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<h2><a href="https://wiki.asterisk.org/wiki/display/AST/New+in+11">New in 11</a></h2>
<h4>Page <b>edited</b> by <a href="https://wiki.asterisk.org/wiki/display/~mjordan">Matt Jordan</a>
</h4>
<br/>
<h4>Changes (1)</h4>
<div id="page-diffs">
<table class="diff" cellpadding="0" cellspacing="0">
<tr><td class="diff-snipped" >...<br></td></tr>
<tr><td class="diff-unchanged" > longer necessray to control the order that the 'permit' and 'deny' columns are <br> returned from queries. <br></td></tr>
<tr><td class="diff-changed-lines" >* DUNDi now allows the built in variables <span class="diff-deleted-words"style="color:#999;background-color:#fdd;text-decoration:line-through;">${NUMBER}, ${IPADDR}</span> <span class="diff-added-words"style="background-color: #dfd;">$\{NUMBER\}, $\{IPADDR\}</span> and <span class="diff-changed-words">$<span class="diff-added-chars"style="background-color: #dfd;">\</span>{SECRET<span class="diff-added-chars"style="background-color: #dfd;">\</span>}</span> to <br></td></tr>
<tr><td class="diff-unchanged" > be used within the dynamic weight attribute when specifying a mapping. <br> * CEL backends can now be configured to show "USER_DEFINED" in the EventName <br></td></tr>
<tr><td class="diff-snipped" >...<br></td></tr>
</table>
</div> <h4>Full Content</h4>
<div class="notificationGreySide">
<h1><a name="Newin11-BuildSystem"></a>Build System</h1>
<ul>
<li>The Asterisk build system will now build and install a shared library<br/>
(libasteriskssl.so) used to wrap various initialization and shutdown functions<br/>
from the libssl and libcrypto libraries provided by OpenSSL. This is done so<br/>
that Asterisk can ensure that these functions do <b>not</b> get called by any<br/>
modules that are loaded into Asterisk, since they should only be called once<br/>
in any single process. If desired, this feature can be disabled by supplying<br/>
the "--disable-asteriskssl" option to the configure script.</li>
<li>A new make target, 'full', has been added to the Makefile. This performs<br/>
the same compilation actions as make all, but will also scan the entirety of<br/>
each source file for documentation. This option is needed to generate AMI<br/>
event documentation. Note that your system must have Python in order for<br/>
this make target to succeed.</li>
<li>The optimization portion of the build system has been reworked to avoid<br/>
broken builds on certain architectures. All architecture-specific<br/>
optimization has been removed in favor of using -march=native to allow gcc<br/>
to detect the environment in which it is running when possible. This can<br/>
be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.</li>
<li>BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,<br/>
make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"</li>
<li>Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you<br/>
previously parsed the header file to obtain the version of Asterisk, you<br/>
will now have to go through Asterisk to get the version information.</li>
</ul>
<h1><a name="Newin11-Applications"></a>Applications</h1>
<h2><a name="Newin11-Bridge"></a>Bridge</h2>
<ul>
<li>Added 'F()' option. Similar to the dial option, this can be supplied with<br/>
arguments indicating where the callee should go after the caller is hung up,<br/>
or without options specified, the priority after the Queue will be used.</li>
</ul>
<h2><a name="Newin11-ConfBridge"></a>ConfBridge</h2>
<ul>
<li>Added menu action admin_toggle_mute_participants. This will mute / unmute<br/>
all non-admin participants on a conference. The confbridge configuration<br/>
file also allows for the default sounds played to all conference users when<br/>
this occurs to be overriden using sound_participants_unmuted and<br/>
sound_participants_muted.</li>
<li>Added menu action participant_count. This will playback the number of<br/>
current participants in a conference.</li>
<li>Added announcement configuration option to user profile. If set the sound<br/>
file will be played to the user, and only the user, upon joining the<br/>
conference bridge.</li>
</ul>
<h2><a name="Newin11-Dial"></a>Dial</h2>
<ul>
<li>Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller<br/>
channels respectively before the callee channels are called.</li>
</ul>
<h2><a name="Newin11-ExternalIVR"></a>ExternalIVR</h2>
<ul>
<li>Added support for IPv6.</li>
<li>Add interrupt ('I') command to ExternalIVR. Sending this command from an<br/>
external process will cause the current playlist to be cleared, including<br/>
stopping any audio file that is currently playing. This is useful when you<br/>
want to interrupt audio playback only when specific DTMF is entered by the<br/>
caller.</li>
</ul>
<h2><a name="Newin11-FollowMe"></a>FollowMe</h2>
<ul>
<li>A new option, 'I' has been added to app_followme. By setting this option,<br/>
Asterisk will not update the caller with connected line changes when they<br/>
occur. This is similar to app_dial and app_queue.</li>
<li>The 'N' option is now ignored if the call is already answered.</li>
<li>Added 'b' and 'B' options to FollowMe that execute a Gosub on callee<br/>
and caller channels respectively before the callee channels are called.</li>
<li>The winning FollowMe outgoing call is now put on hold if the caller put it on<br/>
hold.</li>
</ul>
<h2><a name="Newin11-MixMonitor"></a>MixMonitor</h2>
<ul>
<li>MixMonitor hooks now have IDs associated with them which can be used to<br/>
assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option<br/>
will allow storage of the MixMontior ID in a channel variable. StopMixmonitor<br/>
now accepts that ID as an argument.</li>
<li>Added 'm' option, which stores a copy of the recording as a voicemail in the<br/>
indicated mailboxes.</li>
</ul>
<h2><a name="Newin11-MySQL"></a>MySQL</h2>
<ul>
<li>The connect action in app_mysql now allows you to specify a port number to<br/>
connect to. This is useful if you run a MySQL server on a non-standard<br/>
port number.</li>
</ul>
<h2><a name="Newin11-OSPApplications"></a>OSP Applications</h2>
<ul>
<li>Increased the default number of allowed destinations from 5 to 12.</li>
</ul>
<h2><a name="Newin11-Page"></a>Page</h2>
<ul>
<li>The app_page application now no longer depends on DAHDI or app_meetme. It<br/>
has been re-architected to use app_confbridge internally.</li>
</ul>
<h2><a name="Newin11-Queue"></a>Queue</h2>
<ul>
<li>Added queue options autopausebusy and autopauseunavail for automatically<br/>
pausing a queue member when their device reports busy or congestion.</li>
<li>The 'ignorebusy' option for queue members has been deprecated in favor of<br/>
the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been<br/>
added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a<br/>
per interface basis. Individual ringinuse values can now be set in<br/>
queues.conf via an argument to member definitions. Lastly, the queue<br/>
'ringinuse' setting now only determines defaults for the per member<br/>
'ringinuse' setting and does not override per member settings like it does<br/>
in earlier versions.</li>
<li>Added 'F()' option. Similar to the dial option, this can be supplied with<br/>
arguments indicating where the callee should go after the caller is hung up,<br/>
or without options specified, the priority after the Queue will be used.</li>
<li>Added new option log_member_name_as_agent, which will cause the membername to<br/>
be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a<br/>
state_interface has been set.</li>
</ul>
<h2><a name="Newin11-SayUnixTime"></a>SayUnixTime</h2>
<ul>
<li>Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension<br/>
when receiving DTMF. Use the 'j' option to enable extension jumping. Also<br/>
changed arguments to SayUnixTime so that every option is truly optional even<br/>
when using multiple options (so that j option could be used without having to<br/>
manually specify timezone and format) There are other benefits, e.g., format<br/>
can now be used without specifying time zone as well.</li>
</ul>
<h2><a name="Newin11-Voicemail"></a>Voicemail</h2>
<ul>
<li>Addition of the VM_INFO function - see Function changes.</li>
<li>The imapserver, imapport, and imapflags configuration options can now be<br/>
overriden on a user by user basis.</li>
<li>When voicemail plays a message's envelope with saycid set to yes, when<br/>
reaching the caller id field it will play a recording of a file with the same<br/>
base name as the sender's callerid if there is a similarly named file in<br/>
<astspooldir>/recordings/callerids/</li>
<li>Voicemails now contains a unique message identifier "msg_id", which is stored<br/>
in the message envelope with the sound files. IMAP backends will now store<br/>
the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC<br/>
backends will store the message identifier in a "msg_id" column. See<br/>
UPGRADE.txt for more information.</li>
<li>Added VoiceMailPlayMsg application. This application will play a single<br/>
voicemail message from a mailbox. The result of the application, SUCCESS or<br/>
FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.</li>
</ul>
<h2><a name="Newin11-Functions"></a>Functions</h2>
<ul>
<li>Hangup handlers can be attached to channels using the CHANNEL() function.<br/>
Hangup handlers will run when the channel is hung up similar to the h<br/>
extension. The hangup_handler_push option will push a GoSub compatible<br/>
location in the dialplan onto the channel's hangup handler stack. The<br/>
hangup_handler_pop option will remove the last added location, and optionally<br/>
replace it with a new GoSub compatible location. The hangup_handler_wipe<br/>
option will remove all locations on the stack, and optionally add a new<br/>
location.</li>
<li>The expression parser now recognizes the ABS() absolute value function,<br/>
which will convert negative floating point values to positive values.</li>
<li>FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan<br/>
control of faxdetect.</li>
<li>Addition of the VM_INFO function that can be used to retrieve voicemail<br/>
user information, such as the email address and full name.<br/>
The MAILBOX_EXISTS dialplan function has been deprecated in favour of<br/>
VM_INFO.</li>
<li>The REDIRECTING function now supports the redirecting original party id<br/>
and reason.</li>
<li>Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()<br/>
lets you set some of the configuration options from the <a href="/wiki/pages/createpage.action?spaceKey=AST&title=general&linkCreation=true&fromPageId=20840465" class="createlink">general</a> section<br/>
of features.conf on a per-channel basis. FEATUREMAP() lets you customize<br/>
the key sequence used to activate built-in features, such as blindxfer,<br/>
and automon. See the built-in documentation for details.</li>
<li>MESSAGE(from) for incoming SIP messages now returns "display-name" <uri><br/>
instead of simply the uri. This is the format that MessageSend() can use<br/>
in the from parameter for outgoing SIP messages.</li>
<li>Added the PRESENCE_STATE function. This allows retrieving presence state<br/>
information from any presence state provider. It also allows setting<br/>
presence state information from a CustomPresence presence state provider.<br/>
See AMI/CLI changes for related commands.</li>
<li>Added the AMI_CLIENT function to make manager account attributes available<br/>
to the dialplan. It currently supports returning the current number of<br/>
active sessions for a given account.</li>
</ul>
<h1><a name="Newin11-ChannelDrivers"></a>Channel Drivers</h1>
<h2><a name="Newin11-chanlocal"></a>chan_local</h2>
<ul>
<li>Added a manager event "LocalBridge" for local channel call bridges between<br/>
the two pseudo-channels created.</li>
</ul>
<h2><a name="Newin11-chandahdi"></a>chan_dahdi</h2>
<ul>
<li>Added dialtone_detect option for analog ports to disconnect incoming<br/>
calls when dialtone is detected.</li>
<li>Added option colp_send to send ISDN connected line information. Allowed<br/>
settings are block, to not send any connected line information; connect, to<br/>
send connected line information on initial connect; and update, to send<br/>
information on any update during a call. Default is update.</li>
<li>Add options namedcallgroup and namedpickupgroup to support installations<br/>
where a higher number of groups (>64) is required.</li>
</ul>
<h2><a name="Newin11-chanmotif"></a>chan_motif</h2>
<ul>
<li>A new channel driver named chan_motif has been added which provides support for<br/>
Google Talk and Jingle in a single channel driver. This new channel driver includes<br/>
support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,<br/>
hold, unhold, and ringing notification. It is also compliant with the current Jingle<br/>
specification, current Google Jingle specification, and the original Google Talk<br/>
protocol.</li>
</ul>
<h2><a name="Newin11-chanooh323"></a>chan_ooh323</h2>
<ul>
<li>Added NAT support for RTP. Setting in config is 'nat', which can be set<br/>
globally and overriden on a peer by peer basis.</li>
<li>Direct media functionality has been added. Options in config are:<br/>
directmedia (directrtp) and directrtpsetup (earlydirect)</li>
<li>ChannelUpdate events now contain a CallRef header.</li>
</ul>
<h2><a name="Newin11-chansip"></a>chan_sip</h2>
<ul>
<li>Asterisk will no longer substitute CID number for CID name in the display<br/>
name field if CID number exists without a CID name. This change improves<br/>
compatibility with certain device features such as Avaya IP500's directory<br/>
lookup service.</li>
<li>A new setting for autocreatepeer (autocreatepeer=persistent) allows peers<br/>
created using that setting to not be removed during SIP reload.</li>
<li>Added settings recordonfeature and recordofffeature. When receiving an INFO<br/>
request with a "Record:" header, this will turn the requested feature on/off.<br/>
Allowed values are 'automon', 'automixmon', and blank to disable. Note that<br/>
dynamic features must be enabled and configured properly on the requesting<br/>
channel for this to function properly.</li>
<li>Add support to realtime for the 'callbackextension' option.</li>
<li>When multiple peers exist with the same address, but differing<br/>
callbackextension options, incoming requests that are matched by address<br/>
will be matched to the peer with the matching callbackextension if it is<br/>
available.</li>
<li>Two new NAT options, auto_force_rport and auto_comedia, have been added<br/>
which set the force_rport and comedia options automatically if Asterisk<br/>
detects that an incoming SIP request crossed a NAT after being sent by<br/>
the remote endpoint.</li>
<li>NAT settings are now a combinable list of options. The equivalent of the<br/>
deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.</li>
<li>Adds an option send_diversion which can be disabled to prevent<br/>
diversion headers from automatically being added to INVITE requests.</li>
<li>Add support for lightweight NAT keepalive. If enabled a blank packet will<br/>
be sent to the remote host at a given interval to keep the NAT mapping open.<br/>
This can be enabled using the keepalive configuration option.</li>
<li>Add option 'tonezone' to specify country code for indications. This option<br/>
can be set both globally and overridden for specific peers.</li>
<li>The SIP Security Events Framework now supports IPv6.</li>
<li>Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares<br/>
between multiple user agents. When set, for directmedia reinvites,<br/>
Asterisk will not send an immediate reinvite on an incoming call leg. This<br/>
option is useful when peered with another SIP user agent that is known to<br/>
send immediate direct media reinvites upon call establishment.</li>
<li>Add support for WebSocket transport. This can be configured using 'ws' or 'wss'<br/>
as the transport.</li>
<li>Add options subminexpiry and submaxexpiry to set limits of subscription<br/>
timer independently from registration timer settings. The setting of the<br/>
registration timer limits still is done by options minexpiry, maxexpiry<br/>
and defaultexpiry. For backwards compatibility the setting of minexpiry<br/>
and maxexpiry also is used to configure the subscription timer limits if<br/>
subminexpiry and submaxexpiry are not set in sip.conf.</li>
<li>Set registration timer limits to default values when reloading sip<br/>
configuration and values are not set by configuration.</li>
<li>Add options namedcallgroup and namedpickupgroup to support installations<br/>
where a higher number of groups (>64) is required.</li>
<li>When a MESSAGE request is received, the address the request was received from<br/>
is now saved in the SIP_RECVADDR variable.</li>
<li>Add ANI2/OLI parsing for SIP. The "From" header in INVITE requests is now<br/>
parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags. If present,<br/>
the ANI2/OLI information is set on the channel, which can be retrieved using<br/>
the CALLERID function.</li>
<li>Peers can now be configured to support negotiation of ICE candidates using<br/>
the setting icesupport. See res_rtp_asterisk changes for more information.</li>
<li>Added support for format attribute negotiation. See the Codecs changes for<br/>
more information.</li>
<li>Extra headers specified with SIPAddHeader are sent with the REFER message<br/>
when using Transfer application. See refer_addheaders in sip.conf.sample.</li>
</ul>
<h2><a name="Newin11-chanskinny"></a>chan_skinny</h2>
<ul>
<li>Added skinny version 17 protocol support.</li>
</ul>
<h2><a name="Newin11-chanunistim"></a>chan_unistim</h2>
<ul>
<li>Added ability to use multiple lines for a single phone. This allows multiple<br/>
calls to occur on a single phone, using callwaiting and switching between calls.</li>
<li>Added option 'sharpdial' allowing end dialing by pressing # key</li>
<li>Added option 'interdigit_timer' to control phone dial timeout</li>
<li>Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance</li>
<li>Added global 'debug' option, that enables debug in channel driver</li>
<li>Added ability to translate on-screen menu in multiple languages. Tested on<br/>
Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,<br/>
ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen<br/>
menu of phone</li>
<li>In addition to English added French and Russian languages for on-screen menus</li>
<li>Reworked dialing number input: added dialing by timeout, immediate dial on<br/>
on dialplan compare, phone number length now not limited by screen size</li>
<li>Added ability to pickup a call using features.conf defined value and<br/>
on-screen key</li>
</ul>
<h2><a name="Newin11-chanmISDN%3A"></a>chan_mISDN:</h2>
<ul>
<li>Add options namedcallgroup and namedpickupgroup to support installations<br/>
where a higher number of groups (>64) is required.</li>
</ul>
<h1><a name="Newin11-Core"></a>Core</h1>
<ul>
<li>The minimum DTMF duration can now be configured in asterisk.conf<br/>
as "mindtmfduration". The default value is (as before) set to 80 ms.<br/>
(previously it was only available in source code)</li>
<li>Named ACLs can now be specified in acl.conf and used in configurations that<br/>
use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is<br/>
used to specify an ACL, a similar form of 'acl' will add a named ACL to the<br/>
working ACL. In addition, some CLI commands have been added to provide<br/>
show information and allow for module reloading - see CLI Changes.</li>
<li>Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple<br/>
items (separated by commas), and items in the rule can be negated by prefixing<br/>
them with '!'. This simplifies Asterisk Realtime configurations, since it is no<br/>
longer necessray to control the order that the 'permit' and 'deny' columns are<br/>
returned from queries.</li>
<li>DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to<br/>
be used within the dynamic weight attribute when specifying a mapping.</li>
<li>CEL backends can now be configured to show "USER_DEFINED" in the EventName<br/>
header, instead of putting the user defined event name there. When enabled<br/>
the UserDefType header is added for user defined events. This feature is<br/>
enabled with the setting show_user_defined.</li>
<li>Macro has been deprecated in favor of GoSub. For redirecting and connected<br/>
line purposes use the following variables instead of their macro equivalents:<br/>
REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB,<br/>
CONNECTED_LINE_SEND_SUB_ARGS. For CCSS, use cc_callback_sub instead of<br/>
cc_callback_macro in channel configurations.</li>
<li>Asterisk can now use a system-provided NetBSD editline library (libedit) if it<br/>
is available.</li>
<li>Call files now support the "early_media" option to connect with an outgoing<br/>
extension when early media is received.</li>
</ul>
<h2><a name="Newin11-AGI"></a>AGI</h2>
<ul>
<li>A new channel variable, AGIEXITONHANGUP, has been added which allows<br/>
Asterisk to behave like it did in Asterisk 1.4 and earlier where the<br/>
AGI application would exit immediately after a channel hangup is detected.</li>
<li>IPv6 addresses are now supported when using FastAGI (agi://). Hostnames<br/>
are resolved and each address is attempted in turn until one succeeds or<br/>
all fail.</li>
</ul>
<h2><a name="Newin11-AMI%28AsteriskManagerInterface%29"></a>AMI (Asterisk Manager Interface)</h2>
<ul>
<li>The originate action now has an option "EarlyMedia" that enables the<br/>
call to bridge when we get early media in the call. Previously,<br/>
early media was disregarded always when originating calls using AMI.</li>
<li>Added setvar= option to manager accounts (much like sip.conf)</li>
<li>Originate now generates an error response if the extension given is not found<br/>
in the dialplan</li>
<li>MixMonitor will now show IDs associated with the mixmonitor upon creating<br/>
them if the i(variable) option is used. StopMixMonitor will accept<br/>
MixMonitorID as an option to close specific MixMonitors.</li>
<li>The SIPshowpeer manager action response field "SIP-Forcerport" has been<br/>
updated to include information about peers configured with<br/>
nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is<br/>
detected, and "a" if it is set and nat is not detected. "Y" and "N" are still<br/>
returned if auto_force_rport is not enabled.</li>
<li>Added SIPpeerstatus manager command which will generate PeerStatus events<br/>
similar to the existing PeerStatus events found in chan_sip on demand.</li>
<li>Hangup now can take a regular expression as the Channel option. If you want<br/>
to hangup multiple channels, use /regex/ as the Channel option. Existing<br/>
behavior to hanging up a single channel is unchanged, but if you pass a regex,<br/>
the manager will send you a list of channels back that were hung up.</li>
<li>Support for IPv6 addresses has been added.</li>
<li>AMI Events can now be documented in the Asterisk source. Note that AMI event<br/>
documentation is only generated when Asterisk is compiled using 'make full'.<br/>
See the CLI section for commands to display AMI event information.</li>
<li>The AMI Hangup event now includes the AccountCode header so you can easily<br/>
correlate with AMI Newchannel events.</li>
<li>The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include<br/>
the StateInterface of the queue member.</li>
<li>Added AMI event SessionTimeout in the Call category that is issued when a<br/>
call is terminated due to either RTP stream inactivity or SIP session timer<br/>
expiration.</li>
<li>CEL events can now contain a user defined header UserDefType. See core<br/>
changes for more information.</li>
<li>OOH323 ChannelUpdate events now contain a CallRef header.</li>
<li>Added PresenceState command. This command will report the presence state for<br/>
the given presence provider.</li>
<li>Added Parkinglots command. This will list all parking lots as a series of<br/>
AMI Parkinglot events.</li>
<li>Added MessageSend command. This behaves in the same manner as the<br/>
MessageSend application, and is a technolgoy agnostic mechanism to send out<br/>
of call text messages.</li>
<li>Added "message" class authorization. This grants an account permission to<br/>
send out of call messages. Write-only.</li>
</ul>
<h2><a name="Newin11-CLI"></a>CLI</h2>
<ul>
<li>The "dialplan add include" command has been modified to create context a context<br/>
if one does not already exist. For instance, "dialplan add include foo into bar"<br/>
will create context "bar" if it does not already exist.</li>
<li>A "dialplan remove context" command has been added to remove a context from<br/>
the dialplan</li>
<li>The "mixmonitor list <channel>" command will now show MixMonitor ID, and the<br/>
filenames of all running mixmonitors on a channel.</li>
<li>The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if<br/>
numeric instead of 0, 1, or 2.</li>
<li>"stun show status" will show a table describing how the STUN client is<br/>
behaving.</li>
<li>"acl show <a href="/wiki/pages/createpage.action?spaceKey=AST&title=named+acl&linkCreation=true&fromPageId=20840465" class="createlink">named acl</a>" will show information regarding a Named ACL. The<br/>
acl module can be reloaded with "reload acl".</li>
<li>Added CLI command to display AMI event information - "manager show events",<br/>
which shows a list of all known and documented AMI events, and "manager show<br/>
event <a href="/wiki/pages/createpage.action?spaceKey=AST&title=event+name&linkCreation=true&fromPageId=20840465" class="createlink">event name</a>", which shows detail information about a specific AMI<br/>
event.</li>
<li>The result of the CLI command "queue show" now includes the state interface<br/>
information of the queue member.</li>
<li>The command "core set verbose" will now set a separate level of logging for<br/>
each remote console without affecting any other console.</li>
<li>Added command "cdr show pgsql status" to check connection status</li>
<li>"sip show channel" will now display the complete route set.</li>
<li>Added "presencestate list" command. This command will list all custom<br/>
presence states that have been set by using the PRESENCE_STATE dialplan<br/>
function.</li>
<li>Added "presencestate change <entity> <state>[,<subtype>[,message<a href="/wiki/pages/createpage.action?spaceKey=AST&title=%2Coptions&linkCreation=true&fromPageId=20840465" class="createlink">,options</a>]]"<br/>
command. This changes a custom presence to a new state.</li>
</ul>
<h2><a name="Newin11-Codecs"></a>Codecs</h2>
<ul>
<li>Codec lists may now be modified by the '!' character, to allow succinct<br/>
specification of a list of codecs allowed and disallowed, without the<br/>
requirement to use two different keywords. For example, to specify all<br/>
codecs except g729 and g723, one need only specify allow=all,!g729,!g723.</li>
<li>Add support for parsing SDP attributes, generating SDP attributes, and<br/>
passing it through. This support includes codecs such as H.263, H.264, SILK,<br/>
and CELT. You are able to set up a call and have attribute information pass.<br/>
This should help considerably with video calls.</li>
<li>The iLBC codec can now use a system-provided iLBC library if one is installed,<br/>
just like the GSM codec.</li>
</ul>
<h2><a name="Newin11-DUNDichanges"></a>DUNDi changes</h2>
<ul>
<li>Added CLI commands dundi show hints and dundi show cache which will list DUNDi<br/>
'DONTASK' hints in the cache and list all DUNDi cache entires respectively.</li>
</ul>
<h2><a name="Newin11-Logging"></a>Logging</h2>
<ul>
<li>Asterisk version and build information is now logged at the beginning of a<br/>
log file.</li>
<li>Threads belonging to a particular call are now linked with callids which get<br/>
added to any log messages produced by those threads. Log messages can now be<br/>
easily identified as involved with a certain call by looking at their call id.<br/>
Call ids may also be attached to log messages for just about any case where<br/>
it can be determined to be related to a particular call.</li>
<li>Each logging destination and console now have an independent notion of the<br/>
current verbosity level. Logger.conf now allows an optional argument to<br/>
the 'verbose' specifier, indicating the level of verbosity sent to that<br/>
particular logging destination. Additionally, remote consoles now each<br/>
have their own verbosity level. The command 'core set verbose' will now set<br/>
a separate level for each remote console without affecting any other<br/>
console.</li>
</ul>
<h2><a name="Newin11-MusicOnHold"></a>Music On Hold</h2>
<ul>
<li>Added 'announcement' option which will play at the start of MOH and between<br/>
songs in modes of MOH that can detect transitions between songs (eg.<br/>
files, mp3, etc).</li>
</ul>
<h2><a name="Newin11-Parking"></a>Parking</h2>
<ul>
<li>New per parking lot options: comebackcontext and comebackdialtime. See<br/>
configs/features.conf.sample for more details.</li>
<li>Channel variable PARKER is now set when comebacktoorigin is disabled in<br/>
a parking lot.</li>
<li>Channel variable PARKEDCALL is now set with the name of the parking lot<br/>
when a timeout occurs.</li>
</ul>
<h2><a name="Newin11-CDRs"></a>CDRs</h2>
<h3><a name="Newin11-CDRPostgresqlDriver"></a>CDR Postgresql Driver</h3>
<ul>
<li>Added command "cdr show pgsql status" to check connection status</li>
</ul>
<h3><a name="Newin11-CDRAdaptiveODBCDriver"></a>CDR Adaptive ODBC Driver</h3>
<ul>
<li>Added schema option for databases that support specifying a schema.</li>
</ul>
<h1><a name="Newin11-ResourceModules"></a>Resource Modules</h1>
<h2><a name="Newin11-Calendars"></a>Calendars</h2>
<ul>
<li>A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not<br/>
CALENDAR_WRITE has completed successfully.</li>
</ul>
<h2><a name="Newin11-resrtpasterisk"></a>res_rtp_asterisk</h2>
<ul>
<li>A new option, 'probation' has been added to rtp.conf<br/>
RTP in strictrtp mode can now require more than 1 packet to exit learning<br/>
mode with a new source (and by default requires 4). The probation option<br/>
allows the user to change the required number of packets in sequence to any<br/>
desired value. Use a value of 1 to essentially restore the old behavior.<br/>
Also, with strictrtp on, Asterisk will now drop all packets until learning<br/>
mode has successfully exited. These changes are based on how pjmedia handles<br/>
media sources and source changes.</li>
<li>Add support for ICE/STUN/TURN in res_rtp_asterisk. This option can be<br/>
enabled or disabled using the icesupport setting. A variety of other<br/>
settings have been introduced to configure STUN/TURN connections.</li>
</ul>
<h2><a name="Newin11-rescorosync"></a>res_corosync</h2>
<ul>
<li>A new module, res_corosync, has been introduced. This module uses the<br/>
Corosync cluster enginer (<a href="http://www.corosync.org" class="external-link" rel="nofollow">http://www.corosync.org</a>) to allow a local cluster<br/>
of Asterisk servers to both Message Waiting Indication (MWI) and/or<br/>
Device State (presence) information. This module is very similar to, and<br/>
is a replacement for the res_ais module that was in previous releases of<br/>
Asterisk.</li>
</ul>
<h2><a name="Newin11-resxmpp"></a>res_xmpp</h2>
<ul>
<li>This module adds a cleaned up, drop-in replacement for res_jabber called<br/>
res_xmpp. This provides the same externally facing functionality but is<br/>
implemented differently internally. res_jabber has been deprecated in favor<br/>
of res_xmpp; please see the UPGRADE.txt file for more information.</li>
</ul>
<h1><a name="Newin11-Scripts"></a>Scripts</h1>
<ul>
<li>The safe_asterisk script has been updated to allow several of its parameters<br/>
to be set from environment variables. This also enables a custom run<br/>
directory of Asterisk to be specified, instead of defaulting to /tmp.</li>
<li>The live_ast script will now look for the LIVE_AST_BASE_DIR variable and use<br/>
its value to determine the directory to assume is the top-level directory of<br/>
the source tree. If the variable is not set, it defaults to the current<br/>
behavior and uses the current working directory.</li>
</ul>
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