Asterisk 1.6.0 beta5

Tom Keating : VoIP & Gadgets Blog
Tom Keating
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Asterisk 1.6.0 beta5

AsteriskThe development team just released Asterisk 1.6.0-beta5. According to the announcement with beta5 of 1.6.0 the feature-set is frozen. One thing still missing is "caller name screening" where you can screen the call and accept/reject the call. I'd also like to see "voicemail call screening", which would allow me to screen a caller leaving a message and pull them out. Here's the announcement according to

In addition to a number of bug fixes, the following new features have been added since beta4:

  • The SMDI interface in Asterisk has been reworked to fix a number of issues as well as add some new features. SMDI message information is now accessed in the dialplan using some new dialplan functions. New options have been added to map Asterisk voicemail boxes to SMDI station IDs. Also, MWI will now properly be sent for systems that have some external interface modifying voicemail boxes, such as a web interface, or with an email client in the case of IMAP storage.
  • The Postgres CDR module now supports some of the features of cdr_adaptive_odbc. Specifically, you may add additional columns into the table and they will be set, if you set the corresponding CDR variable name. Also, if you omit columns in your database table, those fields will be silently skipped when inserting the record.
  • The ResetCDR application now has an 'e' option that re-enables the CDR if it has been disabled using the NoCDR option.
  • A new CLI command, "devstate change", has been added which allows you to change the state of a Custom device. Custom device states were previously only settable by using the DEVICE_STATE() dialplan function.
  • The Originate manager action now has its own permission level called originate. Also, if you want this action to be able to execute applications that call out to a subshell, it requires the system privilege, as well. These changes were made to enhance the security of the manager interface.

For a full list of features that have been introduced from Asterisk 1.4 to Asterisk 1.6.0, see the following file:

For a full list of changes to Asterisk 1.6.0 from beta4 to beta5, see the ChangeLog:

And if you're really curious, you can view the full feature-set changes after the jump...
--- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------

AMI - The manager (TCP/TLS/HTTP)
* Manager has undergone a lot of changes, all of them documented
in doc/manager_1_1.txt
* Manager version has changed to 1.1
* Added a new action 'CoreShowChannels' to list currently defined channels
and some information about them.
* Added a new action 'SIPshowregistry' to list SIP registrations.
* Added TLS support for the manager interface and HTTP server
* Added the URI redirect option for the built-in HTTP server
* The output of CallerID in Manager events is now more consistent.
CallerIDNum is used for number and CallerIDName for name.
* Enable https support for builtin web server.
See configs/http.conf.sample for details.
* Added a new action, GetConfigJSON, which can return the contents of an
Asterisk configuration file in JSON format. This is intended to help
improve the performance of AJAX applications using the manager interface
over HTTP.
* SIP and IAX manager events now use "ChannelType" in all cases where we
indicate channel driver. Previously, we used a mixture of "Channel"
and "ChannelDriver" headers.
* Added a "Bridge" action which allows you to bridge any two channels that
are currently active on the system.
* Added a "ListAllVoicemailUsers" action that allows you to get a list of all
the voicemail users setup.
* Added 'DBDel' and 'DBDelTree' manager commands.
* cdr_manager now reports events via the "cdr" level, separating it from
the very verbose "call" level.
* Manager users are now stored in memory. If you change the manager account
list (delete or add accounts) you need to reload manager.
* Added Masquerade manager event for when a masquerade happens between
two channels.
* Added "manager reload" command for the CLI
* Lots of commands that only provided information are now allowed under the
Reporting privilege, instead of only under Call or System.
* The IAX* commands now require either System or Reporting privilege, to
mirror the privileges of the SIP* commands.
* Added ability to retrieve list of categories in a config file.
* Added ability to retrieve the content of a particular category.
* Added ability to empty a context.
* Created new action to create a new file.
* Updated delete action to allow deletion by line number with respect to category.
* Added new action insert to add new variable to category at specified line.
* Updated action newcat to allow new category to be inserted in file above another
existing category.
* Added new event "JitterBufStats" in the IAX2 channel
* Originate now requires the Originate privilege and, if you want to call out
to a subshell, it requires the System privilege, as well. This was done to
enhance manager security.

Dialplan functions
* Added the DEVICE_STATE() dialplan function which allows retrieving any device
state in the dialplan, as well as creating custom device states that are
controllable from the dialplan.
* Extend CALLERID() function with "pres" and "ton" parameters to
fetch string representation of calling number presentation indicator
and numeric representation of type of calling number value.
* MailboxExists converted to dialplan function
* A new option to Dial() for telling IP phones not to count the call
as "missed" when dial times out and cancels.
* Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
held for any given channel. Also, locks are automatically freed when a
channel is hung up.
* Added HINT() dialplan function that allows retrieving hint information.
Hints are mappings between extensions and devices for the sake of
determining the state of an extension. This function can retrieve the list
of devices or the name associated with a hint.
* Added EXTENSION_STATE() dialplan function which allows retrieving the state
of any extension.
* Added SYSINFO() dialplan function which allows retrieval of system information
* Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
the existence of a dialplan target.
* Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
upper and lower case, respectively.
* When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
ID for the call (not the Asterisk call ID or unique ID), provided that the
channel driver supports this. For SIP, you get the SIP call-ID for the
bridged channel which you can store in the CDR with a custom field.

CLI Changes
* New CLI command "core show hint" (usage: core show hint <exten>)
* New CLI command "core show settings"
* Added 'core show channels count' CLI command.
* Added the ability to set the core debug and verbose values on a per-file basis.
* Added 'queue pause member' and 'queue unpause member' CLI commands
* Ability to set process limits ("ulimit") without restarting Asterisk
* Enhanced "agi debug" to print the channel name as a prefix to the debug
output to make debugging on busy systems much easier.
* New CLI commands "dialplan set extenpatternmatching true/false"
* New CLI command: "core set chanvar" to set a channel variable from the CLI.
* Added an easy way to execute Asterisk CLI commands at startup. Any commands
listed in the startup_commands section of cli.conf will get executed.
* Added a CLI command, "devstate change", which allows you to set custom device
states from the func_devstate module that provides the DEVICE_STATE() function
and handling of the "Custom:" devices.

SIP changes
* Improved NAT and STUN support.
chan_sip now can use port numbers in bindaddr, externip and externhost
options, as well as contact a STUN server to detect its external address
for the SIP socket. See sip.conf.sample, 'NAT' section.
* The default SIP useragent= identifier now includes the Asterisk version
* A new option, match_auth_username in sip.conf changes the matching of incoming requests.
If set, and the incoming request carries authentication info,
the username to match in the users list is taken from the Digest header
rather than from the From: field. This feature is considered experimental.
* The "musiconhold" and "musicclass" settings in sip.conf are now removed,
since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
* The "localmask" setting was removed in version 1.2 and the reminder about it
being removed is now also removed.
* A new option "busylevel" for setting a level of calls where asterisk reports
a device as busy, to separate it from call-limit. This value is also added
to the SIP_PEER dialplan function.
* A new realtime family called "sipregs" is now supported to store SIP registration
data. If this family is defined, "sippeers" will be used for configuration and
"sipregs" for registrations. If it's not defined, "sippeers" will be used for
registration data, as before.
* The SIPPEER function have new options for port address, call and pickup groups
* Added support for T.140 realtime text in SIP/RTP
* The "checkmwi" option has been removed from sip.conf, as it is no longer
required due to the restructuring of how MWI is handled. See the descriptions
in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
for more information.
* Added rtpdest option to CHANNEL() dialplan function.
* Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
* SIP now adds a header to the CANCEL if the call was answered by another phone
in the same dial command, or if the new c option in dial() is used.
* The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
states it is not needed. For phones, however, that do require it the "registertrying" option
has been added so it can be enabled.
* A new option called "callcounter" (global/peer/user level) enables call counters needed
for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
used to enable this functionality).
* New settings for timer T1 and timer B on a global level or per device. This makes it
possible to force timeout faster on non-responsive SIP servers. These settings are
considered advanced, so don't use them unless you have a problem.
* Added a dial string option to be able to set the To: header in an INVITE to any
SIP uri.
* Added a new global and per-peer option, qualifyfreq, which allows you to configure
the qualify frequency.
* Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
were not properly torn down due to network or endpoint failures during an established
SIP session.
* Added TCP and TLS support for SIP. See doc/siptls.txt and configs/sip.conf.sample for
more information on how it is used.

IAX2 changes
* Added the trunkmaxsize configuration option to chan_iax2.
* Added the srvlookup option to iax.conf
* Added support for OSP. The token is set and retrieved through the CHANNEL()
dialplan function.

XMPP Google Talk/Jingle changes
* Added the bindaddr option to gtalk.conf.

Skinny changes
* Added skinny show device, skinny show line, and skinny show settings CLI commands.
* Proper codec support in chan_skinny.
* Added settings for IP and Ethernet QoS requests

* Added separate settings for media QoS in mgcp.conf

Console Channel Driver changes
* Added experimental support for video send & receive to chan_oss.
This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
a video source.

Phone channel changes (chan_phone)
* Added G729 passthrough support to chan_phone for Sigma Designs boards.

H.323 channel Changes
* H323 remote hold notification support added (by NOTIFY message
and/or H.450 supplementary service)

Local channel changes
* The device state functionality in the Local channel driver has been updated
to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
to just UNKNOWN if the extension exists.
* Added jitterbuffer support for chan_local. This allows you to use the
generic jitterbuffer on incoming calls going to Asterisk applications.
For example, this would allow you to use a jitterbuffer for an incoming
SIP call to Voicemail by putting a Local channel in the middle. This
feature is enabled by using the 'j' option in the Dial string to the Local
channel in conjunction with the existing 'n' option for local channels.

Zaptel channel driver (chan_zap) Changes
* SS7 support in chan_zap (via libss7 library)
* In India, some carriers transmit CID via dtmf. Some code has been added
that will handle some situations. The cidstart=polarity_IN choice has been added for
those carriers that transmit CID via dtmf after a polarity change.
* CID matching information is now shown when doing 'dialplan show'.
* Added zap show version CLI command to chan_zap.
* Added setvar support to zapata.conf channel entries.
* Added two new options: mwimonitor and mwimonitornotify. These options allow
you to enable MWI monitoring on FXO lines. When the MWI state changes,
the script specified in the mwimonitornotify option is executed. An internal
event indicating the new state of the mailbox is also generated, so that
the normal MWI facilities in Asterisk work as usual.
* Added signalling type 'auto', which attempts to use the same signalling type
for a channel as configured in Zaptel. This is primarily designed for analog
ports, but will also work for digital ports that are configured for FXS or FXO
signalling types. This mode is also the default now, so if your zapata.conf
does not specify signalling for a channel (which is unlikely as the sample
configuration file has always recommended specifying it for every channel) then
the 'auto' mode will be used for that channel if possible.
* Added a 'zap set dnd' command to allow CLI control of the Do-Not-Disturb
state for a channel; also ensured that the DNDState Manager event is
emitted no matter how the DND state is set or cleared.

New Channel Drivers
* Added a new channel driver, chan_unistim. See doc/unistim.txt and
configs/unistim.conf.sample for details. This new channel driver allows
you to use Nortel i2002, i2004, and i2050 phones with Asterisk.
* Added a new channel driver, chan_console, which uses portaudio as a cross
platform audio interface. It was written as a channel driver that would
work with Mac CoreAudio, but portaudio supports a number of other audio
interfaces, as well. Note that this channel driver requires v19 or higher
of portaudio; older versions have a different API.

DUNDi changes
* Added the ability to specify arguments to the Dial application when using
the DUNDi switch in the dialplan.
* Added the ability to set weights for responses dynamically. This can be
done using a global variable or a dialplan function. Using the SHELL()
function would allow you to have an external script set the weight for
each response.
* Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
functions will allow you to initiate a DUNDi query from the dialplan,
find out how many results there are, and access each one.

ENUM changes
* Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
functions will allow you to initiate an ENUM lookup from the dialplan,
and Asterisk will cache the results. ENUMRESULT can be used to access
the results without doing multiple DNS queries.

Voicemail Changes
* Added the ability to customize which sound files are used for some of the
prompts within the Voicemail application by changing them in voicemail.conf
* Added the ability for the "voicemail show users" CLI command to show users
configured by the dynamic realtime configuration method.
* MWI (Message Waiting Indication) handling has been significantly
restructured internally to Asterisk. It is now totally event based
instead of polling based. The voicemail application will notify other
modules that have subscribed to MWI events when something in the mailbox
This also means that if any other entity outside of Asterisk is changing
the contents of mailboxes, then the voicemail application still needs to
poll for changes. Examples of situations that would require this option
are web interfaces to voicemail or an email client in the case of using
IMAP storage. So, two new options have been added to voicemail.conf
to account for this: "pollmailboxes" and "pollfreq". See the sample
configuration file for details.
* Added "tw" language support
* Added support for storage of greetings using an IMAP server
* Added ability to customize forward, reverse, stop, and pause keys for message playback
* SMDI is now enabled in voicemail using the smdienable option.
* A "lockmode" option has been added to asterisk.conf to configure the file
locking method used for voicemail, and potentially other things in the
future. The default is the old behavior, lockfile. However, there is a
new method, "flock", that uses a different method for situations where the
lockfile will not work, such as on SMB/CIFS mounts.
* Added the ability to backup deleted messages, to ease recovery in the case
that a user accidentally deletes a message, and discovers that they need it.
* Reworked the SMDI interface in Asterisk. The new way to access SMDI information
is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
voicemail boxes. The SMDI interface can also poll for MWI changes when some
outside entity is modifying the state of the mailbox (such as IMAP storage or
a web interface of some kind).

Queue changes
* Added the general option 'shared_lastcall' so that member's wrapuptime may be
used across multiple queues.
* Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
setqueueentryvar options for each queue, see queues.conf.sample for details.
* Added keepstats option to queues.conf which will keep queue
statistics during a reload.
* setinterfacevar option in queues.conf also now sets a variable
called MEMBERNAME which contains the member's name.
* Added 'Strategy' field to manager event QueueParams which represents
the queue strategy in use.
* Added option to run macro when a queue member is connected to a caller,
see queues.conf.sample for details.
* app_queue now has a 'loose' option which is almost exactly like 'strict' except it
does not count paused queue members as unavailable.
* Added min-announce-frequency option to queues.conf which allows you to control the
minimum amount of time between queue announcements for use when the caller's queue
position changes frequently.
* Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
queue log.
* Added ability for non-realtime queues to have realtime members
* Added the "linear" strategy to queues.
* Added the "wrandom" strategy to queues.
* Added new channel variable QUEUE_MIN_PENALTY
* QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
rules in queuerules.conf. See configs/queuerules.conf.sample for details
* Added a new parameter for member definition, called state_interface. This may be
used so that a member may be called via one interface but have a different interface's
device state reported.

MeetMe Changes
* The 'o' option to provide an optimization has been removed and its functionality
has been enabled by default.
* When a conference is created, the UNIQUEID of the channel that caused it to be
created is stored. Then, every channel that joins the conference will have the
MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
callers that come and go from long standing conferences.
* Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
except it does operations on a channel by name, instead of number in a conference.
This is a very useful feature in combination with the 'X' option to ChanSpy.
* Added 'C' option to Meetme which causes a caller to continue in the dialplan
when kicked out.
* Added new RealTime functionality to provide support for scheduled conferencing.
This includes optional messages to the caller if they attempt to join before
the schedule start time, or to allow the caller to join the conference early.
Also included is optional support for limiting the number of callers per
RealTime conference.
* Added the S() and L() options to the MeetMe application. These are pretty
much identical to the S() and L() options to Dial(). They let you set
timeouts for the conference, as well as have warning sounds played to
let the caller know how much time is left, and when it is running out.
* Added the ability to do "meetme concise" with the "meetme" CLI command.
This extends the concise capabilities of this CLI command to include
listing all conferences, instead of an addition to the other sub commands
for the "meetme" command.
* Added the ability to specify the music on hold class used to play into the
conference when there is only one member and the M option is used.

Other Dialplan Application Changes
* Argument support for Gosub application
* From the to-do lists: straighten out the app timeout args:
Wait() app now really does 0.3 seconds- was truncating arg to an int.
WaitExten() same as Wait().
Congestion() - Now takes floating pt. argument.
Busy() - now takes floating pt. argument.
Read() - timeout now can be floating pt.
WaitForRing() now takes floating pt timeout arg.
SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
* Added 's' option to Page application.
* Added 'E' and 'V' commands to ExternalIVR.
* Added 'o' and 'X' options to Chanspy.
* Added a new dialplan application, Bridge, which allows you to bridge the
calling channel to any other active channel on the system.
* Added the ability to specify a music on hold class to play instead of ringing
for the SLATrunk application.
* The Read application no longer exits the dialplan on error. Instead, it sets
READSTATUS to ERROR, which you can catch and handle separately.
* Added 'm' option to Directory, which lists out names, 8 at a time, instead
of asking for verification of each name, one at a time.
* Privacy() no longer uses privacy.conf, as all options are specifyable as
direct options to the app.
* AMD() has a new "maximum word length" option. "show application AMD" from the CLI
for more details
* GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
* The ChannelRedirect application no longer exits the dialplan if the given channel
does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
or NOCHANNEL if the given channel was not found.

Music On Hold Changes
* A new option, "digit", has been added for music on hold classes in
musiconhold.conf. If this is set for a music on hold class, a caller
listening to music on hold can press this digit to switch to listening
to this music on hold class.
* Support for realtime music on hold has been added.
* In conjunction with the realtime music on hold, a general section has
been added to musiconhold.conf, its sole variable is cachertclasses. If this
is set, then music on hold classes found in realtime will be cached in memory.

AEL Changes
* AEL upgraded to use the Gosub with Arguments instead
of Macro application, to hopefully reduce the problems
seen with the artificially low stack ceiling that
Macro bumps into. Macros can only call other Macros
to a depth of 7. Tests run using gosub, show depths
limited only by virtual memory. A small test demonstrated
recursive call depths of 100,000 without problems.
-- in addition to this, all apps that allowed a macro
to be called, as in Dial, queues, etc, are now allowing
a gosub call in similar fashion.
* AEL now generates LOCAL(argname) declarations when it
Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
etc. That makes the arguments local in scope. The user
can define their own local variables in macros, now,
by saying "local myvar=someval;" or using Set() in this
fashion: Set(LOCAL(myvar)=someval); ("local" is now
an AEL keyword).
* utils/conf2ael introduced. Will convert an extensions.conf
file into extensions.ael. Very crude and unfinished, but
will be improved as time goes by. Should be useful for a
first pass at conversion.
* aelparse will now read extensions.conf to see if a referenced
macro or context is there before issueing a warning.

Call Features (res_features) Changes
* Added the parkedcalltransfers option to features.conf
* The built-in method for doing attended transfers has been updated to
include some new options that allow you to have the transferee sent
back to the person that did the transfer if the transfer is not successful.
See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
in features.conf.sample.
* Added support for configuring named groups of custom call features in
features.conf. This means that features can be written a single time, and
then mapped into groups of features for different key mappings or easier
access control.
* Updated the ParkedCall application to allow you to not specify a parking
extension. If you don't specify a parking space to pick up, it will grab
the first one available.
* Added cli command 'features reload' to reload call features from features.conf
* Moved into core asterisk binary.

Language Support Changes
* Brazilian Portuguese (pt-BR) in VM, and say.c was added
* Added support for the Hungarian language for saying numbers, dates, and times.

AGI Changes
* Added SPEECH commands for speech recognition. A complete listing can be found
using agi show.

Logger changes
* Added rotatestrategy option to logger.conf, along with two new options:
"timestamp" which will use the time to name the logger files instead of
sequence number; and "rotate", which rotates the names of the logfiles,
similar to the way syslog rotates files.
* Added exec_after_rotate option to logger.conf, which allows a system
command to be run after rotation. This is primarily useful with
rotatestrategry=rotate, to allow a limit on the number of logfiles kept
and to ensure that the oldest log file gets deleted.
* Added realtime support for the queue log

Call Detail Records
* The cdr_manager module has a [mappings] feature, like cdr_custom,
to add fields to the manager event from the CDR variables.
* Added cdr_adaptive_odbc, a new module that adapts to the structure of your
backend database CDR table. Specifically, additional, non-standard
columns are supported, merely by setting the corresponding CDR variable in
your dialplan. In addition, you may alias any column to another name (for
example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
simply "alias src => ANI" in the configuration file). Records may be
posted to more than one backend, simply by specifying multiple categories
in the configuration file. And finally, you may filter which CDRs get
posted to each backend, by specifying a filter (which the record must
match) for the particular category. Filters are additive (meaning all
rules must match to post that CDR).
* The Postgres CDR module now supports some features of the cdr_adaptive_odbc
module. Specifically, you may add additional columns into the table and
they will be set, if you set the corresponding CDR variable name. Also,
if you omit columns in your database table, they will be silently skipped
(but a record will still be inserted, based on what columns remain). Note
that the other two features from cdr_adaptive_odbc (alias and filter) are
not currently supported.
* The ResetCDR application now has an 'e' option that re-enables a CDR if it
has been disabled using the NoCDR application.

Miscellaneous New Modules
* Added a new CDR module, cdr_sqlite3_custom.
* Added a new realtime configuration module, res_config_sqlite
* Added a new codec translation module, codec_resample, which re-samples
signed linear audio between 8 kHz and 16 kHz to help support wideband
* Added a new module, res_phoneprov, which allows auto-provisioning of phones
based on configuration templates that use Asterisk dialplan function and
variable substitution. It should be possible to create phone profiles and
templates that work for the majority of phones provisioned over http. It
is currently only intended to provision a single user account per phone.
An example profile and set of templates for Polycom phones is provided.
NOTE: Polycom firmware is not included, but should be placed in
AST_DATA_DIR/phoneprov/configs to match up with the included templates.
* Added a new module, app_jack, which provides interfaces to JACK, the Jack
Audio Connection Kit ( Two interfaces are
provided; there is a JACK() application, and a JACK_HOOK() function. Both
interfaces create an input and output JACK port. The application makes
these ports the endpoint of the call. The audio coming from the channel
goes out the output port and whatever comes back in on the input port is
what gets sent to the channel. The JACK_HOOK() function turns on a JACK
audiohook on the channel. This lets you run the audio coming from a
channel through JACK, and whatever comes back in is what gets forwarded
on as the channel's audio. This is very useful for building custom
vocoders or doing recording or analysis of the channel's audio in another
* Added a new module, res_config_curl, which permits using a HTTP POST url
to retrieve, create, update, and delete realtime information from a remote
web server. Note that this module requires to be loaded for
backend functionality.
* Added a new module, res_config_ldap, which permits the use of an LDAP
server for realtime data access.
* Added support for writing and running your dialplan in lua using the pbx_lua
module. See configs/extensions.lua.sample for examples of how to do this.

* Ability to use libcap to set high ToS bits when non-root
on Linux. If configure is unable to find libcap then you
can use --with-cap to specify the path.
* Added maxfiles option to options section of asterisk.conf which allows you to specify
what Asterisk should set as the maximum number of open files when it loads.
* Added the jittertargetextra configuration option.
* Added support for setting the CoS for VLAN traffic (802.1p). See the sample
configuration files for the IP channel drivers. The new option is "cos".
This information is also documented in doc/qos.tex, or the IP Quality of Service
section of asterisk.pdf.
* When originating a call using AMI or pbx_spool that fails the reason for failure
will now be available in the failed extension using the REASON dialplan variable.
* Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
It allows you to configure a prefix for auto-monitor recordings.
* A new extension pattern matching algorithm, based on a trie, is introduced
here, that could noticeably speed up mid-sized to large dialplans.
It is NOT used by default, as duplicating the behaviour of the old pattern
matcher is still under development. A config file option, in extensions.conf,
in the [general] section, called "extenpatternmatchingnew", is by default
set to false; setting that to true will force the use of the new algorithm.
Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
be used to switch the algorithms at run time.
* A new option when starting a remote asterisk (rasterisk, asterisk -r) for
specifying which socket to use to connect to the running Asterisk daemon
* Added logging to 'make update' command. See update.log
* Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
do not come from the remote party.
* Added the 'n' option to the SpeechBackground application to tell it to not
answer the channel if it has not already been answered.
* Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
turned on, via the CHANNEL(trace) dialplan function. Could be useful for
dialplan debugging.

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