that they should be. Since there
is no harm in this and since calls that are sent back to origin are capable
of continuing in the dialplan, this seemed like a no-brainer. Also it
supports some bridging tests I've been working on.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381068 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-08 17:36:23 +00:00
Damien Wedhorn 44872e797c Reset skinny vmexten and immeddial char on reload.
Make skinny reset vmexten and immeddial to '\0' on reload to ensure that
it is set to '\0' if the appropriate item is removed/commented in 
skinny.conf. Also small fix re immeddial char in skinny.conf and add
immedial setting to skinny show settings.

(closes issue ASTERISK-21037)
Reported by: snuffy
Tested by: snuffy, myself
Patches: 
    immed_dial_fix.diff uploaded by snuffy (license 5024)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380925 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-06 06:55:02 +00:00
Russell Bryant dfdf3d9909 Add queue_log_realtime_use_gmt option to logger.conf
Add an option that lets you specify that the timestamps going into the realtime
queue log should be in GMT instead of local time.

Review: https://reviewboard.asterisk.org/r/2287/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380209 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-28 01:50:54 +00:00
Joshua Colp 3fa4278a31 Merge the sorcery data access layer API.
Sorcery is a unifying data access layer which provides a pluggable mechanism to allow
object creation, retrieval, updating, and deletion using different backends (or wizards).

This is a fancy way of saying "one interface to rule them all" where them is configuration,
realtime, and anything else that comes along.

Review: https://reviewboard.asterisk.org/r/2259/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380069 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-25 14:01:04 +00:00
Damien Wedhorn e9446501c9 Add force dial keys to skinny.
Adds a dial softkey when the device is in DAFD. The softkey is greyed (unusable) 
until a possible dialplan match is entered. Code includes updating 
transmit_selectsoftkeys to allow the use of a button mask. Also add option
to use # or * as a dial now button. Original patch by snuffy cleaned up by myself.

Review: https://reviewboard.asterisk.org/r/2277/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380057 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-25 05:49:54 +00:00
Richard Mudgett 8ed2c74fe3 app_queue: Fix multiple calls to a queue member that is in only one queue.
When ringinuse=no queue members can receive more than one call if these
calls happen at nearly the same time.

* Fix so a queue member does not receive more than one call from a queue.

NOTE: This fix does not prevent multiple calls to a member if the member
is in more than one queue.

* Did some refactoring to eliminate some code redundancy.

(issue ASTERISK-16115)
Reported by: nik600
Patches:
      jira_asterisk_16115_single_q_v1.8.patch (license #5621) patch uploaded by rmudgett
      Modified

* Revert the -r341580 and -r341599 changes adding the queues.conf
check_state_unknown option as it was added in an attempt to fix this
problem.  The fix did not need to be optional.  The fix should not have
tried to explicitly set the device state.  Setting the device state by
something other than the device introduces a race condition.  I also could
not see how the change would be effective other than delaying the
app_queue code long enough for the device state to propagate to app_queue.
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2013-01-08 23:44:26 +00:00
Tilghman Lesher 7739ed9989 Add aliases to the Directory.
This is an interesting feature that allows additional strings to be used to
search the Directory, primarily intended to be used with nicknames, but could
be used with affiliations and the like.  Because the name field is used in
more than one place (such as email notifications), it is important that these
additional strings not be placed in the name field, but be specified
separately.

Review: https://reviewboard.asterisk.org/r/2244/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378414 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-03 16:04:11 +00:00
Brent Eagles ab894d5af9 This change adds a SIP peer configuration feature to allow the peer's
configured codecs to take precedence on an outgoing call.

This change introduces a new peer configuration property named
'ignore_requested_pref' that causes the requested codec to be ignored when
determining the preferred codec for an outgoing call leg. The consequence is
that Asterisk's usual efforts to prefer avoiding transcoding can be overridden
on a peer-by-peer basis where appropriate. 


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377971 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-13 15:22:27 +00:00
Joshua Colp 898ca023d5 Tweak extension used for incoming calls received on Motif.
Based on feedback from numerous individuals this patch tweaks incoming calls
to first look for an extension with the name of the endpoint. If no such extension
exists the call will silently fall back to the "s" extension as it previously
did.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376984 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-01 00:47:42 +00:00
Joshua Colp 866d968149 Remove a fixed size limitation for producing SDP and change how ICE support is disabled by default.
With ICE support enabled in chan_sip and a large number of interfaces on the system it was
possible for the produced SDP to be truncated due to some fixed size buffers. These buffers
have now been changed so they will dynamically grow as needed.

ICE support is now also enabled by default in res_rtp_asterisk to provide a smoother experience
for chan_motif users where it is required. To maintain the previous behavior in chan_sip it is
no longer enabled by default there.

(closes issue ASTERISK-20643)
Reported by: coopvr
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376131 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-11 17:15:47 +00:00
Jonathan Rose d4a357b82f chan_sip: Fix a bug causing SIP reloads to remove all entries from the registry
A regression was introduced in chan_sip by changes to sip reload introduced by
r349097. That patch moved peer purging from the beginning of the reload to
after the general configuration was finished. This patch fixes that by undoing
the repositioning of the original peer purging code and using a similar
function after performing general configuration that purges only autocreated
peers that were created when persist mode isn't enabled.

(closes issue ASTERISK-20611)
Reported by: Alisher
Review: https://reviewboard.asterisk.org/r/2171/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375576 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-01 15:03:04 +00:00
Pedro Kiefer 8b34dc8192 Adds new formats to app_alarmreceiver, ALAW calls support and enhanced protection.
Commiting this on behalf of Kaloyan Kovachev (license 5506).
AlarmReceiver now supports the following DTMF signaling types:
 - ContactId
 - 4x1
 - 4x2
 - High Speed
 - Super Fast
We are also auto-detecting which signaling is being received. So support for
those protocols should work out-the-box. Correctly identify ALAW / ULAW calls.
Some enhanced protection for broken panels and malicious callers where added.

(closes issue ASTERISK-20289)
Reported by: Kaloyan Kovachev

Review: https://reviewboard.asterisk.org/r/2088/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375150 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-17 19:02:46 +00:00
Richard Mudgett 3e1d2917bb dahdi.conf.sample: Add description for "buffers" setting.
This contains an edited version of the patch originally created by John
Bigelow.

(closes issue ASTERISK-14435)
Reported by: John Bigelow
Patches:
      buffers.patch (license #5091) patch uploaded by John Bigelow
      0001-dahdi.conf.sample-Add-description-for-buffers-settin.patch (license #5417) patch uploaded by Shaun Ruffell
      Modified
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2012-10-08 22:31:09 +00:00
Matthew Jordan 3620fcff36 Disable ICE support by default
Since there are a number of legacy devices out there that fail to handle ICE
candidates properly (which is a nice way of saying something much uglier),
disable it by default.

Support for ICE candidates can be enabled in rtp.conf using the icesupport
setting.
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2012-10-08 20:39:26 +00:00
Alec L Davis 90f8c90b10 dsp.c User Configurable DTMF_HITS_TO_BEGIN and DTMF_MISSES_TO_END
Instead of a recompile, allow values to be adjusted in dsp.conf

For binary distributions allows easy adjustment for wobbly GSM calls, and other reasons.

Defaults to DTMF_HITS_TO_BEGIN=2 and DTMF_MISSES_TO_END=3

(closes issue ASTERISK-17493)
Reported by: alecdavis
Tested by: alecdavis
alecdavis (license 585)

Review https://reviewboard.asterisk.org/r/2144/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374493 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-04 20:21:36 +00:00
Alec L Davis 4af961a03a dsp.c User configuration of DTMF_NORMAL_TWIST and DTMF_REVERSE_TWIST values
Asterisk's DTMF Specifications are based on AT&T specs, which may not be compatible in other countries.

Various countries have different specifications for the maximum power level differences
between the DTMF low group and high group of frequencies.

Power level difference between frequencies for different Administrations/RPOAs
 NTT        = Max. 5 dB
 AT&T       = 4dB(reverse) to 8dB(normal)
 Danish     = Max. 6 dB
 Australian = Max. 10 dB
 Brazilian  = Max. 9 dB
 ETSI       = Max. 6 dB from ETSI ES 201 235-3 V1.3.1 (2006-03)

Now allow 4 variables to be individually configured in dsp.conf, with reasonable min/max of 2dB to 20dB.
Default is AT&T specifications

Add's the following variables to dsp.conf
;dtmf_normal_twist=6.31 
;dtmf_reverse_twist=2.51 
;relax_dtmf_normal_twist=6.31 
;relax_dtmf_reverse_twist=3.98 


(closes issue ASTERISK-20442)
Reported by: tbsky
Tested by: tbsky,alecdavis

alecdavis (license 585)

Review https://reviewboard.asterisk.org/r/2141/
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2012-10-04 04:50:16 +00:00
Mark Michelson 2b56626b43 Remove dead code and documentation for nonexistent feature.
multiplelogin was removed from chan_agent back in 1.6.0 when
AgentCallbackLogin() was removed.

(closes issue AST-948)
reported by Steve Pitts
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2012-09-25 22:57:56 +00:00
Terry Wilson b7233b18eb Properly handle UAC/UAS roles for SIP session timers
The SIP session timer mechanism contains a mandatory 'refresher' parameter
(included in the Session-Expires header) which is used in the session timer
offer/answer signaling within a SIP Invite dialog. It looks like asterisk is
interpreting the uac resp. uas role only as the initial role of client and
server (caller is uac, callee is uas). The standard rfc 4028 however assigns
the client role to the ((RE)-Invite) requester, the server role to the
((RE)-Invite) responder.

This patch has Asterisk track the actual refresher as "us" or "them" as opposed
to relying on just the configured "uas" or "uac" properties.

(closes issue AST-922)
Reported by: Thomas Airmont

Review: https://reviewboard.asterisk.org/r/2118/
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2012-09-25 19:08:02 +00:00