If ignore_failed_channels is set to "true" for a channel, the channel
will continue to be configured even if configuring it has failed.
This allows Asterisk to start before all the DAHDI initialization is
done and thus not force the starting order dahdi -> asterisk.
Review: https://reviewboard.asterisk.org/r/3063/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404542 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Added another NAT example to pjsip.conf.sample. We had a few mentions of NAT configuration throughout the sample, but I added another for a little bit more clarity.
Additionally many pjsip options were affected by the change to snake case, so I fixed any instances of those options in pjsip.conf.
I regenerated the config option list (at the bottom of the file) from a new xml config doc dump, so all the snake case changes should be reflected there, as well as any other changes to those options.
(issue ASTERISK-23004)
(closes issue ASTERISK-23004)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3086/
........
Merged revisions 404405 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404406 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change is in preparation for external MWI support.
Removed code from the system for normal mailbox handling that appends
@default to the mailbox identifier if it does not have a context. The
only exception is the legacy hasvoicemail users.conf option. The legacy
option will only work for app_voicemail mailboxes. The system cannot make
any assumptions about the format of the mailbox identifer used by
app_voicemail.
chan_sip and chan_dahdi/sig_pri had the most changes because they both
tried to interpret the mailbox identifier. chan_sip just stored and
compared the two components. chan_dahdi actually used the box
information.
The ISDN MWI support configuration options had to be reworked because
chan_dahdi was parsing the box@context format to get the box number. As a
result the mwi_vm_boxes chan_dahdi.conf option was added and is documented
in the chan_dahdi.conf.sample file.
Review: https://reviewboard.asterisk.org/r/3072/
........
Merged revisions 404348 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404350 65c4cc65-6c06-0410-ace0-fbb531ad65f3
on incoming h.323 calls. Option can be set globally or per user section.
(closes issue ASTERISK-22020)
Reported by: Ross Beer
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404211 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch allows individual dialplan functions to be marked as
'dangerous', to inhibit their execution from external sources.
A 'dangerous' function is one which results in a privilege escalation.
For example, if one were to read the channel variable SHELL(rm -rf /)
Bad Things(TM) could happen; even if the external source has only read
permissions.
Execution from external sources may be enabled by setting
'live_dangerously' to 'yes' in the [options] section of asterisk.conf.
Although doing so is not recommended.
Also, the ABI was changed to something more reasonable, since Asterisk
12 does not yet have a public release.
(closes issue ASTERISK-22905)
Review: http://reviewboard.digium.internal/r/432/
........
Merged revisions 403913 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 403917 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 403959 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403960 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In some cases messages need to be sent to a direct URI (sip:<ip address>). This
patch adds in that support by using a default outbound endpoint. When sending
messages, if no endpoint can be found then the default one is used.
To facilitate this a new default_outbound_endpoint option was added to the
globals section for pjsip.conf.
Review: https://reviewboard.asterisk.org/r/2944/
........
Merged revisions 403680 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403687 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ConfBridge allows custom DTMF menus to be created in the confbridge.conf
file by assigning a DTMF key sequence to a sequence of actions as follows:
DTMF-sequence = action,action...
Unfortunately, the normal config file processing code interprets an
initial '#' character as starting a directive such as #include.
* Add the ability to escape the first non-blank character in a config line
so the '#' character can be used without triggering the directive
processing code.
(closes issue AFS-2)
(closes issue ASTERISK-22478)
Reported by: Nicolas Tanski
Patches:
jira_asterisk_22478_v11.patch (license #5621) patch uploaded by rmudgett (modified)
Review: https://reviewboard.asterisk.org/r/2969/
........
Merged revisions 402407 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 402416 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402417 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When reworking chan_iax2 for IPv6, the ability to bind to multiple addresses
was removed by mistake. This patch restores this functionality and adds notes
about IPv6 addresses in the sample config.
(closes issue ASTERISK-22741)
Reported by: Joshua Colp
Tested by: Michael L. Young
Patches:
asterisk-22741-fix-binding-multiple-addr.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2945/
........
Merged revisions 401488 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401489 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ConfBridge now has the ability to set the language of announcements to the
conference. The language can be set on a bridge profile in
confbridge.conf or by the dialplan function
CONFBRIDGE(bridge,language)=en.
(closes issue ASTERISK-19983)
Reported by: Jonathan White
Patches:
M19983_rev2.diff (license #5138) patch uploaded by junky (modified)
Tested by: rmudgett
........
Merged revisions 400741 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 400742 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400744 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r399887 | dlee | 2013-09-26 10:41:47 -0500 (Thu, 26 Sep 2013) | 1 line
Minor performance bump by not allocate manager variable struct if we don't need it
........
r400138 | dlee | 2013-09-30 10:24:00 -0500 (Mon, 30 Sep 2013) | 23 lines
Stasis performance improvements
This patch addresses several performance problems that were found in
the initial performance testing of Asterisk 12.
The Stasis dispatch object was allocated as an AO2 object, even though
it has a very confined lifecycle. This was replaced with a straight
ast_malloc().
The Stasis message router was spending an inordinate amount of time
searching hash tables. In this case, most of our routers had 6 or
fewer routes in them to begin with. This was replaced with an array
that's searched linearly for the route.
We more heavily rely on AO2 objects in Asterisk 12, and the memset()
in ao2_ref() actually became noticeable on the profile. This was
#ifdef'ed to only run when AO2_DEBUG was enabled.
After being misled by an erroneous comment in taskprocessor.c during
profiling, the wrong comment was removed.
Review: https://reviewboard.asterisk.org/r/2873/
........
r400178 | dlee | 2013-09-30 13:26:27 -0500 (Mon, 30 Sep 2013) | 24 lines
Taskprocessor optimization; switch Stasis to use taskprocessors
This patch optimizes taskprocessor to use a semaphore for signaling,
which the OS can do a better job at managing contention and waiting
that we can with a mutex and condition.
The taskprocessor execution was also slightly optimized to reduce the
number of locks taken.
The only observable difference in the taskprocessor implementation is
that when the final reference to the taskprocessor goes away, it will
execute all tasks to completion instead of discarding the unexecuted
tasks.
For systems where unnamed semaphores are not supported, a really
simple semaphore implementation is provided. (Which gives identical
performance as the original taskprocessor implementation).
The way we ended up implementing Stasis caused the threadpool to be a
burden instead of a boost to performance. This was switched to just
use taskprocessors directly for subscriptions.
Review: https://reviewboard.asterisk.org/r/2881/
........
r400180 | dlee | 2013-09-30 13:39:34 -0500 (Mon, 30 Sep 2013) | 28 lines
Optimize how Stasis forwards are dispatched
This patch optimizes how forwards are dispatched in Stasis.
Originally, forwards were dispatched as subscriptions that are invoked
on the publishing thread. This did not account for the vast number of
forwards we would end up having in the system, and the amount of work it
would take to walk though the forward subscriptions.
This patch modifies Stasis so that rather than walking the tree of
forwards on every dispatch, when forwards and subscriptions are changed,
the subscriber list for every topic in the tree is changed.
This has a couple of benefits. First, this reduces the workload of
dispatching messages. It also reduces contention when dispatching to
different topics that happen to forward to the same aggregation topic
(as happens with all of the channel, bridge and endpoint topics).
Since forwards are no longer subscriptions, the bulk of this patch is
simply changing stasis_subscription objects to stasis_forward objects
(which, admittedly, I should have done in the first place.)
Since this required me to yet again put in a growing array, I finally
abstracted that out into a set of ast_vector macros in
asterisk/vector.h.
Review: https://reviewboard.asterisk.org/r/2883/
........
r400181 | dlee | 2013-09-30 13:48:57 -0500 (Mon, 30 Sep 2013) | 28 lines
Remove dispatch object allocation from Stasis publishing
While looking for areas for performance improvement, I realized that an
unused feature in Stasis was negatively impacting performance.
When a message is sent to a subscriber, a dispatch object is allocated
for the dispatch, containing the topic the message was published to, the
subscriber the message is being sent to, and the message itself.
The topic is actually unused by any subscriber in Asterisk today. And
the subscriber is associated with the taskprocessor the message is being
dispatched to.
First, this patch removes the unused topic parameter from Stasis
subscription callbacks.
Second, this patch introduces the concept of taskprocessor local data,
data that may be set on a taskprocessor and provided along with the data
pointer when a task is pushed using the ast_taskprocessor_push_local()
call. This allows the task to have both data specific to that
taskprocessor, in addition to data specific to that invocation.
With those two changes, the dispatch object can be removed completely,
and the message is simply refcounted and sent directly to the
taskprocessor.
Review: https://reviewboard.asterisk.org/r/2884/
........
Merged revisions 399887,400138,400178,400180-400181 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400186 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The patch from ASTERISK-21965 was committed perhaps a bit too hastily. Walter
and Tzafrir have pointed out numerous issues with the approach and have
propsed an alternative in r/2757. Since it's not a time critical issue and
is not worth holding up the release of 12 for it, I've gone ahead and reverted
r394939 from 12/trunk and re-opened ASTERISK-21965.
........
Merged revisions 397938 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The general gist is to have a clear boundary between old SIP stuff
and new SIP stuff by having the word "SIP" for old stuff and "PJSIP"
for new stuff. Here's a brief rundown of the changes:
* The word "Gulp" in dialstrings, functions, and CLI commands is now
"PJSIP"
* chan_gulp.c is now chan_pjsip.c
* Function names in chan_gulp.c that were "gulp_*" are now "chan_pjsip_*"
* All files that were "res_sip*" are now "res_pjsip*"
* The "res_sip" directory is now "res_pjsip"
* Files in the "res_pjsip" directory that began with "sip_*" are now "pjsip_*"
* The configuration file is now "pjsip.conf" instead of "res_sip.conf"
* The module info for all PJSIP-related files now uses "PJSIP" instead of "SIP"
* CLI and AMI commands created by Asterisk's PJSIP modules now have "pjsip" as
the starting word instead of "sip"
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395764 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This renames all files and API calls from several variants of
Stasis-HTTP to ARI including:
* Stasis-HTTP -> ARI
* STASIS_HTTP -> ARI
* stasis_http -> ari (ast_ari for global symbols, file names as well)
* stasis http -> ARI
Review: https://reviewboard.asterisk.org/r/2706/
(closes issue ASTERISK-22136)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395603 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
........
Merged revisions 394940 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
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
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
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
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
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/
........
Merged revisions 393928 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
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
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
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
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
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
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
........
Merged revisions 391215 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 391241 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391245 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer
channel driver specific. If the channel variable is set on the
transferrer channel, the sound will be played to the target of an attended
transfer.
* The channel variable BRIDGEPEER becomes a comma separated list of peers
in a multi-party bridge. The BRIDGEPEER value can have a maximum of 10
peers listed. Any more peers in the bridge will not be included in the
list. BRIDGEPEER is not valid in holding bridges like parking since those
channels do not talk to each other even though they are in a bridge.
* The channel variable BRIDGEPVTCALLID is only valid for two party bridges
and will contain a value if the BRIDGEPEER's channel driver supports it.
* The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and
is removed. The more useful DYNAMIC_WHO_ACTIVATED gives the channel name
that activated the dynamic feature.
* The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are
set only on the channel executing the dynamic feature. Executing a
dynamic feature on the bridge peer in a multi-party bridge will execute it
on all peers of the activating channel.
(closes issue ASTERISK-21555)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2582/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390771 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
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
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
