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411 lines
19 KiB
411 lines
19 KiB
===========================================================
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===
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=== Information for upgrading between Asterisk versions
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===
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=== These files document all the changes that MUST be taken
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=== into account when upgrading between the Asterisk
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=== versions listed below. These changes may require that
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=== you modify your configuration files, dialplan or (in
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=== some cases) source code if you have your own Asterisk
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=== modules or patches. These files also include advance
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=== notice of any functionality that has been marked as
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=== 'deprecated' and may be removed in a future release,
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=== along with the suggested replacement functionality.
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===
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=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
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=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
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=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
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=== UPGRADE-1.8.txt -- Upgrade info for 1.6 to 1.8
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=== UPGRADE-10.txt -- Upgrade info for 1.8 to 10
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=== UPGRADE-11.txt -- Upgrade info for 10 to 11
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===
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===========================================================
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AgentMonitorOutgoing
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- Application removed. It was a holdover from when AgentCallbackLogin was
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removed.
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NoCDR:
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- This application is deprecated. Please use the CDR_PROP function instead.
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ResetCDR:
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- The 'w' and 'a' options have been removed. Dispatching CDRs to registered
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backends occurs on an as-needed basis in order to preserve linkedid
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propagation and other needed behavior.
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- The 'e' option is deprecated. Please use the CDR_PROP function to enable
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CDRs on a channel that they were previously disabled on.
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- The ResetCDR application is no longer a part of core Asterisk, and instead
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is now delivered as part of app_cdr.
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ForkCDR:
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- ForkCDR no longer automatically resets the forked CDR. See the 'r' option
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for more information.
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- Variables are no longer purged from the original CDR. See the 'v' option for
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more information.
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- The 'A' option has been removed. The Answer time on a CDR is never updated
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once set.
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- The 'd' option has been removed. The disposition on a CDR is a function of
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the state of the channel and cannot be altered.
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- The 'D' option has been removed. Who the Party B is on a CDR is a function
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of the state of the respective channels, and cannot be altered.
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- The 'r' option has been changed. Previously, ForkCDR always reset the CDR
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such that the start time and, if applicable, the answer time was updated.
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Now, by default, ForkCDR simply forks the CDR, maintaining any times. The
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'r' option now triggers the Reset, setting the start time (and answer time
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if applicable) to the current time.
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- The 's' option has been removed. A variable can be set on the original CDR
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if desired using the CDR function, and removed from a forked CDR using the
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same function.
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- The 'T' option has been removed. The concept of DONT_TOUCH and LOCKED no
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longer applies in the CDR engine.
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- The 'v' option now prevents the copy of the variables from the original CDR
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to the forked CDR. Previously the variables were always copied but were
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removed from the original. Removing variables from a CDR can have unintended
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side effects - this option allows the user to prevent propagation of
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variables from the original to the forked without modifying the original.
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AMI:
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- The SIP SIPqualifypeer action now sends a response indicating it will qualify
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a peer once a peer has been found to qualify. Once the qualify has been
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completed it will now issue a SIPqualifypeerdone event.
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- Version 1.4 - The details of what happens to a channel when a masquerade
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happens (transfers, parking, etc) have changed.
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- The Masquerade event now includes the Uniqueid's of the clone and original
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channels.
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- Channels no longer swap Uniqueid's as a result of the masquerade.
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- Instead of a shell game of renames, there's now a single rename, appending
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<ZOMBIE> to the name of the original channel.
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- The AMI events 'ParkedCall', 'ParkedCallTimeOut', 'ParkedCallGiveUp', and
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'UnParkedCall' have changed significantly in the new res_parking module.
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- The 'Channel' and 'From' headers are gone. For the channel that was parked
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or is coming out of parking, a 'Parkee' channel snapshot is issued and it
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has a number of fields associated with it. The old 'Channel' header relayed
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the same data as the new 'ParkeeChannel' header.
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- The 'From' field was ambiguous and changed meaning depending on the event.
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for most of these, it was the name of the channel that parked the call
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(the 'Parker'). There is no longer a header that provides this channel name,
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however the 'ParkerDialString' will contain a dialstring to redial the
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device that parked the call.
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- On UnParkedCall events, the 'From' header would instead represent the
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channel responsible for retrieving the parkee. It receives a channel
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snapshot labeled 'Retriever'. The 'from' field is is replaced with
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'RetrieverChannel'.
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- Lastly, the 'Exten' field has been replaced with 'ParkingSpace'.
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CEL:
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- The Uniqueid field for a channel is now a stable identifier, and will not
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change due to transfers, parking, etc.
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Build System:
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- Asterisk now optionally uses libxslt to improve XML documentation generation
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and maintainability. If libxslt is not available on the system, some XML
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documentation will be incomplete.
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Core:
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- The following channel variables have changed behavior which is described in
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the CHANGES file: TRANSFER_CONTEXT, BRIDGEPEER, BRIDGEPVTCALLID,
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ATTENDED_TRANSFER_COMPLETE_SOUND, DYNAMIC_FEATURENAME, and DYNAMIC_PEERNAME.
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Queues:
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- Queue logging for PAUSEALL/UNPAUSEALL now only occurs if the interface this is
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performed on is a member of at least one queue.
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- Queue strategy rrmemory now has a predictable order similar to strategy
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rrordered. Members will be called in the order that they are added to the
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queue.
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- CDR behavior in app_queue has been modified slightly. The CDR record will
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now only record a disposition of BUSY if all Queue members were actually
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busy on a call or some Queue members were busy or paused. Previously, any
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Queue member being paused would result in a disposition of BUSY.
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- Removed the queues.conf check_state_unknown option. It is no longer
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necessary.
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- It is now possible to play the Queue prompts to the first user waiting in a
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call queue. Note that this may impact the ability for agents to talk with
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users, as a prompt may still be playing when an agent connects to the user.
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This ability is disabled by default but can be enabled on an individual
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queue using the 'announce-to-first-user' option.
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Dial:
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- Now recognizes 'W' to pause sending DTMF for one second in addition to
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the previously existing 'w' that paused sending DTMF for half a second.
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ExternalIVR:
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- Now recognizes 'W' to pause sending DTMF for one second in addition to
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the previously existing 'w' that paused sending DTMF for half a second.
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SendDTMF:
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- Now recognizes 'W' to pause sending DTMF for one second in addition to
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the previously existing 'w' that paused sending DTMF for half a second.
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SetAMAFlags
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- This application is deprecated in favor of the CHANNEL function.
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chan_agent:
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- The updatecdr option has been removed. Altering the names of channels on a
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CDR is not supported - the name of the channel is the name of the channel,
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and pretending otherwise helps no one.
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- The AGENTUPDATECDR channel variable has also been removed, for the same
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reason as the updatecdr option.
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- chan_agent is removed and replaced with AgentLogin and AgentRequest dialplan
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applications. Agents are connected with callers using the new AgentRequest
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dialplan application. The Agents:<agent-id> device state is available to
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monitor the status of an agent. See agents.conf.sample for valid
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configuration options.
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chan_bridge
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- chan_bridge is removed and its functionality is incorporated into ConfBridge
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itself.
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chan_dahdi:
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- Analog port dialing and deferred DTMF dialing for PRI now distinguishes
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between 'w' and 'W'. The 'w' pauses dialing for half a second. The 'W'
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pauses dialing for one second.
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- The default for inband_on_proceeding has changed to no.
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chan_local:
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- The /b option has been removed.
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Dialplan:
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- All channel and global variable names are evaluated in a case-sensitive manner.
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In previous versions of Asterisk, variables created and evaluated in the
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dialplan were evaluated case-insensitively, but built-in variables and variable
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evaluation done internally within Asterisk was done case-sensitively.
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- Asterisk has always had code to ignore dash '-' characters that are not
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part of a character set in the dialplan extensions. The code now
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consistently ignores these characters when matching dialplan extensions.
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- BRIDGE_FEATURES channel variable is now casesensitive for feature letter codes.
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Uppercase variants apply them to the calling party while lowercase variants
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apply them to the called party.
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Features:
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- The features.conf [applicationmap] <FeatureName> ActivatedBy option is
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no longer honored. The feature is always activated by the channel that has
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DYNAMIC_FEATURES defined on it when it enters the bridge. Use predial to set
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different values of DYNAMIC_FEATURES on the channels
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- Executing a dynamic feature on the bridge peer in a multi-party bridge will
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execute it on all peers of the activating channel.
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- There is no longer an explicit 'features reload' CLI command. Features can still be
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reloaded using 'module reload features'.
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Parking:
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- The arguments for the Park, ParkedCall, and ParkAndAnnounce applications have
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been modified significantly. See the application documents for specific details.
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Also parking lot configuration is now done in res_parking.conf instead of
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features.conf
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From 10 to 11:
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Voicemail:
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- All voicemails now have a "msg_id" which uniquely identifies a message. For
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users of filesystem and IMAP storage of voicemail, this should be transparent.
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For users of ODBC, you will need to add a "msg_id" column to your voice mail
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messages table. This should be a string capable of holding at least 32 characters.
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All messages created in old Asterisk installations will have a msg_id added to
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them when required. This operation should be transparent as well.
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Parking:
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- The comebacktoorigin setting must now be set per parking lot. The setting in
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the general section will not be applied automatically to each parking lot.
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- The BLINDTRANSFER channel variable is deleted from a channel when it is
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bridged to prevent subtle bugs in the parking feature. The channel
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variable is used by Asterisk internally for the Park application to work
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properly. If you were using it for your own purposes, copy it to your
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own channel variable before the channel is bridged.
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res_ais:
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- Users of res_ais in versions of Asterisk prior to Asterisk 11 must change
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to use the res_corosync module, instead. OpenAIS is deprecated, but
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Corosync is still actively developed and maintained. Corosync came out of
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the OpenAIS project.
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Dialplan Functions:
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- MAILBOX_EXISTS has been deprecated. Use VM_INFO with the 'exists' parameter
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instead.
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- Macro has been deprecated in favor of GoSub. For redirecting and connected
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line purposes use the following variables instead of their macro equivalents:
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REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS,
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CONNECTED_LINE_SEND_SUB, CONNECTED_LINE_SEND_SUB_ARGS.
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- The REDIRECTING function now supports the redirecting original party id
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and reason.
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- The HANGUPCAUSE and HANGUPCAUSE_KEYS functions have been introduced to
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provide a replacement for the SIP_CAUSE hash. The HangupCauseClear
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application has also been introduced to remove this data from the channel
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when necessary.
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func_enum:
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- ENUM query functions now return a count of -1 on lookup error to
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differentiate between a failed query and a successful query with 0 results
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matching the specified type.
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CDR:
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- cdr_adaptive_odbc now supports specifying a schema so that Asterisk can
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connect to databases that use schemas.
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Configuration Files:
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- Files listed below have been updated to be more consistent with how Asterisk
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parses configuration files. This makes configuration files more consistent
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with what is expected across modules.
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- cdr.conf: [general] and [csv] sections
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- dnsmgr.conf
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- dsp.conf
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- The 'verbose' setting in logger.conf now takes an optional argument,
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specifying the verbosity level for each logging destination. The default,
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if not otherwise specified, is a verbosity of 3.
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AMI:
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- DBDelTree now correctly returns an error when 0 rows are deleted just as
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the DBDel action does.
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- The IAX2 PeerStatus event now sends a 'Port' header. In Asterisk 10, this was
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erroneously being sent as a 'Post' header.
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CCSS:
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- Macro is deprecated. Use cc_callback_sub instead of cc_callback_macro
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in channel configurations.
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app_meetme:
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- The 'c' option (announce user count) will now work even if the 'q' (quiet)
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option is enabled.
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app_followme:
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- Answered outgoing calls no longer get cut off when the next step is started.
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You now have until the last step times out to decide if you want to accept
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the call or not before being disconnected.
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chan_gtalk:
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- chan_gtalk has been deprecated in favor of the chan_motif channel driver. It is recommended
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that users switch to using it as it is a core supported module.
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chan_jingle:
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- chan_jingle has been deprecated in favor of the chan_motif channel driver. It is recommended
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that users switch to using it as it is a core supported module.
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SIP
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===
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- A new option "tonezone" for setting default tonezone for the channel driver
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or individual devices
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- A new manager event, "SessionTimeout" has been added and is triggered when
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a call is terminated due to RTP stream inactivity or SIP session timer
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expiration.
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- SIP_CAUSE is now deprecated. It has been modified to use the same
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mechanism as the HANGUPCAUSE function. Behavior should not change, but
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performance should be vastly improved. The HANGUPCAUSE function should now
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be used instead of SIP_CAUSE. Because of this, the storesipcause option in
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sip.conf is also deprecated.
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- The sip paramater for Originating Line Information (oli, isup-oli, and
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ss7-oli) is now parsed out of the From header and copied into the channel's
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ANI2 information field. This is readable from the CALLERID(ani2) dialplan
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function.
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- ICE support has been added and is enabled by default. Some endpoints may have
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problems with the ICE candidates within the SDP. If this is the case ICE support
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can be disabled globally or on a per-endpoint basis using the icesupport
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configuration option. Symptoms of this include one way media or no media flow.
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chan_unistim
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- Due to massive update in chan_unistim phone keys functions and on-screen
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information changed.
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users.conf:
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- A defined user with hasvoicemail=yes now finally uses a Gosub to stdexten
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as documented in extensions.conf.sample since v1.6.0 instead of a Macro as
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documented in v1.4. Set the asterisk.conf stdexten=macro parameter to
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invoke the stdexten the old way.
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res_jabber
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- This module has been deprecated in favor of the res_xmpp module. The res_xmpp
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module is backwards compatible with the res_jabber configuration file, dialplan
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functions, and AMI actions. The old CLI commands can also be made available using
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the res_clialiases template for Asterisk 11.
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From 1.8 to 10:
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cel_pgsql:
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- This module now expects an 'extra' column in the database for data added
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using the CELGenUserEvent() application.
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ConfBridge
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- ConfBridge's dialplan arguments have changed and are not
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backwards compatible.
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File Interpreters
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- The format interpreter formats/format_sln16.c for the file extension
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'.sln16' has been removed. The '.sln16' file interpreter now exists
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in the formats/format_sln.c module along with new support for sln12,
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sln24, sln32, sln44, sln48, sln96, and sln192 file extensions.
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HTTP:
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- A bindaddr must be specified in order for the HTTP server
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to run. Previous versions would default to 0.0.0.0 if no
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bindaddr was specified.
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Gtalk:
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- The default value for 'context' and 'parkinglots' in gtalk.conf has
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been changed to 'default', previously they were empty.
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chan_dahdi:
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- The mohinterpret=passthrough setting is deprecated in favor of
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moh_signaling=notify.
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pbx_lua:
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- Execution no longer continues after applications that do dialplan jumps
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(such as app.goto). Now when an application such as app.goto() is called,
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control is returned back to the pbx engine and the current extension
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function stops executing.
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- the autoservice now defaults to being on by default
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- autoservice_start() and autoservice_start() no longer return a value.
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Queue:
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- Mark QUEUE_MEMBER_PENALTY Deprecated it never worked for realtime members
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- QUEUE_MEMBER is now R/W supporting setting paused, ignorebusy and penalty.
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Asterisk Database:
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- The internal Asterisk database has been switched from Berkeley DB 1.86 to
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SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
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utility in the UTILS section of menuselect. If an existing astdb is found and no
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astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
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convert an existing astdb to the SQLite3 version automatically at runtime. If
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moving back from Asterisk 10 to Asterisk 1.8, the astdb2bdb utility can be used
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to create a Berkeley DB copy of the SQLite3 astdb that Asterisk 10 uses.
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Manager:
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- The AMI protocol version was incremented to 1.2 as a result of changing two
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instances of the Unlink event to Bridge events. This change was documented
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as part of the AMI 1.1 update, but two Unlink events were inadvertently left
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unchanged.
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Module Support Level
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- All modules in the addons, apps, bridge, cdr, cel, channels, codecs,
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formats, funcs, pbx, and res have been updated to include MODULEINFO data
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that includes <support_level> tags with a value of core, extended, or deprecated.
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More information is available on the Asterisk wiki at
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https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
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Deprecated modules are now marked to not build by default and must be explicitly
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enabled in menuselect.
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chan_sip:
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- Setting of HASH(SIP_CAUSE,<slave-channel-name>) on channels is now disabled
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by default. It can be enabled using the 'storesipcause' option. This feature
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has a significant performance penalty.
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- In order to improve compliance with RFC 3261, SIP usernames are now properly
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escaped when encoding reserved characters. Prior to this change, the use of
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these characters in certain SIP settings affecting usernames could cause
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injections of these characters in their raw form into SIP headers which could
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in turn cause all sorts of nasty behaviors. All characters that are not
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alphanumeric or are not contained in the the following lists specified by
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RFC 3261 section 25.1 will be escaped as %XX when encoding a SIP username:
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* mark: "-" / "_" / "." / "!" / "~" / "*" / "'" / "(" / ")"
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* user-unreserved: "&" / "=" / "+" / "$" / "," / ";" / "?" / "/"
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UDPTL:
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- The default UDPTL port range in udptl.conf.sample differed from the defaults
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in the source. If you didn't have a config file, you got 4500 to 4599. Now the
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default is 4000 to 4999.
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===========================================================
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===========================================================
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